Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / d8579e0133fdc7afdd7fb10520f195646cee5754 / call
  1. 05d9822 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss. by Taylor Brandstetter · 7 years ago
  2. b709cf8 Remove Call::ParseRtpPacket by Danil Chapovalov · 7 years ago
  3. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  4. 4bece9a Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. by Åsa Persson · 7 years ago
  5. 4332d09 Reland "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  6. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  7. 39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  8. 68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  9. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
  10. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  11. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
  12. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  13. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  14. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  15. 440216f Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets. by Bjorn Terelius · 7 years ago
  16. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  17. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  18. 5aea38c Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}. by Alex Loiko · 7 years ago
  19. 06319b7 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Mac. by Alex Loiko · 7 years ago
  20. 1405afe Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Linux due to flakiness. by lliuu · 7 years ago
  21. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  22. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  23. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  24. 99a81b6 Remove #include of rtc_stream_config.h from rtc_event_log.h by Elad Alon · 7 years ago
  25. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  26. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  27. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  28. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago