Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / pc / rtpsenderreceiver_unittest.cc
  1. 9e6fd2b Add streams() to RtpReceiverInterface and implementations. by Henrik Boström · 7 years ago
  2. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  3. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  4. c9e1560 Modernize and cleanup ChannelManager by Steve Anton · 7 years ago
  5. b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 7 years ago
  6. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  7. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  8. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/rtpsenderreceiver_unittest.cc]
  9. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  10. 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 7 years ago
  11. 539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 7 years ago
  12. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 7 years ago
  13. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  14. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  15. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  16. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 7 years ago
  17. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 7 years ago
  18. 1a2183d Removing unnecessary parameters from CreateXChannel methods. by deadbeef · 7 years ago
  19. 757146b Remove PC factory options param from LocalAudioSource::Create. by deadbeef · 7 years ago
  20. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 7 years ago
  21. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 7 years ago
  22. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 7 years ago
  23. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (98%) from webrtc/api/rtpsenderreceiver_unittest.cc]
  24. b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
  25. 6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
  26. 5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
  27. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  28. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  29. 5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
  30. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  31. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  32. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  33. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  34. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  35. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 8 years ago
  36. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  37. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 8 years ago
  38. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  39. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  40. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  41. 5dd42fd Fixing a segfault that can occur when changing the track of an RtpSender. by deadbeef · 8 years ago
  42. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 8 years ago
  43. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 8 years ago
  44. 2ded9b1 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 8 years ago
  45. d61bf80 Removed MediaStreamTrackInterface::set_state by perkj · 8 years ago
  46. af510af Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests. by nisse · 8 years ago
  47. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
  48. f0dcfe2 Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource. by perkj · 8 years ago
  49. 0d3eef2 Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it. by perkj · 8 years ago
  50. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 8 years ago
  51. a3ede6c Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 8 years ago
  52. f2880a0 Change webrtc::VideoSourceInterface to inherit rtc::VideoSourceInterface. by perkj · 8 years ago
  53. b24317b Fix license headers in webrtc/api. by kjellander · 8 years ago
  54. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 8 years ago[Renamed (97%) from talk/app/webrtc/rtpsenderreceiver_unittest.cc]
  55. 8e8908a Delete FrameInput method and FrameInputWrapper class. by nisse · 8 years ago
  56. a96e2d7 Move talk/media to webrtc/media by kjellander · 8 years ago
  57. 08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 8 years ago
  58. e73afba New rtc::VideoSinkInterface. by nisse · 8 years ago
  59. 2098fca Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ ) by nisse · 8 years ago
  60. a862d45 New rtc::VideoSinkInterface. by Niels Möller · 8 years ago
  61. 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
  62. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  63. 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
  64. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  65. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  66. 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
  67. 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
  68. 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
  69. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  70. d4cec0d Remove MediaChannel::SetRemoteRenderer(). by solenberg · 9 years ago
  71. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  72. 70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 9 years ago