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gerrit-public.fairphone.software / platform / external / webrtc / 4e140da62fb15e22aae9bb9a16682f1c01128dc5 / pc / rtpsenderreceiver_unittest.cc
  1. e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 7 years ago
  2. 13b8bad Final name changing of MediaStreamInterface.label() to id(). by Seth Hampson · 7 years ago
  3. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
  4. 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
  5. 57858b3 Reland "Update RTCStatsCollector to work with RtpTransceivers" by Steve Anton · 7 years ago
  6. ee2388f Revert "Update RTCStatsCollector to work with RtpTransceivers" by Guido Urdaneta · 7 years ago
  7. 56bae8d Update RTCStatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
  8. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
  9. d367921 Configure media flow correctly with Unified Plan by Steve Anton · 7 years ago
  10. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  11. 02ee47c Signal track ID correctly when Unified Plan semantics selected by Steve Anton · 7 years ago
  12. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  13. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  14. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  15. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  16. 9e6fd2b Add streams() to RtpReceiverInterface and implementations. by Henrik Boström · 7 years ago
  17. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  18. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  19. c9e1560 Modernize and cleanup ChannelManager by Steve Anton · 7 years ago
  20. b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 7 years ago
  21. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  22. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  23. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/rtpsenderreceiver_unittest.cc]
  24. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  25. 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 8 years ago
  26. 539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 8 years ago
  27. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 8 years ago
  28. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  29. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  30. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  31. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  32. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  33. 1a2183d Removing unnecessary parameters from CreateXChannel methods. by deadbeef · 8 years ago
  34. 757146b Remove PC factory options param from LocalAudioSource::Create. by deadbeef · 8 years ago
  35. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  36. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
  37. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  38. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (98%) from webrtc/api/rtpsenderreceiver_unittest.cc]
  39. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  40. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  41. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  42. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  43. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  44. 5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
  45. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  46. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  47. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  48. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  49. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  50. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 9 years ago
  51. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  52. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 9 years ago
  53. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  54. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  55. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  56. 5dd42fd Fixing a segfault that can occur when changing the track of an RtpSender. by deadbeef · 9 years ago
  57. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 9 years ago
  58. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 9 years ago
  59. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  60. d61bf80 Removed MediaStreamTrackInterface::set_state by perkj · 9 years ago
  61. af510af Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests. by nisse · 9 years ago
  62. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  63. f0dcfe2 Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource. by perkj · 9 years ago
  64. 0d3eef2 Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it. by perkj · 9 years ago
  65. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  66. a3ede6c Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 9 years ago
  67. f2880a0 Change webrtc::VideoSourceInterface to inherit rtc::VideoSourceInterface. by perkj · 9 years ago
  68. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  69. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (97%) from talk/app/webrtc/rtpsenderreceiver_unittest.cc]
  70. 8e8908a Delete FrameInput method and FrameInputWrapper class. by nisse · 9 years ago
  71. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  72. 08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 9 years ago
  73. e73afba New rtc::VideoSinkInterface. by nisse · 9 years ago
  74. 2098fca Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) by nisse · 9 years ago
  75. a862d45 New rtc::VideoSinkInterface. by Niels Möller · 9 years ago
  76. 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
  77. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  78. 6eca7e3 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( by tommi · 9 years ago
  79. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  80. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  81. 5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
  82. 6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
  83. 8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
  84. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  85. d4cec0d Remove MediaChannel::SetRemoteRenderer(). by solenberg · 9 years ago
  86. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  87. 70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 9 years ago