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gerrit-public.fairphone.software / platform / external / webrtc / d5585ca95632b94a6069b31edda53c0b45647e7d / video
  1. edf4ff7 Only treat H.264 frames containing SPS, PPS, and IDR as key frames. by Rasmus Brandt · 7 years ago
  2. ccdfcca New PacketQueue2 behind WebRTC-RoundRobinPacing field trial. by philipel · 7 years ago
  3. 1c1a681 Revert "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  4. d29b54c Set start time for encoded framerate tracker on first incoming frame (instead of by Åsa Persson · 7 years ago
  5. f74d641 Simplify setting/unsetting REMB in RtcpSender by Danil Chapovalov · 7 years ago
  6. 4b1a363 Add fine grained dropped video frames counters on sending side by Ilya Nikolaevskiy · 7 years ago
  7. b06b358 Update aggregating interval in getStats for receive side. by Ilya Nikolaevskiy · 7 years ago
  8. 0122e84 Reland "Remove sent framerate and bitrate calculations from MediaOptimization." by Åsa Persson · 7 years ago
  9. b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
  10. ed23be9 Move HistogramPercentileCounter to rtc_base from RecieveStatisticProxy. by Ilya Nikolaevskiy · 7 years ago
  11. ca0ed63 Revert "Remove sent framerate and bitrate calculations from MediaOptimization." by Åsa Persson · 7 years ago
  12. 18945c3 Revert "Reduce max possible size of map that holds encoded frame info." by Åsa Persson · 7 years ago
  13. 51e21aa Simplify RtpRtcp interface for REMB by Danil Chapovalov · 7 years ago
  14. 2ff7ecf Reduce max possible size of map that holds encoded frame info. by Åsa Persson · 7 years ago
  15. af721b7 Remove sent framerate and bitrate calculations from MediaOptimization. by Åsa Persson · 7 years ago
  16. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  17. 3f670e0 Fix potential crash bug in debug builds by Ilya Nikolaevskiy · 7 years ago
  18. ae81975 Make PictureIdTest more strict. by Åsa Persson · 7 years ago
  19. 4bece9a Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. by Åsa Persson · 7 years ago
  20. daa4f7a Calculate and report to UMA 95th percentile of Interframe Delay by Ilya Nikolaevskiy · 7 years ago
  21. d692ef9 Update comments for rename of ScalingObserverInterface. by Niels Möller · 7 years ago
  22. 22ec952 Delete in_order argument to RtpReceiver::IncomingRtpPacket by Niels Möller · 7 years ago
  23. 4332d09 Reland "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  24. c62f6c7 RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs by Karl Wiberg · 7 years ago
  25. 83ccca1 Create and use RtcEventLogOutput for output by Elad Alon · 7 years ago
  26. 39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  27. 68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  28. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
  29. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
  30. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  31. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
  32. 2c72fe8 Fix crash with rtc_event_log in video_loopback by Ilya Nikolaevskiy · 7 years ago
  33. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  34. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  35. c856dc2 Convert PayloadUnion from a union to a class, step 2 by Karl Wiberg · 7 years ago
  36. a82fcd0 Remove unused mocks of process thread by Danil Chapovalov · 7 years ago
  37. af8659a Rename test output to test artifacts. by Edward Lemur · 7 years ago
  38. 48462b6 Continuously request keyframes if decoding does not recover. by philipel · 7 years ago
  39. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  40. e21be1d Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) by philipel · 7 years ago
  41. b0573bc Reorganize config of RTP header extensions for video receive streams. by Niels Möller · 7 years ago
  42. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  43. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  44. 7cd28b9 Set protected_by_flexfec flag properly in tests. by brandtr · 7 years ago
  45. 73b60b8 Remove the redundant method GetPayloadSpecifics by Karl Wiberg · 7 years ago
  46. 8d75ac7 Add stats for forced software encoder fallback for VP8. by asapersson · 7 years ago
  47. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  48. 743117f Disable FullStackTest.LargeRoomVP8_*thumb on iOS by oprypin · 7 years ago
  49. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  50. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago