1. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
  2. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  3. 8853289 Un-flaking TestSrtpError by using a fake clock. by Taylor Brandstetter · 8 years ago
  4. 6bb1ef2 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 9 years ago
  5. 059e183 Reland of "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #1 id:1 of https://codereview.webrtc.org/2098703004/ ) by honghaiz · 9 years ago
  6. ae4d0d9 Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #5 id:120001 of https://codereview.webrtc.org/2041593002/ ) by honghaiz · 9 years ago
  7. 5b5d2cd Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )" by Honghai Zhang · 9 years ago
  8. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  9. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  10. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 9 years ago
  11. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  12. dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 9 years ago
  13. 7f216b7 Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 9 years ago
  14. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
  15. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  16. 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
  17. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  18. e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  19. 52dce73f Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  20. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  21. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  22. 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
  23. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  24. 292d658 Fix for intermittent tsan2 errors from SendRtpToRtpOnThread and SendSrtpToSrtpOnThread. by ossu · 9 years ago
  25. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  26. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  27. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 9 years ago
  28. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  29. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/channel_unittest.cc]
  30. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  31. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  32. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  33. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  34. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  35. 482b12e Remove BundleFilter filtering of RTCP. by pbos · 9 years ago
  36. 5237aaf Convert usage of ARRAY_SIZE to arraysize. by tfarina · 9 years ago
  37. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  38. 4bac9c5 Change SetOutputScaling to set a single level, not left/right levels. by solenberg · 9 years ago
  39. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  40. 5629a1d Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004. by solenberg · 9 years ago
  41. 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 9 years ago
  42. dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 9 years ago
  43. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  44. 27dc29b Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) by guoweis · 9 years ago
  45. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  46. 34fbfff Remove VideoMediaChannel::SetRender(). by Peter Boström · 9 years ago
  47. cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 9 years ago
  48. a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 9 years ago
  49. 47ee2f3 TransportController refactoring. by deadbeef · 9 years ago
  50. 22011c1 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle). by solenberg · 9 years ago
  51. 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 9 years ago
  52. 9af63f4 TransportController refactoring. by deadbeef · 9 years ago
  53. b071a19 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters. by Fredrik Solenberg · 9 years ago
  54. 1dd98f3 - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel) by solenberg · 9 years ago
  55. 66f4339 Remove [Voice|Video]MediaChannel::GetOptions(). by solenberg · 9 years ago
  56. 8006f07 Remove unused TypingMonitor class. by solenberg · 9 years ago
  57. d828198 Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer. by Henrik Boström · 9 years ago
  58. b6d4ec4 Support generation of EC keys using P256 curve and support ECDSA certs. by Torbjorn Granlund · 9 years ago
  59. 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 9 years ago
  60. a9b4c32 Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity. by Peter Thatcher · 9 years ago
  61. a6d2444 Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code. by Peter Thatcher · 10 years ago
  62. 3b1e647b6 Remove media sinks from Channel. by pbos · 10 years ago
  63. af55ccc Add RtcpMuxPolicy support to PeerConnection. by Peter Thatcher · 10 years ago
  64. 7fb711f Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class. by Fredrik Solenberg · 10 years ago
  65. 0e81fdf Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. by pkasting@chromium.org · 10 years ago
  66. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  67. aacc234 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  68. 4cb3856 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
  69. 536f999 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  70. f050791 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." by pthatcher@webrtc.org · 10 years ago
  71. 4afb599 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  72. e2b7585 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. by pthatcher@webrtc.org · 10 years ago
  73. 18a3896 Revert r7886:7887. by pbos@webrtc.org · 10 years ago
  74. dee76f3 Move the obvious/easy Jingle-specific code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  75. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  76. 28100cb Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." by henrike@webrtc.org · 10 years ago
  77. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  78. 81ddc78 (Auto)update libjingle 77701902-> 77709729 by buildbot@webrtc.org · 10 years ago
  79. 34f2a9e Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 10 years ago
  80. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  81. 65b98d1 (Auto)update libjingle 72839629-> 72847605 by buildbot@webrtc.org · 10 years ago
  82. 5b1ebac (Auto)update libjingle 72820109-> 72822008 by buildbot@webrtc.org · 10 years ago
  83. d509678 (Auto)update libjingle 72819313-> 72820109 by buildbot@webrtc.org · 10 years ago
  84. 94b996c (Auto)update libjingle 72785516-> 72819313 by buildbot@webrtc.org · 10 years ago
  85. 476efa2 (Auto)update libjingle 72785180-> 72785516 by buildbot@webrtc.org · 10 years ago
  86. e0d03f1 (Auto)update libjingle 72443101-> 72446860 by buildbot@webrtc.org · 10 years ago
  87. 6e203d5 (Auto)update libjingle 72442050-> 72443101 by buildbot@webrtc.org · 10 years ago
  88. 52148c2 (Auto)update libjingle 72430895-> 72442050 by buildbot@webrtc.org · 10 years ago
  89. 7cb60cc (Auto)update libjingle 72407428-> 72430895 by buildbot@webrtc.org · 10 years ago
  90. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  91. b5348c6 Minor refactoring of the session classes. by tommi@webrtc.org · 11 years ago
  92. 910473b Fix C++11 -Wnarrowing in channel_unittest.cc. by pbos@webrtc.org · 11 years ago
  93. 6bfd619 (Auto)update libjingle 67052073-> 67134648 by buildbot@webrtc.org · 11 years ago
  94. 3e92468 (Auto)update libjingle 67037200-> 67043374 by buildbot@webrtc.org · 11 years ago
  95. 5ee0f05 (Auto)update libjingle 66138442-> 66236292 by buildbot@webrtc.org · 11 years ago
  96. f5bebd4 (Auto)update libjingle 64247466-> 64326665 by henrike@webrtc.org · 11 years ago
  97. 4b26e2e Update libjingle to 59676287 by sergeyu@chromium.org · 11 years ago
  98. a989080 Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  99. 2018269 Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  100. a129b6c Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago