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gerrit-public.fairphone.software / platform / external / webrtc / bbfcc703ad1b5ae141dcd9c6a951241f6fc03967 / pc / channel_unittest.cc
  1. 365381f Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
  2. e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 7 years ago
  3. 95e7dbb Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" by Zhi Huang · 7 years ago
  4. 27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
  5. 97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
  6. ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
  7. 0807d15 Remove more dead code from BaseChannel by Steve Anton · 7 years ago
  8. cf6e24a Forward the SignalNetworkRouteChanged from DtlsSrtpTransport to BaseChannel. by Zhi Huang · 7 years ago
  9. 0228485 Delete MediaMonitor. by Niels Möller · 7 years ago
  10. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  11. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  12. 3828c06 Replace cricket::ContentAction with webrtc::SdpType by Steve Anton · 7 years ago
  13. cd3fc5d Use the DtlsSrtpTransport in BaseChannel. by Zhi Huang · 7 years ago
  14. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  15. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  16. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  17. 801b868 Remove the CA_UPDATE and related 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. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  22. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  23. 8a63f78 Rewrite the remaining few WebRtcSession tests. by Steve Anton · 7 years ago
  24. 8b35df7 Try re-enabling VoiceChannel::TestInit. by Kári Tristan Helgason · 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_unittest.cc]
  29. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
  30. 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  31. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  32. 05b07bb Fix places that trigger no-unused-lambda-capture - change to using static-constexpr. by eladalon · 7 years ago
  33. 1cc5fc3 Fix places that trigger no-unused-lambda-capture by eladalon · 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. af99b6d Delete SignalSrtpError. by nisse · 7 years ago
  39. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  40. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  41. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
  42. 7c85658 Roll chromium_revision 33a7a547b9..0e44c5e141 (452838:453130) by kjellander · 8 years ago
  43. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
  44. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
  45. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  46. 1b54a5f Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  47. f33491e Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ ) by deadbeef · 8 years ago
  48. eaa826c Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  49. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  50. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  51. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  52. c8ee882 Replace use of ASSERT in test code. by nisse · 8 years ago
  53. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  54. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  55. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  56. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  57. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  58. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  59. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  60. 49f34fd Relanding: Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  61. 57fd726 Revert of Refactoring that removes P2PTransport and DtlsTransport classes. (patchset #9 id:150001 of https://codereview.webrtc.org/2517883002/ ) by deadbeef · 8 years ago
  62. bd28681 Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  63. c6b6e09 Relaxing timeouts for TestMediaMonitor. by deadbeef · 8 years ago
  64. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  65. 74097fd Delete unused file screencastid.h. by nisse · 8 years ago
  66. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  67. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
  68. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  69. 8853289 Un-flaking TestSrtpError by using a fake clock. by Taylor Brandstetter · 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. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 8 years ago
  77. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  78. dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 8 years ago
  79. 7f216b7 Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 8 years ago
  80. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  81. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  82. 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
  83. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  84. e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  85. 52dce73 Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  86. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  87. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 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 · 9 years ago
  89. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  90. 292d658 Fix for intermittent tsan2 errors from SendRtpToRtpOnThread and SendSrtpToSrtpOnThread. by ossu · 9 years ago
  91. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  92. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  93. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 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 (99%) from talk/session/media/channel_unittest.cc]
  96. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  97. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  98. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  99. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  100. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago