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webrtc / src.git / 3b3622fafc6d1fd910074f7bab78c8427d80649e / pc / peerconnection.cc
  1. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  2. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/pc/peerconnection.cc]
  3. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 8 years ago
  4. 248fd4f Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread ( https://codereview.webrtc.org/3007473002/ ) by eladalon · 8 years ago
  5. 23814b7 Revert of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #4 id:200001 of https://codereview.webrtc.org/3005153002/ ) by eladalon · 8 years ago
  6. d67cefb Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #1 id:1 of https://codereview.webrtc.org/3010143002/ ) by eladalon · 8 years ago
  7. 141aacb Fix the Chromium crash when creating an answer without a remote description. by zhihuang · 8 years ago
  8. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 8 years ago
  9. 3c74766 Revert of Adding support for Unified Plan offer/answer negotiation. (patchset #9 id:500001 of https://codereview.webrtc.org/2991693002/ ) by olka · 8 years ago
  10. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 8 years ago
  11. d21eab3e Add "max_ipv6_networks" field to RTCConfiguration. by deadbeef · 8 years ago
  12. ec390b5 When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded by korniltsev.anatoly · 8 years ago
  13. 038834f Reinstate "Add additional check when setting RTCConfiguration" by Steve Anton · 8 years ago
  14. 300bf8e Reinstate "API for periodically regathering ICE candidates" by Steve Anton · 8 years ago
  15. 3beb207 Revert "API for periodically regathering ICE candidates" by Magnus Jedvert · 8 years ago
  16. 26d5e2e Revert "Add additional check when setting RTCConfiguration" by Magnus Jedvert · 8 years ago
  17. 8110bed Add additional check when setting RTCConfiguration by Steve Anton · 8 years ago
  18. aa41f0c API for periodically regathering ICE candidates by Steve Anton · 8 years ago
  19. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  20. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  21. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  22. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  23. 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 · 8 years ago
  24. 441718e Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  25. 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 · 8 years ago
  26. 3386025 Initialize PeerConnection members in declaration order and destroy them in reverse order. by terelius · 8 years ago
  27. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  28. 1dcb164 Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 8 years ago
  29. 81bf7b0 Pass ownership of candidate to PeerConnection::OnIceCandidate by jbauch · 8 years ago
  30. 42a4263 Making candidate pool size behave as decided in JSEP. by deadbeef · 8 years ago
  31. 7f06766 Delete deprecated PeerConnection methods, and corresponding using declarations. by nisse · 8 years ago
  32. b09b3f9 Add the option to disable IPv6 ICE candidates on WiFi. by zhihuang · 8 years ago
  33. 6dfd53a Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange by zstein · 8 years ago
  34. c1b57a1 Test field trial group with startswith rather than equals. by sprang · 8 years ago
  35. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  36. 6038e97 Adding RTCErrorOr class to be used by ORTC APIs. by deadbeef · 8 years ago
  37. 7798501 Fix the Chrome crash caused by RtcEventLog by zhihuang · 8 years ago
  38. 9dd77ba Clarifying error messages in ParseIceServerUrl for invalid transport parameters. by zstein · 8 years ago
  39. d1f5fda Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration. by skvlad · 8 years ago
  40. 5107246 Allow applications to limit the ICE check rate through RTCConfiguration by skvlad · 8 years ago
  41. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  42. 7ce109a Replace the easy cases of VERIFY usage. by nisse · 8 years ago
  43. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/peerconnection.cc]
  44. e8abe3e Revert of New method StatsObserver::OnCompleteReports, passing ownership. (patchset #2 id:20001 of https://codereview.webrtc.org/2584553002/ ) by nisse · 8 years ago
  45. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  46. eb4ca4e Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  47. 293e926 Reland of: Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  48. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  49. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  50. 0483362 Add disabled certificate check support to IceServer PeerConnection API. by hnsl · 8 years ago
  51. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  52. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  53. 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
  54. 7a5fa6c Adding error output param to SetConfiguration, using new RTCError type. by deadbeef · 8 years ago
  55. fe4a8a4 Implement current/pending session description methods. by deadbeef · 8 years ago
  56. 3061276 Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros. by nisse · 8 years ago
  57. b36ee8d New method StatsObserver::OnCompleteReports, passing ownership. by nisse · 8 years ago
  58. d5236e2 Revert of Add disabled certificate check support to IceServer PeerConnection API. (patchset #8 id:140001 of https://codereview.webrtc.org/2557803002/ ) by magjed · 8 years ago
  59. b78306a Fix segfault when PeerConnection is destroyed during stats collection. by hbos · 8 years ago
  60. b0f04fd Add disabled certificate check support to IceServer PeerConnection API. by hnsl · 8 years ago
  61. 277b250 Refactor "secure bool" into explicit PROTO_TLS. by hnsl · 8 years ago
  62. 6de92f9 Don't allow changing ICE pool size after SetLocalDescription. by deadbeef · 8 years ago
  63. bd44bb0 Fix out of bound reads in ParseIceServerUrl() for various input. by hnsl · 8 years ago
  64. d1a38b5 Implement the "needs-ice-restart" logic for SetConfiguration. by deadbeef · 8 years ago
  65. 3edec7c Adding error enum to be used by PeerConnectionInterface methods. by deadbeef · 8 years ago
  66. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  67. 81c3a03 Added a callback function OnAddTrack to PeerConnectionObserver by zhihuang · 8 years ago
  68. 46c7389 Adding GetConfiguration to PeerConnection. by deadbeef · 8 years ago
  69. 82ebe02 Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed]. by hbos · 8 years ago
  70. e9e94c3 Return false if PeerConnection::GetStats() is called on invalid tracks by zhihuang · 8 years ago
  71. af38847 Make SetLocalDescrption succeed with data-channel only offer and max-bundle policy. by zhihuang · 8 years ago
  72. 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
  73. 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
  74. fc9414a Reland of "Remove the obsolete enum webrtc::PeerConnectionInterface::IceState." by johan · 8 years ago
  75. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  76. d93f50c Add UMA metrics for ICE regathering reasons. by Honghai Zhang · 8 years ago
  77. 74e1a4f PeerConnection[Interface]::GetStats(RTCStatsCollectorCallback*) added. by hbos · 9 years ago
  78. 4cedf2b Add signaling to support ICE renomination. by Honghai Zhang · 9 years ago
  79. bfd398c Add a switch to redetermine role when ICE restarts. by Honghai Zhang · 9 years ago
  80. 68343a8 Revert of Remove the obsolete enum webrtc::PeerConnectionInterface::IceState. (patchset #1 id:1 of https://codereview.webrtc.org/2256663002/ ) by perkj · 9 years ago
  81. 31dea98 Remove the obsolete enum webrtc::PeerConnectionInterface::IceState. by johan · 9 years ago
  82. 9763d56 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 9 years ago
  83. 907abe4 Revert of Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. (patchset #8 id:280001 of https://codereview.webrtc.org/2166873002/ ) by deadbeef · 9 years ago
  84. 34b54c3 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 9 years ago
  85. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 9 years ago
  86. 29ff844 Add PeerConnection IsClosed check. by zhihuang · 9 years ago
  87. 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 · 9 years ago
  88. b9e7b4a Add config to prune low-priority TURN ports for creating connections by Honghai Zhang · 9 years ago
  89. f4e8cf0 Revert of Add config to prune TURN ports (patchset #12 id:360001 of https://codereview.webrtc.org/2093623004/ ) by danilchap · 9 years ago
  90. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 9 years ago
  91. 17aac05 Add config to prune low-priority TURN ports for creating connections by Honghai Zhang · 9 years ago
  92. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  93. f8e6577 Add virtual Initialize methods to PortAllocator and NetworkManager. by Taylor Brandstetter · 9 years ago
  94. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  95. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 9 years ago
  96. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  97. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 9 years ago
  98. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  99. ba8d433 Revert of Add virtual Initialize methods to PortAllocator and NetworkManager. (patchset #4 id:60001 of https://codereview.webrtc.org/2097653002/ ) by deadbeef · 9 years ago
  100. a6bdb09 Add virtual Initialize methods to PortAllocator and NetworkManager. by Taylor Brandstetter · 9 years ago