1. 1d03a75 Remove cricket::RtpTransceiverDirection by Steve Anton · 7 years ago
  2. 9158ef6 Reland "Add AddTransceiver and GetTransceivers to PeerConnection" by Steve Anton · 7 years ago
  3. 8b13f96 Revert "Add AddTransceiver and GetTransceivers to PeerConnection" by Steve Anton · 7 years ago
  4. f93d280 Add AddTransceiver and GetTransceivers to PeerConnection by Steve Anton · 7 years ago
  5. 3163867 Reland "SetRemoteDescriptionObserverInterface added." by Henrik Boström · 7 years ago
  6. a4ecf55 Revert "SetRemoteDescriptionObserverInterface added." by Henrik Boström · 7 years ago
  7. 6c7ec32 SetRemoteDescriptionObserverInterface added. by Henrik Boström · 7 years ago
  8. de93943 Revert "Revert "Encode log events periodically instead of for every event."" by Bjorn Terelius · 7 years ago
  9. 79e7960 Add SDP semantics option to RTCConfiguration by Steve Anton · 7 years ago
  10. 4171afb Use RtpTransceivers in PeerConnection by Steve Anton · 7 years ago
  11. 33c5c7f Revert "Encode log events periodically instead of for every event." by Zhi Huang · 7 years ago
  12. b154c27 Encode log events periodically instead of for every event. by Bjorn Terelius · 7 years ago
  13. d25da37 Inline various trivial methods from the WebRtcSession merge by Steve Anton · 7 years ago
  14. 75737c0 Merge WebRtcSession into PeerConnection by Steve Anton · 7 years ago
  15. ba81867 Prepare WebRtcSession to be merged into PeerConnection by Steve Anton · 7 years ago
  16. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  17. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  18. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  19. d5585ca Move almost all references from WebRtcSession to PeerConnection by Steve Anton · 7 years ago
  20. 8a63f78 Rewrite the remaining few WebRtcSession tests. by Steve Anton · 7 years ago
  21. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  22. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  23. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  24. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  25. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  26. 99c3fe5 Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter by Elad Alon · 7 years ago
  27. bdcee28 TurnCustomizer - an interface for modifying stun messages sent by TurnPort by Jonas Oreland · 7 years ago
  28. 933d8b0 Reland "Added PeerConnectionObserver::OnRemoveTrack." by Henrik Boström · 7 years ago
  29. 6c0c55c Revert "Added PeerConnectionObserver::OnRemoveTrack." by Alex Loiko · 7 years ago
  30. ba97ba7 Added PeerConnectionObserver::OnRemoveTrack. by Henrik Boström · 7 years ago
  31. 604427b Revert "TurnCustomizer - an interface for modifying stun messages sent by TurnPort" by Guido Urdaneta · 7 years ago
  32. b23ed7f TurnCustomizer - an interface for modifying stun messages sent by TurnPort by Jonas Oreland · 7 years ago
  33. acb2417 Fix apparent copy/paste error in comment (PeerConnection) by Elad Alon · 7 years ago
  34. 8c0f7a7 Add GetRemoteAudioSSLCertificate() to PeerConnection by Steve Anton · 7 years ago
  35. 978b876 Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  36. bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 7 years ago
  37. 3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  38. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  39. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/peerconnection.h]
  40. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  41. 3c74766 Revert of Adding support for Unified Plan offer/answer negotiation. (patchset #9 id:500001 of https://codereview.webrtc.org/2991693002/ ) by olka · 7 years ago
  42. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  43. ec390b5 When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded by korniltsev.anatoly · 7 years ago
  44. 038834f Reinstate "Add additional check when setting RTCConfiguration" by Steve Anton · 7 years ago
  45. 26d5e2e Revert "Add additional check when setting RTCConfiguration" by Magnus Jedvert · 7 years ago
  46. 8110bed Add additional check when setting RTCConfiguration by Steve Anton · 7 years ago
  47. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  48. 4b97980 Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 7 years ago
  49. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 7 years ago
  50. 9641c13 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 7 years ago
  51. 3386025 Initialize PeerConnection members in declaration order and destroy them in reverse order. by terelius · 7 years ago
  52. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 7 years ago
  53. 1dcb164 Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 7 years ago
  54. 81bf7b0 Pass ownership of candidate to PeerConnection::OnIceCandidate by jbauch · 7 years ago
  55. 6dfd53a Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange by zstein · 7 years ago
  56. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (97%) from webrtc/api/peerconnection.h]
  57. 293e926 Reland of: Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  58. 1e23461 Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ ) by deadbeef · 8 years ago
  59. 7a5fa6c Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  60. fe4a8a4 Implement current/pending session description methods. by deadbeef · 8 years ago
  61. 46c7389 Adding GetConfiguration to PeerConnection. by deadbeef · 8 years ago
  62. 82ebe02 Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed]. by hbos · 8 years ago
  63. e7c338f Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." (patchset #1 id:1 of https://codereview.webrtc.org/2402993002/ ) by sprang · 8 years ago
  64. 57cb873 Revert of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." (patchset #1 id:1 of https://codereview.webrtc.org/2361053003/ ) by sprang · 8 years ago
  65. fc9414a Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." by johan · 8 years ago
  66. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  67. 74e1a4f PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added. by hbos · 8 years ago
  68. 4cedf2b Add signaling to support ICE renomination. by Honghai Zhang · 8 years ago
  69. 68343a8 Revert of Remove the obsolete enum webrtc::PeerConnectionInterface::IceState. (patchset #1 id:1 of https://codereview.webrtc.org/2256663002/ ) by perkj · 8 years ago
  70. 31dea98 Remove the obsolete enum webrtc::PeerConnectionInterface::IceState. by johan · 8 years ago
  71. 14d5dbe Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 8 years ago
  72. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 8 years ago
  73. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 8 years ago
  74. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  75. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 8 years ago
  76. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  77. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 8 years ago
  78. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  79. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 8 years ago
  80. d03c23b Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 8 years ago
  81. d7973cc Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ ) by hbos · 8 years ago
  82. 400781a Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 8 years ago
  83. 91dd567 Only use PortAllocator on the network thread. by deadbeef · 8 years ago
  84. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  85. a1c3035 Relanding: Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 8 years ago
  86. c55fb30 Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ ) by deadbeef · 8 years ago
  87. 48e9d05 Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 8 years ago
  88. 6ab3db2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 8 years ago
  89. 65fc62e Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  90. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 8 years ago
  91. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 8 years ago
  92. c36b31b Embed a cricket::MediaConfig in RTCConfiguration. by nisse · 8 years ago
  93. d61bf80 Removed MediaStreamTrackInterface::set_state by perkj · 8 years ago
  94. 7fb69db Reland the CL to remove candidates when doing continual gathering by Honghai Zhang · 8 years ago
  95. 6f59a4f Revert of Remove candidates when doing continual gathering (patchset #15 id:560001 of https://codereview.webrtc.org/1648813004/ ) by tommi · 8 years ago
  96. 84430da When doing candidate re-gathering in the same generation, Remove the existing local candidate on the same network by honghaiz · 8 years ago
  97. aac2dea Changed defaults for CreateAnswer in non-constraint mode by hta · 8 years ago
  98. f0dcfe2 Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource. by perkj · 8 years ago
  99. a2a49d9 This CL provides interfaces that do not use constraints for by hta · 8 years ago
  100. 0ed85b2 Track pending ICE restarts independently for different media sections. by deadbeef · 8 years ago