1. bfd398c Add a switch to redetermine role when ICE restarts. by Honghai Zhang · 8 years ago
  2. 9763d56 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 8 years ago
  3. 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 · 8 years ago
  4. 34b54c3 Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. by zhihuang · 8 years ago
  5. 29ff844 Add PeerConnection IsClosed check. by zhihuang · 8 years ago
  6. f8e6577 Add virtual Initialize methods to PortAllocator and NetworkManager. by Taylor Brandstetter · 9 years ago
  7. 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
  8. a6bdb09 Add virtual Initialize methods to PortAllocator and NetworkManager. by Taylor Brandstetter · 9 years ago
  9. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 9 years ago
  10. d79599d Turning FakeDtlsIdentityStore into FakeRTCCertificateGenerator. by Henrik Boström · 9 years ago
  11. d03c23b Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 9 years ago
  12. 6034705 Add a flag to filter out high-cost networks. by honghaiz · 9 years ago
  13. 98cde26 Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel. by Taylor Brandstetter · 9 years ago
  14. d7973cc Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ ) by hbos · 9 years ago
  15. 400781a Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. by Henrik Boström · 9 years ago
  16. 417eebe Fixing the behavior of the candidate filter with pooled candidates. by Taylor Brandstetter · 9 years ago
  17. e9021a3 Propogate network-worker thread split to api by danilchap · 9 years ago
  18. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  19. a1c3035 Relanding: Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 9 years ago
  20. c55fb30 Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ ) by deadbeef · 9 years ago
  21. 48e9d05 Implement RTCConfiguration.iceCandidatePoolSize. by Taylor Brandstetter · 9 years ago
  22. dc4eb8c Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 9 years ago
  23. 3fe372d Fix all -Wnon-virtual-dtor warnings. by Henrik Kjellander · 9 years ago
  24. 6ab3db2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 9 years ago
  25. 65fc62e Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  26. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 9 years ago
  27. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 9 years ago
  28. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 9 years ago
  29. 7ff1737 Re-enabling tests that were disabled for Windows debug builds. by Taylor Brandstetter · 9 years ago
  30. 71bdda0 Add RTCConfiguration getter and setter methods. The immediate plan is to move some flags into an embedded MediaConfig (https://codereview.webrtc.org/1818033002/), which will be possible after Chrome is updated to use these new setter methods. by Niels Möller · 9 years ago
  31. d45b95c Making new unit test assertions use the standard timeout. by Taylor Brandstetter · 9 years ago
  32. 85e46a8 Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates by Per · 9 years ago
  33. d61bf80 Removed MediaStreamTrackInterface::set_state by perkj · 9 years ago
  34. af510af Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests. by nisse · 9 years ago
  35. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  36. 194e3bc Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 9 years ago
  37. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  38. 102362b Truly disable tests. by Stefan Holmer · 9 years ago
  39. 55d6e7c Disable tests due to issue 5659. by Stefan Holmer · 9 years ago
  40. 2bbff99 Helpers in peer connection unit tests: Use scoped_ptr instead of raw pointers by kwiberg · 9 years ago
  41. aac2dea Changed defaults for CreateAnswer in non-constraint mode by hta · 9 years ago
  42. a3ede6c Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 9 years ago
  43. a2a49d9 This CL provides interfaces that do not use constraints for by hta · 9 years ago
  44. 0db023a Move suspend_below_min_bitrate from VideoOptions to MediaConfig. by nisse · 9 years ago
  45. 0ed85b2 Track pending ICE restarts independently for different media sections. by deadbeef · 9 years ago
  46. 51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago
  47. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
  48. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  49. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/peerconnectioninterface_unittest.cc]
  50. dfb769d Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete by perkj · 9 years ago
  51. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  52. 884f585 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  53. e1f9d83 Adding AddTrack/RemoveTrack to native PeerConnection API. by deadbeef · 9 years ago
  54. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  55. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  56. f475d365 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 9 years ago
  57. 37ebcf0 Reland "Add APK targets to build libjingle tests for Android." by phoglund · 9 years ago
  58. 0c7e9f5 Removing webrtc::PortAllocatorFactoryInterface. by Taylor Brandstetter · 9 years ago
  59. bd7d8f7 Adding a MediaStream parameter to createSender. by deadbeef · 9 years ago
  60. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  61. eb45981 Restoring behavior where PeerConnection tracks changes to MediaStreams. by deadbeef · 9 years ago
  62. bc14164 Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ ) by stefan · 9 years ago
  63. a78c021 Add APK targets to build libjingle_peerconnection_unittests for Android. by perkj · 9 years ago
  64. bda7e0b Fixing issue with default stream upon setting 2nd remote description. by deadbeef · 9 years ago
  65. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  66. 5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
  67. 6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
  68. 653b8e0 Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ ) by deadbeef · 9 years ago
  69. 18a944b Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ ) by deadbeef · 9 years ago
  70. d3b26d9 Adding the ability to change ICE servers through SetConfiguration. by deadbeef · 9 years ago
  71. 8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
  72. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  73. c80741f Fixing some issues with the direction attribute of m-lines in offers. by deadbeef · 9 years ago
  74. 5e97fb5 Don't create remote streams if m-line direction doesn't include "send". by deadbeef · 9 years ago
  75. ab9b2d1 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ ) by deadbeef · 9 years ago
  76. fc648b6 Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ ) by deadbeef · 9 years ago
  77. 97c3929 Moving MediaStreamSignaling logic into PeerConnection. by deadbeef · 9 years ago
  78. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  79. 0a6c4ca Catching more errors when parsing ICE server URLs. by deadbeef · 9 years ago
  80. 5e56c59 DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface). by Henrik Boström · 9 years ago
  81. fabe2c9 Remove deprecated functions. by jbauch · 10 years ago
  82. 61e00b0 Create a in-memory DTLS identity store that keeps a free identity generated in the background. by jiayl@webrtc.org · 10 years ago
  83. be77872 Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background." by jiayl@webrtc.org · 10 years ago
  84. 369f682 Create a in-memory DTLS identity store that keeps a free identity generated in the background. by jiayl@webrtc.org · 10 years ago
  85. 8ad9660 Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background." by jiayl@webrtc.org · 10 years ago
  86. df512cc Create a in-memory DTLS identity store that keeps a free identity generated in the background. by jiayl@webrtc.org · 10 years ago
  87. 005b6ff Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails. by pkasting@chromium.org · 10 years ago
  88. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  89. c2dd5ee Prepare for removal of PeerConnectionObserver::OnError. by perkj@webrtc.org · 10 years ago
  90. 34f2a9e Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 10 years ago
  91. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  92. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  93. 908f57e Disable GetStatsForInvalidTrack while I rewrite it. by tommi@webrtc.org · 11 years ago
  94. 001fd2d Fire OnRenegotiationNeeded only for the first SCTP DataChannel. by jiayl@webrtc.org · 11 years ago
  95. f875f15 (Auto)update libjingle 64709629-> 64813990 by buildbot@webrtc.org · 11 years ago
  96. 61c1b8e (Auto)update libjingle 64585415-> 64594651 by buildbot@webrtc.org · 11 years ago
  97. db41b4d Remove the deprecated GetStats method from PeerConnectionInterface. by jiayl@webrtc.org · 11 years ago
  98. 67ee6b9 Update talk to 60923971 by mallinath@webrtc.org · 11 years ago
  99. a576faf Enable SCTP and use OPENSSL on Anroid and NSS on other platforms. by jiayl@webrtc.org · 11 years ago
  100. 0dac537 Revert 5447 "Update talk to 60420316." by mallinath@webrtc.org · 11 years ago