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gerrit-public.fairphone.software / platform / external / webrtc / 8699a3229f63455a4f6ed04330742f0a0175bb31 / pc / channel.h
  1. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  2. 6b63cd5 Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests by Steve Anton · 7 years ago
  3. b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 7 years ago
  4. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  5. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  6. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  7. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channel.h]
  8. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
  9. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  10. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
  11. 634977b SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 7 years ago
  12. e8ab543 Make BaseChannel::rtp_transport_ a unique_ptr. by zstein · 7 years ago
  13. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  14. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  15. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  16. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  17. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  18. f79ade1 Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 7 years ago
  19. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  20. d72098a Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 7 years ago
  21. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 7 years ago
  22. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  23. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  24. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  25. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  26. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  27. d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 8 years ago
  28. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
  29. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  30. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  31. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  32. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  33. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  34. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  35. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  36. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  37. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  38. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  39. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  40. acd935b Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ ) by nisse · 8 years ago
  41. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  42. 7341ab8 Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ ) by nisse · 8 years ago
  43. 45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  44. d89ab14 Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit. by johan · 8 years ago
  45. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  46. 062ce9f Combining "SetTransportChannel" and "SetRtcpTransportChannel". by deadbeef · 8 years ago
  47. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
  48. 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 8 years ago
  49. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  50. 6bb1ef2 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 8 years ago
  51. 184a3fd Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 8 years ago
  52. 6379793 Removing obsolete method from channel.h. by deadbeef · 8 years ago
  53. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 8 years ago
  54. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  55. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  56. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 8 years ago
  57. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  58. dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 8 years ago
  59. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  60. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  61. 52dce73 Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  62. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  63. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  64. 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
  65. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  66. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  67. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  68. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  69. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 9 years ago
  70. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  71. 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
  72. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  73. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  74. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (98%) from talk/session/media/channel.h]
  75. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  76. 08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 9 years ago
  77. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  78. a6c39d9 Remove unimplemented VideoChannel code. by Peter Boström · 9 years ago
  79. 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
  80. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  81. e6bf587 Deleted VideoCapturer::screencast_max_pixels, together with by nisse · 9 years ago
  82. 4638331 DTLS-SRTP set up is bypassed when the channel has been writable. by guoweis · 9 years ago
  83. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  84. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  85. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  86. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  87. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  88. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  89. 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
  90. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  91. ec9d187 Added override keyword to overridden methods to stop compiler warnings. by rlester · 9 years ago
  92. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  93. d4cec0d Remove MediaChannel::SetRemoteRenderer(). by solenberg · 9 years ago
  94. 4bac9c5 Change SetOutputScaling to set a single level, not left/right levels. by solenberg · 9 years ago
  95. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  96. 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 9 years ago
  97. dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 9 years ago
  98. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  99. 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
  100. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago