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gerrit-public.fairphone.software / platform / external / webrtc / d8be107c8b2e24b9648fb9c8d3243d495d218c53 / video / quality_threshold.h
  1. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 7 years ago
  2. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  3. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/quality_threshold.h]
  4. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  5. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  6. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  7. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  8. a40672a Add UMA stats to bad call detection. by palmkvist · 8 years ago
  9. 349092b Logging basic bad call detection by palmkvist · 8 years ago