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gerrit-public.fairphone.software / platform / external / webrtc / c2a0eb2699f7871c876a487105070f84c38a3dd0 / pc / channel.cc
  1. 04eaa15 Change the flag when RtpTransport objects send packet. by Zhi Huang · 7 years ago
  2. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  3. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  4. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  5. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channel.cc]
  6. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
  7. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  8. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 8 years ago
  9. 634977b SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 8 years ago
  10. e8ab543 Make BaseChannel::rtp_transport_ a unique_ptr. by zstein · 8 years ago
  11. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  12. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  13. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  14. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  15. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  16. 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 · 8 years ago
  17. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  18. 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 · 8 years ago
  19. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 8 years ago
  20. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  21. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
  22. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  23. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  24. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  25. d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 8 years ago
  26. dfcab72 Reland: Improve testing of SRTP external auth code paths. by jbauch · 8 years ago
  27. d81f121 Revert of Improve testing of SRTP external auth code paths. (patchset #2 id:20001 of https://codereview.webrtc.org/2722423003/ ) by jbauch · 8 years ago
  28. ac170d5 Improve testing of SRTP external auth code paths. by jbauch · 8 years ago
  29. d48f488 Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. by jbauch · 8 years ago
  30. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  31. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
  32. 7ce109a Replace the easy cases of VERIFY usage. by nisse · 8 years ago
  33. 7d25426 Delete unneeded includes of base/common.h. by nisse · 8 years ago
  34. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  35. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  36. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  37. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  38. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  39. 8e814d7 Provide better message for when RTCP mux "require" policy is triggered. by deadbeef · 8 years ago
  40. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  41. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  42. eb4ca4e Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  43. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  44. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  45. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  46. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  47. 8f425f9 Relaxing DCHECK for packets sent before SRTP is enabled. by deadbeef · 8 years ago
  48. 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
  49. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  50. 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
  51. 45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  52. d89ab14 Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit. by johan · 8 years ago
  53. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  54. 062ce9f Combining "SetTransportChannel" and "SetRtcpTransportChannel". by deadbeef · 8 years ago
  55. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
  56. 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 8 years ago
  57. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  58. c309e0e Don't stop sending media on EWOULDBLOCK by skvlad · 8 years ago
  59. 6bb1ef2 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 9 years ago
  60. 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
  61. 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
  62. 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
  63. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  64. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  65. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  66. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 9 years ago
  67. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  68. dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 9 years ago
  69. 7f216b7 Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 9 years ago
  70. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
  71. 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
  72. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  73. e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  74. 52dce73 Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  75. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  76. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  77. 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
  78. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  79. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  80. 0510331 Drop VideoOptions from VideoSendParameters. by nisse · 9 years ago
  81. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  82. ca8b404 Add tracing to interesting media-related methods. by Peter Boström · 9 years ago
  83. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  84. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 9 years ago
  85. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  86. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  87. 686a8ef Replace scoped_ptr with unique_ptr in webrtc/media/ by kwiberg · 9 years ago
  88. 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
  89. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  90. 4b4dc86 Remove conference_mode flag from AudioOptions and VideoOptions. by nisse · 9 years ago
  91. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  92. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/channel.cc]
  93. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  94. 08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 9 years ago
  95. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  96. a6c39d9 Remove unimplemented VideoChannel code. by Peter Boström · 9 years ago
  97. 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
  98. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  99. e6bf587 Deleted VideoCapturer::screencast_max_pixels, together with by nisse · 9 years ago
  100. 4638331 DTLS-SRTP set up is bypassed when the channel has been writable. by guoweis · 9 years ago