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gerrit-public.fairphone.software / platform / external / webrtc / b1f2d604560f8e27552676fe9ec27ba946bd98d1 / video / call_stats.h
  1. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  2. dbdb3a0 Refactoring PayloadRouter. by Stefan Holmer · 6 years ago
  3. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  4. 38c5d93 Reduce locking for CallStats (preparation for TaskQueue). by Tommi · 7 years ago
  5. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  6. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/call_stats.h]
  7. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  8. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  9. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  10. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  11. 27f982b Replace scoped_ptr with unique_ptr in webrtc/video/ by kwiberg · 9 years ago
  12. a26ac92 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ ) by pbos · 9 years ago
  13. da33a8a Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ ) by torbjorng · 9 years ago
  14. f14c47a Remove ignored return code from modules. by Peter Boström · 9 years ago
  15. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
  16. d8de115 Remove mutable from rtc::CriticalSections. by pbos · 9 years ago
  17. 97888bd Swap use of CriticalSectionWrapper for rtc::CriticalSection in webrtc/video. by Tommi · 9 years ago
  18. 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 (94%) from webrtc/video_engine/call_stats.h]
  19. d3c9447 Nuke TickTime::UseFakeClock. by Peter Boström · 9 years ago
  20. 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 (94%) from webrtc/video/call_stats.h]
  21. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago[Renamed (94%) from webrtc/video_engine/call_stats.h]
  22. ff761fb modules: more interface -> include renames by Henrik Kjellander · 9 years ago
  23. 3c089d7 Add RTC_ prefix to contructormagic macros. by henrikg · 9 years ago
  24. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  25. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 10 years ago
  26. 16825b1 Use int64_t more consistently for times, in particular for RTT values. by pkasting@chromium.org · 10 years ago
  27. 0b1534c Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. by pkasting@chromium.org · 10 years ago
  28. 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
  29. 1972ff8 Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. by henrik.lundin@webrtc.org · 10 years ago
  30. 88fbb2d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 11 years ago
  31. 2fa7f79 Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 11 years ago
  32. 125ffd7 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 11 years ago
  33. 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
  34. 8ca8a71 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 12 years ago
  35. ccd4b2a Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 12 years ago
  36. aea96d3 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 12 years ago
  37. b586507 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 12 years ago
  38. b2f474e Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago