Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / f74d641fc1ec325927e5f5b78bf3eda7daa8672e / call
  1. 4b1a363 Add fine grained dropped video frames counters on sending side by Ilya Nikolaevskiy · 8 years ago
  2. a5fbc23 BWE allocation strategy by Alex Narest · 8 years ago
  3. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 8 years ago
  4. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 8 years ago
  5. b3944f0 Media track ID visibility at BWE level by Alex Narest · 8 years ago
  6. 05d9822 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss. by Taylor Brandstetter · 8 years ago
  7. b709cf8 Remove Call::ParseRtpPacket by Danil Chapovalov · 8 years ago
  8. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 8 years ago
  9. 4bece9a Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. by Åsa Persson · 8 years ago
  10. 4332d09 Reland "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 8 years ago
  11. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 8 years ago
  12. 39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 8 years ago
  13. 68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 8 years ago
  14. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 8 years ago
  15. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 8 years ago
  16. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 8 years ago
  17. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 8 years ago
  18. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 8 years ago
  19. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 8 years ago
  20. 440216f Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. by Bjorn Terelius · 8 years ago
  21. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 8 years ago
  22. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 8 years ago
  23. 5aea38c Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}. by Alex Loiko · 8 years ago
  24. 06319b7 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Mac. by Alex Loiko · 8 years ago
  25. 1405afe Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Linux due to flakiness. by lliuu · 8 years ago
  26. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 8 years ago
  27. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 8 years ago
  28. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 8 years ago
  29. 99a81b6 Remove #include of rtc_stream_config.h from rtc_event_log.h by Elad Alon · 8 years ago
  30. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  31. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 8 years ago
  32. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  33. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago