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webrtc / src.git / 33b01f2162af32457e0a6a6e0eb670254701df61 / webrtc / pc / channel.cc
  1. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
  2. 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
  3. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  4. e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  5. 52dce73f Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  6. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  7. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  8. 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
  9. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  10. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  11. 0510331 Drop VideoOptions from VideoSendParameters. by nisse · 9 years ago
  12. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  13. ca8b404 Add tracing to interesting media-related methods. by Peter Boström · 9 years ago
  14. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  15. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 9 years ago
  16. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  17. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  18. 686a8ef Replace scoped_ptr with unique_ptr in webrtc/media/ by kwiberg · 9 years ago
  19. 7324eb9 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago
  20. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  21. 4b4dc86 Remove conference_mode flag from AudioOptions and VideoOptions. by nisse · 9 years ago
  22. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  23. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/channel.cc]
  24. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  25. 08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 9 years ago
  26. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  27. a6c39d9 Remove unimplemented VideoChannel code. by Peter Boström · 9 years ago
  28. 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
  29. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  30. e6bf587 Deleted VideoCapturer::screencast_max_pixels, together with by nisse · 9 years ago
  31. 4638331 DTLS-SRTP set up is bypassed when the channel has been writable. by guoweis · 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. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  35. 9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
  36. 6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  37. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  38. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  39. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  40. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  41. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  42. 318166b Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) by guoweis · 9 years ago
  43. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  44. 482b12e Remove BundleFilter filtering of RTCP. by pbos · 9 years ago
  45. be57983 Rename Maybe to Optional by Karl Wiberg · 9 years ago
  46. 102c6a6 Replace rtc::cricket::Settable with rtc::Maybe by kwiberg · 9 years ago
  47. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  48. 1ac5614 Remove default receive channel from WVoE; baby step 3. by solenberg · 9 years ago
  49. d4cec0d Remove MediaChannel::SetRemoteRenderer(). by solenberg · 9 years ago
  50. 4bac9c5 Change SetOutputScaling to set a single level, not left/right levels. by solenberg · 9 years ago
  51. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  52. 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 10 years ago
  53. dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 10 years ago
  54. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 10 years ago
  55. 27dc29b Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) by guoweis · 10 years ago
  56. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 10 years ago
  57. 34fbfff Remove VideoMediaChannel::SetRender(). by Peter Boström · 10 years ago
  58. cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 10 years ago
  59. a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 10 years ago
  60. 47ee2f3 TransportController refactoring. by deadbeef · 10 years ago
  61. 22011c1 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle). by solenberg · 10 years ago
  62. 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 10 years ago
  63. 9af63f4 TransportController refactoring. by deadbeef · 10 years ago
  64. 1dd98f3 - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel) by solenberg · 10 years ago
  65. 8006f07 Remove unused TypingMonitor class. by solenberg · 10 years ago
  66. bfab5cb Fix some minor errors with the voice engine caused by the refactor CL https://codereview.webrtc.org/1229283003/. by Peter Thatcher · 10 years ago
  67. dbe5bd9 Delete unused function SetSessionError. by Nico Weber · 10 years ago
  68. c2ee2c8 Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well. by Peter Thatcher · 10 years ago
  69. 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 10 years ago
  70. a9b4c32 Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity. by Peter Thatcher · 10 years ago
  71. a6d2444 Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code. by Peter Thatcher · 10 years ago
  72. 3b1e647b6 Remove media sinks from Channel. by pbos · 10 years ago
  73. af55ccc Add RtcpMuxPolicy support to PeerConnection. by Peter Thatcher · 10 years ago
  74. c56ac1e rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
  75. cbf0927 Revert "rtc::Buffer: Remove backwards compatibility band-aids" by Karl Wiberg · 10 years ago
  76. 9e1a6d7 rtc::Buffer: Remove backwards compatibility band-aids by Karl Wiberg · 10 years ago
  77. 7fb711f Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class. by Fredrik Solenberg · 10 years ago
  78. 7c027b6 Enable more Clang warnings for talk/ by Henrik Kjellander · 10 years ago
  79. 9478437 rtc::Buffer improvements by Karl Wiberg · 10 years ago
  80. eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 10 years ago
  81. 592470b Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession. by pthatcher@webrtc.org · 10 years ago
  82. 6ad507a Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE. by pthatcher@webrtc.org · 10 years ago
  83. 4eeef58 Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession. by pthatcher@webrtc.org · 10 years ago
  84. b4aac13 Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well. by pthatcher@webrtc.org · 10 years ago
  85. 058b1f1 Remove GetReceiveBandwidthEstimatorStats. by pbos@webrtc.org · 10 years ago
  86. a67ca1a Only report the first rtp packet because it indicates the media has started flowing. by honghaiz@google.com · 10 years ago
  87. 586f2ed Change GetStreamBySsrc to not copy StreamParams. by tommi@webrtc.org · 10 years ago
  88. 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
  89. 18a3896 Revert r7886:7887. by pbos@webrtc.org · 10 years ago
  90. dee76f3 Move the obvious/easy Jingle-specific code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
  91. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  92. 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
  93. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago
  94. 65b98d1 (Auto)update libjingle 72839629-> 72847605 by buildbot@webrtc.org · 11 years ago
  95. 5b1ebac (Auto)update libjingle 72820109-> 72822008 by buildbot@webrtc.org · 11 years ago
  96. d509678 (Auto)update libjingle 72819313-> 72820109 by buildbot@webrtc.org · 11 years ago
  97. 94b996c (Auto)update libjingle 72785516-> 72819313 by buildbot@webrtc.org · 11 years ago
  98. 476efa2 (Auto)update libjingle 72785180-> 72785516 by buildbot@webrtc.org · 11 years ago
  99. e0d03f1 (Auto)update libjingle 72443101-> 72446860 by buildbot@webrtc.org · 11 years ago
  100. 6e203d5 (Auto)update libjingle 72442050-> 72443101 by buildbot@webrtc.org · 11 years ago