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gerrit-public.fairphone.software / platform / external / webrtc / 665174fdbb4e0540eccb27cf7412348f1b65534c / video / stream_synchronization.cc
  1. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  2. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  3. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  4. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/stream_synchronization.cc]
  5. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  6. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  7. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  8. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 7 years ago
  9. fe50b4d Make class of static functions in rtp_to_ntp.h: - UpdateRtcpList - RtpToNtp by asapersson · 8 years ago
  10. 4cd2790 Move RTP for synchroninzation and rename classes, files and variables. by mflodman · 8 years ago
  11. f8cdd18 Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs" by asapersson · 8 years ago
  12. 7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago[Renamed (99%) from webrtc/video_engine/stream_synchronization.cc]
  13. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago[Renamed (99%) from webrtc/video/stream_synchronization.cc]
  14. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago[Renamed (99%) from webrtc/video_engine/stream_synchronization.cc]
  15. 415d2cd Use webrtc/base/logging.h for video. by Peter Boström · 9 years ago
  16. 36a1438 Remove ViEFrameProviderBase. by Peter Boström · 9 years ago
  17. 66773a0 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
  18. 5574dac Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
  19. 47fadba Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
  20. 12dc1a3 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  21. e46c8d3 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
  22. f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  23. d35964a Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync. by pwestin@webrtc.org · 11 years ago
  24. 6311733 Updated the sync module with a slow moving filter by pwestin@webrtc.org · 11 years ago
  25. 0d8d010 Handle multiple calls to set initial delay by mikhal@webrtc.org · 11 years ago
  26. ef9f76a Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  27. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/video_engine/stream_synchronization.cc]
  28. 64d9dec Move RtpToNtp functionality to its own file. by stefan@webrtc.org · 12 years ago
  29. 2dcbcc1 Changing two asserts which should have returned errors instead. by stefan@webrtc.org · 12 years ago
  30. 7c3523c Change audio/video sync to be based on mapping RTP timestamps to NTP. by stefan@webrtc.org · 12 years ago
  31. 5f28498 First step in refactoring audio/video synchronization. Adds unittests. by stefan@webrtc.org · 12 years ago