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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / be039c2c5df841a40f5b9e965ad889219b06e459 / video_engine / call_stats_unittest.cc
  1. c2162d1 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
  2. f2e6fb3 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  3. b35efcc Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  4. 65e6f91 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
  5. cb139b1 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 12 years ago
  6. 8a8517a Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled. by mflodman@webrtc.org · 12 years ago