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gerrit-public.fairphone.software / platform / external / webrtc / 27c293665e3a2b0e2de12236529773a7b9057f6e / video / call_stats_unittest.cc
  1. c89468a Fix CallStatsUnittests to update the RTT on the process thread (as in production). by Tommi · 5 years ago
  2. 1f4173e Fix ClangTidy issues in video/ by Benjamin Wright · 5 years ago
  3. 1aa7581 Replace all usage of rtc::NewClosure with webrtc::ToQueuedTask by Danil Chapovalov · 5 years ago
  4. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  5. 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
  6. 38c5d93 Reduce locking for CallStats (preparation for TaskQueue). by Tommi · 6 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/video/call_stats_unittest.cc]
  9. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  10. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  11. 01d70a3 Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h. by asapersson · 8 years ago
  12. d28db7f Delete all use of tick_util.h. by Niels Möller · 8 years ago
  13. 27f982b Replace scoped_ptr with unique_ptr in webrtc/video/ by kwiberg · 8 years ago
  14. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 8 years ago
  15. 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/call_stats_unittest.cc]
  16. d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  17. 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/call_stats_unittest.cc]
  18. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago[Renamed (99%) from webrtc/video_engine/call_stats_unittest.cc]
  19. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  20. 98f5351 system_wrappers: rename interface -> include by Henrik Kjellander · 9 years ago
  21. 2328a94 Add average rtt to CallStatsObserver and an average rtt histogram. by stefan · 9 years ago
  22. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 9 years ago
  23. 16825b1 Use int64_t more consistently for times, in particular for RTT values. by pkasting@chromium.org · 10 years ago
  24. 8084f95 Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval. by asapersson@webrtc.org · 10 years ago
  25. 1ae1d0c Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
  26. f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  27. 8ca8a71 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  28. ccd4b2a Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
  29. aea96d3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 11 years ago
  30. b2f474e Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago