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gerrit-public.fairphone.software / platform / external / webrtc / 8493594dc2ae27954a5084c2fd31a6e4325fe7f1 / call / video_send_stream.h
  1. 70473fc Reland "Add hugeFramesSent GetStats metric" by Ilya Nikolaevskiy · 7 years ago
  2. 8ddc2e6 Revert "Add hugeFramesSent GetStats metric" by Max Morin · 7 years ago
  3. f9f71b9 Add hugeFramesSent GetStats metric by Ilya Nikolaevskiy · 7 years ago
  4. cc7125f Sets sending status for active RtpRtcp modules. by Seth Hampson · 7 years ago
  5. 3587b83 Make RTCP report interval configurable by Jiawei Ou · 7 years ago
  6. 6539f69 Add VideoSendStream::Config::EncoderSettings::experiment_cpu_load_estimator. by Niels Möller · 7 years ago
  7. 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
  8. be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
  9. 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
  10. c3ed630 Add stats googHasEnteredLowResolution. by Åsa Persson · 7 years ago
  11. d79314f Reland "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  12. 1c1a681 Revert "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  13. 4b1a363 Add fine grained dropped video frames counters on sending side by Ilya Nikolaevskiy · 7 years ago
  14. b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
  15. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  16. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  17. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/video_send_stream.h]
  18. 50864a8 Add reporting of googContentType via GetStats on send side by ilnik · 7 years ago
  19. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  20. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago