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gerrit-public.fairphone.software / platform / external / webrtc / 4970670c78efdf64048874c3bd34b3ebca7e65e7 / video / stream_synchronization_unittest.cc
  1. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  2. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  3. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  4. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  5. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/stream_synchronization_unittest.cc]
  6. fe50b4d Make class of static functions in rtp_to_ntp.h: - UpdateRtcpList - RtpToNtp by asapersson · 8 years ago
  7. b7e7b49 Use NtpTime in RtcpMeasurement instead of uint sec/uint frac. by asapersson · 8 years ago
  8. de9e5ff Add stats for frequency offset when converting RTP timestamp to NTP time. by asapersson · 8 years ago
  9. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  10. 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_unittest.cc]
  11. 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_unittest.cc]
  12. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago[Renamed (99%) from webrtc/video_engine/stream_synchronization_unittest.cc]
  13. 0fcaf99 Enable cpplint for webrtc/video_engine by kjellander@webrtc.org · 9 years ago
  14. 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 · 11 years ago
  15. f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  16. 4e545b3 Fixed remaining nits from Stefan by pwestin@webrtc.org · 12 years ago
  17. 6311733 Updated the sync module with a slow moving filter by pwestin@webrtc.org · 12 years ago
  18. 0d8d010 Handle multiple calls to set initial delay by mikhal@webrtc.org · 12 years ago
  19. ef9f76a Adding a receive side API for buffering mode. by mikhal@webrtc.org · 12 years ago
  20. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/video_engine/stream_synchronization_unittest.cc]
  21. 64d9dec Move RtpToNtp functionality to its own file. by stefan@webrtc.org · 12 years ago
  22. 7c3523c Change audio/video sync to be based on mapping RTP timestamps to NTP. by stefan@webrtc.org · 12 years ago
  23. d7a71d0 Prepare to roll Chromium to 149181. by andrew@webrtc.org · 12 years ago
  24. 5f28498 First step in refactoring audio/video synchronization. Adds unittests. by stefan@webrtc.org · 12 years ago