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gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / api / rtpsender.h
  1. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  2. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/api/rtpsender.h]
  3. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  4. 5214a0a Add support for content hints to VideoTrack. by pbos · 8 years ago
  5. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  6. 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
  7. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  8. 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
  9. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  10. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 8 years ago
  11. 5a4a75a Combining SetVideoSend and SetSource into one method. by deadbeef · 8 years ago
  12. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  13. 6ab3db2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 8 years ago
  14. 65fc62e Remove webrtc/base/scoped_ptr.h by kwiberg · 8 years ago
  15. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 8 years ago
  16. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 8 years ago
  17. fcc640f Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector, by nisse · 8 years ago
  18. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
  19. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 8 years ago
  20. b24317b Fix license headers in webrtc/api. by kjellander · 8 years ago
  21. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 8 years ago[Renamed (95%) from talk/app/webrtc/rtpsender.h]
  22. a96e2d7 Move talk/media to webrtc/media by kjellander · 8 years ago
  23. e1f9d83 Adding AddTrack/RemoveTrack to native PeerConnection API. by deadbeef · 9 years ago
  24. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  25. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  26. 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
  27. 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
  28. 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
  29. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  30. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  31. 70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 9 years ago
  32. 6979b02 Adding stub files for RtpSender/RtpReceiver. by deadbeef · 9 years ago