Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / dde8702f11f3ba96c935c70354ae96eb80be2a44 / pc / channel_unittest.cc
  1. 0228485 Delete MediaMonitor. by Niels Möller · 7 years ago
  2. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  3. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  4. 3828c06 Replace cricket::ContentAction with webrtc::SdpType by Steve Anton · 7 years ago
  5. cd3fc5d Use the DtlsSrtpTransport in BaseChannel. by Zhi Huang · 7 years ago
  6. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  7. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  8. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  9. 801b868 Remove the CA_UPDATE and related code. by Zhi Huang · 7 years ago
  10. 942bc2e Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  11. 8c316c1 Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal." by Zhi Huang · 7 years ago
  12. 7167745 Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  13. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  14. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  15. 8a63f78 Rewrite the remaining few WebRtcSession tests. by Steve Anton · 7 years ago
  16. 8b35df7 Try re-enabling VoiceChannel::TestInit. by Kári Tristan Helgason · 7 years ago
  17. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  18. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  19. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  20. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channel_unittest.cc]
  21. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
  22. 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
  23. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  24. 05b07bb Fix places that trigger no-unused-lambda-capture - change to using static-constexpr. by eladalon · 7 years ago
  25. 1cc5fc3 Fix places that trigger no-unused-lambda-capture by eladalon · 7 years ago
  26. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  27. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  28. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  29. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  30. af99b6d Delete SignalSrtpError. by nisse · 7 years ago
  31. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  32. 56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  33. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 7 years ago
  34. 7c85658 Roll chromium_revision 33a7a547b9..0e44c5e141 (452838:453130) by kjellander · 8 years ago
  35. 5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
  36. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
  37. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  38. 1b54a5f Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  39. 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
  40. eaa826c Removing #defines previously used for building without BoringSSL/OpenSSL. by deadbeef · 8 years ago
  41. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  42. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  43. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  44. c8ee882 Replace use of ASSERT in test code. by nisse · 8 years ago
  45. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  46. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  47. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  48. 953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  49. c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
  50. 67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
  51. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  52. 49f34fd Relanding: Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  53. 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
  54. bd28681 Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  55. c6b6e09 Relaxing timeouts for TestMediaMonitor. by deadbeef · 8 years ago
  56. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  57. 74097fd Delete unused file screencastid.h. by nisse · 8 years ago
  58. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  59. bad33bf Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 8 years ago
  60. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  61. 8853289 Un-flaking TestSrtpError by using a fake clock. by Taylor Brandstetter · 8 years ago
  62. 6bb1ef2 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 8 years ago
  63. 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
  64. 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
  65. 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
  66. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 8 years ago
  67. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  68. 6c87a67 Do not create a temporary transport channel when using max-bundle by skvlad · 8 years ago
  69. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  70. dae07ba Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 8 years ago
  71. 7f216b7 Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 8 years ago
  72. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  73. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 8 years ago
  74. 0e533ef Update the call when the network route changes by Honghai Zhang · 8 years ago
  75. 67cf2c1 Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  76. e0d4637 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  77. 52dce73 Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  78. cc411c0 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  79. eec21bd Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  80. 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
  81. 944c390 Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  82. 292d658 Fix for intermittent tsan2 errors from SendRtpToRtpOnThread and SendSrtpToSrtpOnThread. by ossu · 9 years ago
  83. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  84. 3102294 Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  85. c11b184 Remove CaptureManager and related calls in ChannelManager. by perkj · 9 years ago
  86. 65c7f67 Fix license headers in webrtc/pc by kjellander · 9 years ago
  87. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago[Renamed (99%) from talk/session/media/channel_unittest.cc]
  88. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  89. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 9 years ago
  90. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  91. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  92. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  93. 482b12e Remove BundleFilter filtering of RTCP. by pbos · 9 years ago
  94. 5237aaf Convert usage of ARRAY_SIZE to arraysize. by tfarina · 9 years ago
  95. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  96. 4bac9c5 Change SetOutputScaling to set a single level, not left/right levels. by solenberg · 9 years ago
  97. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  98. 5629a1d Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004. by solenberg · 9 years ago
  99. 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 9 years ago
  100. dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 9 years ago