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gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / pc / channel.cc
  1. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  2. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  3. 1d88d74 Remove the unused code. by Zhi Huang · 7 years ago
  4. 801b868 Remove the CA_UPDATE and related code. by Zhi Huang · 7 years ago
  5. 5f5918f Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks. by Zhi Huang · 7 years ago
  6. 942bc2e Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  7. 8c316c1 Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal." by Zhi Huang · 7 years ago
  8. 7167745 Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  9. c99b6c7 Remove the SetEncryptedHeaderExtensionIds methods. by Zhi Huang · 7 years ago
  10. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  11. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  12. 04eaa15 Change the flag when RtpTransport objects send packet. by Zhi Huang · 7 years ago
  13. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  14. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  15. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  16. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channel.cc]
  17. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
  18. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  19. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
  20. 634977b SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 7 years ago
  21. e8ab543 Make BaseChannel::rtp_transport_ a unique_ptr. by zstein · 7 years ago
  22. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  23. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  24. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  25. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  26. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  27. 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
  28. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  29. 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
  30. 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
  31. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  32. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  33. 8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 7 years ago
  34. fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 7 years ago
  35. 292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 7 years ago
  36. d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 7 years ago
  37. dfcab72 Reland: Improve testing of SRTP external auth code paths. by jbauch · 7 years ago
  38. d81f121 Revert of Improve testing of SRTP external auth code paths. (patchset #2 id:20001 of https://codereview.webrtc.org/2722423003/ ) by jbauch · 7 years ago
  39. ac170d5 Improve testing of SRTP external auth code paths. by jbauch · 7 years ago
  40. d48f488 Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. by jbauch · 7 years ago
  41. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 7 years ago
  42. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 7 years ago
  43. 7ce109a Replace the easy cases of VERIFY usage. by nisse · 7 years ago
  44. 7d25426 Delete unneeded includes of base/common.h. by nisse · 7 years ago
  45. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 7 years ago
  46. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  47. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  48. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  49. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  50. 8e814d7 Provide better message for when RTCP mux "require" policy is triggered. by deadbeef · 8 years ago
  51. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  52. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  53. eb4ca4e Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  54. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  55. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  56. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  57. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  58. 8f425f9 Relaxing DCHECK for packets sent before SRTP is enabled. by deadbeef · 8 years ago
  59. 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
  60. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  61. 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
  62. 45c8b89 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  63. d89ab14 Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit. by johan · 8 years ago
  64. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 8 years ago
  65. 062ce9f Combining "SetTransportChannel" and "SetRtcpTransportChannel". by deadbeef · 8 years ago
  66. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
  67. 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 8 years ago
  68. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  69. c309e0e Don't stop sending media on EWOULDBLOCK by skvlad · 8 years ago
  70. 6bb1ef2 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 8 years ago
  71. 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 · 8 years ago
  72. 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 · 8 years ago
  73. 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 · 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. 7f216b7 Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 8 years ago
  81. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  82. 0e533ef Update the call when the network route changes by Honghai Zhang · 8 years ago
  83. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 8 years ago
  84. e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 8 years ago
  85. 52dce73 Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 8 years ago
  86. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 8 years ago
  87. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  88. 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
  89. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 8 years ago
  90. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
  91. 0510331 Drop VideoOptions from VideoSendParameters. by nisse · 8 years ago
  92. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 8 years ago
  93. ca8b404 Add tracing to interesting media-related methods. by Peter Boström · 8 years ago
  94. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 8 years ago
  95. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 8 years ago
  96. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 8 years ago
  97. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 8 years ago
  98. 686a8ef Replace scoped_ptr with unique_ptr in webrtc/media/ by kwiberg · 8 years ago
  99. 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 · 8 years ago
  100. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 8 years ago