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gerrit-public.fairphone.software / platform / external / webrtc / 2ae140ae7e34bd1e102ca832a14b7b0adbb72d15 / pc / channel.h
  1. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  2. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
  3. e2a9318 Delete ConnectionMonitor. by Niels Möller · 7 years ago
  4. 0228485 Delete MediaMonitor. by Niels Möller · 7 years ago
  5. 053c1f8 Delete unused signal VoiceChannel::SignalAudioMonitor. by Niels Möller · 7 years ago
  6. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  7. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  8. dc8b5ab Remove dead code for media channel errors by Steve Anton · 7 years ago
  9. 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
  10. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  11. 593e325 Change RTCStatsCollector to only access channels from signaling thread by Steve Anton · 7 years ago
  12. 9a44f96 Delete rtc_base/window.h. by Niels Möller · 7 years ago
  13. 3828c06 Replace cricket::ContentAction with webrtc::SdpType by Steve Anton · 7 years ago
  14. 2dfc42d Prepare to make BaseChannel depend on RtpTransportInternal only. by Zhi Huang · 7 years ago
  15. cd3fc5d Use the DtlsSrtpTransport in BaseChannel. by Zhi Huang · 7 years ago
  16. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  17. 1d88d74 Remove the unused code. by Zhi Huang · 7 years ago
  18. 942bc2e Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  19. 8c316c1 Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal." by Zhi Huang · 7 years ago
  20. 7167745 Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  21. c99b6c7 Remove the SetEncryptedHeaderExtensionIds methods. by Zhi Huang · 7 years ago
  22. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  23. 6b63cd5 Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests by Steve Anton · 7 years ago
  24. b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 7 years ago
  25. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  26. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  27. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  28. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channel.h]
  29. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
  30. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  31. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
  32. 634977b SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 7 years ago
  33. e8ab543 Make BaseChannel::rtp_transport_ a unique_ptr. by zstein · 7 years ago
  34. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  35. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  36. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  37. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  38. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  39. 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
  40. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  41. 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
  42. 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
  43. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  44. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  45. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 7 years ago
  46. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 7 years ago
  47. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 7 years ago
  48. d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 7 years ago
  49. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
  50. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  51. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  52. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  53. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  54. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  55. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  56. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  57. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  58. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  59. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  60. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  61. 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
  62. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  63. 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
  64. 45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  65. d89ab14 Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit. by johan · 8 years ago
  66. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  67. 062ce9f Combining "SetTransportChannel" and "SetRtcpTransportChannel". by deadbeef · 8 years ago
  68. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
  69. 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 8 years ago
  70. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  71. 6bb1ef2 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 8 years ago
  72. 184a3fd Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 8 years ago
  73. 6379793 Removing obsolete method from channel.h. by deadbeef · 8 years ago
  74. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 8 years ago
  75. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  76. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  77. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 8 years ago
  78. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  79. dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 8 years ago
  80. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  81. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 8 years ago
  82. 52dce73 Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 8 years ago
  83. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 8 years ago
  84. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  85. 194e3bc Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 8 years ago
  86. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  87. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
  88. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 8 years ago
  89. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 8 years ago
  90. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 8 years ago
  91. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  92. 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
  93. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  94. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  95. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (98%) from talk/session/media/channel.h]
  96. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  97. 08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 9 years ago
  98. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  99. a6c39d9 Remove unimplemented VideoChannel code. by Peter Boström · 9 years ago
  100. 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