1. 3f2634e Reland "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices." by Sergey Silkin · 7 years ago
  2. ae29428 Revert "Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices." by Ivo Creusen · 7 years ago
  3. 47836b4 Keep spatial_idx=kNoSpatialIdx(255) if there is no layer indices. by Sergey Silkin · 7 years ago
  4. 88f080a Move SPS/PPS/IDR requirement from RtpFrameObject to PacketBuffer. by Rasmus Brandt · 7 years ago
  5. 7172ea1 Don't use old RTCP SR reports for remote clock estimation by Ilya Nikolaevskiy · 7 years ago
  6. 996eb9e Fix typo in VideoSendTiming header extension structure by Danil Chapovalov · 7 years ago
  7. a056599 Delete VCMSendStatisticsCallback and corresponding use of ProcessThread by Niels Möller · 7 years ago
  8. a194e58 Move sequence_number_utils.h to rtc_base/ by Bjorn Terelius · 7 years ago
  9. 7c8cca3 Add check for send-side bwe before applying alr settings by Erik Språng · 7 years ago
  10. d79314f Reland "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  11. d9f99c1 Replace Atomic32 with std::atomic in video/ by Yuwei Huang · 7 years ago
  12. edf4ff7 Only treat H.264 frames containing SPS, PPS, and IDR as key frames. by Rasmus Brandt · 7 years ago
  13. ccdfcca New PacketQueue2 behind WebRTC-RoundRobinPacing field trial. by philipel · 7 years ago
  14. 1c1a681 Revert "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  15. d29b54c Set start time for encoded framerate tracker on first incoming frame (instead of by Åsa Persson · 7 years ago
  16. f74d641 Simplify setting/unsetting REMB in RtcpSender by Danil Chapovalov · 7 years ago
  17. 4b1a363 Add fine grained dropped video frames counters on sending side by Ilya Nikolaevskiy · 7 years ago
  18. b06b358 Update aggregating interval in getStats for receive side. by Ilya Nikolaevskiy · 7 years ago
  19. 0122e84 Reland "Remove sent framerate and bitrate calculations from MediaOptimization." by Åsa Persson · 7 years ago
  20. b3944f0 Media track ID visibility at BWE level by Alex Narest · 7 years ago
  21. ed23be9 Move HistogramPercentileCounter to rtc_base from RecieveStatisticProxy. by Ilya Nikolaevskiy · 7 years ago
  22. ca0ed63 Revert "Remove sent framerate and bitrate calculations from MediaOptimization." by Åsa Persson · 7 years ago
  23. 18945c3 Revert "Reduce max possible size of map that holds encoded frame info." by Åsa Persson · 7 years ago
  24. 51e21aa Simplify RtpRtcp interface for REMB by Danil Chapovalov · 7 years ago
  25. 2ff7ecf Reduce max possible size of map that holds encoded frame info. by Åsa Persson · 7 years ago
  26. af721b7 Remove sent framerate and bitrate calculations from MediaOptimization. by Åsa Persson · 7 years ago
  27. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  28. 3f670e0 Fix potential crash bug in debug builds by Ilya Nikolaevskiy · 7 years ago
  29. ae81975 Make PictureIdTest more strict. by Åsa Persson · 7 years ago
  30. 4bece9a Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled. by Åsa Persson · 7 years ago
  31. daa4f7a Calculate and report to UMA 95th percentile of Interframe Delay by Ilya Nikolaevskiy · 7 years ago
  32. d692ef9 Update comments for rename of ScalingObserverInterface. by Niels Möller · 7 years ago
  33. 22ec952 Delete in_order argument to RtpReceiver::IncomingRtpPacket by Niels Möller · 7 years ago
  34. 4332d09 Reland "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  35. c62f6c7 RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs by Karl Wiberg · 7 years ago
  36. 83ccca1 Create and use RtcEventLogOutput for output by Elad Alon · 7 years ago
  37. 39cefdb Revert "Reland "Remove WEBRTC_TRACE."" by Fredrik Solenberg · 7 years ago
  38. 68007e9 Reland "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  39. c3fa8e1 New method RtpReceiver::GetLatestTimestamps. by Niels Möller · 7 years ago
  40. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
  41. 729b910 Revert "Remove WEBRTC_TRACE." by Fredrik Solenberg · 7 years ago
  42. 2209b90 Remove WEBRTC_TRACE. by Fredrik Solenberg · 7 years ago
  43. 2c72fe8 Fix crash with rtc_event_log in video_loopback by Ilya Nikolaevskiy · 7 years ago
  44. 3102734 Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." by Rasmus Brandt · 7 years ago
  45. 2666cf7 Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). by Rasmus Brandt · 7 years ago
  46. c856dc2 Convert PayloadUnion from a union to a class, step 2 by Karl Wiberg · 7 years ago
  47. a82fcd0 Remove unused mocks of process thread by Danil Chapovalov · 7 years ago
  48. af8659a Rename test output to test artifacts. by Edward Lemur · 7 years ago
  49. 48462b6 Continuously request keyframes if decoding does not recover. by philipel · 7 years ago
  50. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 7 years ago
  51. 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
  52. b0573bc Reorganize config of RTP header extensions for video receive streams. by Niels Möller · 7 years ago
  53. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  54. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  55. 7cd28b9 Set protected_by_flexfec flag properly in tests. by brandtr · 7 years ago
  56. 73b60b8 Remove the redundant method GetPayloadSpecifics by Karl Wiberg · 7 years ago
  57. 8d75ac7 Add stats for forced software encoder fallback for VP8. by asapersson · 7 years ago
  58. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  59. 743117f Disable FullStackTest.LargeRoomVP8_*thumb on iOS by oprypin · 7 years ago
  60. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  61. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago