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gerrit-public.fairphone.software / platform / external / webrtc / 13e743419d3d27f152898a5f497e62196482ccb0 / video / call_stats.cc
  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/video/call_stats.cc]
  3. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  4. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  5. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  6. 1d02d3e Remove RTC_LOGGED_* macro. by asapersson · 8 years ago
  7. d28db7f Delete all use of tick_util.h. by Niels Möller · 9 years ago
  8. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  9. 58d992e Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*). by asapersson · 9 years ago
  10. a26ac92 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ ) by pbos · 9 years ago
  11. da33a8a Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ ) by torbjorng · 9 years ago
  12. f14c47a Remove ignored return code from modules. by Peter Boström · 9 years ago
  13. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
  14. 97888bd Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video. by Tommi · 9 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 (98%) from webrtc/video_engine/call_stats.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 (98%) from webrtc/video/call_stats.cc]
  18. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago[Renamed (98%) from webrtc/video_engine/call_stats.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. 3c089d7 Add RTC_ prefix to contructormagic macros. by henrikg · 9 years ago
  22. 2328a94 Add average rtt to CallStatsObserver and an average rtt histogram. by stefan · 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. 0b1534c Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. by pkasting@chromium.org · 10 years ago
  25. 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
  26. 5574dac Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 11 years ago
  27. 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
  28. 12dc1a3 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  29. 8ca8a71 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 12 years ago
  30. ccd4b2a Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 12 years ago
  31. aea96d3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 12 years ago
  32. b2f474e Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago