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gerrit-public.fairphone.software / platform / external / webrtc / 71a77c4b3b314a5e3b4e6b2f12d4886cff1b60d7 / video / receive_statistics_proxy_unittest.cc
  1. ef3998f Add directive to make webrtc metrics optional. by Ying Wang · 5 years ago
  2. 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  3. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 5 years ago
  4. e76b3ab Add per frame decode time histograms for 4k/HD and VP9/H264 by Johannes Kron · 5 years ago
  5. 608083b Reset QP sum when QP is not reported on decoded frame. by Mirta Dvornicic · 5 years ago
  6. 0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
  7. d781965 Delete StreamDataCountersCallback from ReceiveStatistics by Niels Möller · 5 years ago
  8. 12ebfa6 Delete RtcpStatisticsCallback from ReceiveStatistics by Niels Möller · 5 years ago
  9. 4d7c405 Split out RtcpCnameCallback from RtcpStatisticsCallback by Niels Möller · 5 years ago
  10. 9a9f18a Get WebRTC.Video.ReceivedPacketsLostInPercent from ReceiveStatistics by Niels Möller · 5 years ago
  11. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  12. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  13. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 6 years ago
  14. fc6f3e5 Include duration of pauses into sum of squared frames duration. by Sergey Silkin · 6 years ago
  15. dd41da6 Delete unused methods from VCMReceiveStatisticsCallback by Niels Möller · 6 years ago
  16. 9b0b1e0 Delete unused method VCMReceiveStatisticsCallback::OnReceiveRatesUpdated by Niels Möller · 6 years ago
  17. 12d1285 Use the new TEST_SUITE GoogleTest API (regression). by Mirko Bonadei · 6 years ago
  18. 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 6 years ago
  19. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 6 years ago
  20. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  21. 50e7745 Harmonic frame rate metric. by Sergey Silkin · 6 years ago
  22. 278f825 Calculate video quality metrics only for rendered frames. by Sergey Silkin · 6 years ago
  23. 1ebfb6a Introduce VideoFrame::id to keep track of frames inside application. by Artem Titov · 6 years ago
  24. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  25. cdc959f Compute video freeze metrics on rendered frames instead of on decoded by Ilya Nikolaevskiy · 6 years ago
  26. bea18ca Add number of freezes per minute metric. by Sergey Silkin · 6 years ago
  27. 147013a Move call of stat's OnPreDecode to VideoReceiveStream::Decode by Niels Möller · 6 years ago
  28. 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
  29. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  30. 81327d5 Move stats for delayed frames to renderer from VCMTiming to ReceiveStatisticsProxy. by Åsa Persson · 6 years ago
  31. 879f5a3 Add test against crashes in VideoQualityObserver by Ilya Nikolaevskiy · 6 years ago
  32. 94150ee Implement VideoQualityObserver by Ilya Nikolaevskiy · 6 years ago
  33. b9b07ea Move stats for decoded frames per second from VCMTiming to ReceiveStatisticsProxy. by Åsa Persson · 7 years ago
  34. 8e07c13 Optional: Use nullopt and implicit construction in /video by Oskar Sundbom · 7 years ago
  35. c7c4191 Declare the RTCP packets_lost field as signed in the API. by Harald Alvestrand · 7 years ago
  36. b06b358 Update aggregating interval in getStats for receive side. by Ilya Nikolaevskiy · 7 years ago
  37. ed23be9 Move HistogramPercentileCounter to rtc_base from RecieveStatisticProxy. by Ilya Nikolaevskiy · 7 years ago
  38. daa4f7a Calculate and report to UMA 95th percentile of Interframe Delay by Ilya Nikolaevskiy · 7 years ago
  39. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  40. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/receive_statistics_proxy_unittest.cc]
  41. 2e1b40b Implement googContentType GetStats metric reported on receive side. by ilnik · 7 years ago
  42. 75204c5 Change reporting of timing frames conditions in GetStats on receive side by ilnik · 7 years ago
  43. 6d5b4d6 Piggybacking simulcast id and ALR experiment id into video content type extension. by ilnik · 7 years ago
  44. a79cc28 Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
  45. 3e86e7e Ignore inter-frame delay stats samples when stream is inactive by sprang · 7 years ago
  46. 892dab5 Fix incorrect InterframeDelayMaxInMs histogram metrics by sprang · 7 years ago
  47. 186d9c3 Renamed fields in common_types.h/RtcpStatistics. by srte · 7 years ago
  48. f04afde Report interframe delay sum in old GetStats by ilnik · 7 years ago
  49. 2edc684 Report timing frames info in GetStats. by ilnik · 7 years ago
  50. 2077f2f Add some unit tests to ReceiveStatsticsProxy. by asapersson · 7 years ago
  51. f93752a Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ ) by nisse · 7 years ago
  52. 948b275 Update decode/render fps stats when calling VideoReceiveStream::GetStats by sprang · 8 years ago
  53. 61b22dd Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ ) by nisse · 8 years ago
  54. 3870a07 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854883002/ ) by nisse · 8 years ago
  55. 6e6a485 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2852303002/ ) by nisse · 8 years ago
  56. d71ebd7 Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2845333002/ ) by nisse · 8 years ago
  57. aec49d2 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ ) by nisse · 8 years ago
  58. 713a3bb Delete deprecated and transitional stuff related to video frame refactoring. by nisse · 8 years ago
  59. b99baf8 Only record received key frame histogram stats if a certain number of frames (kMinRequiredSamples) have been received from OnCompleteFrame callback. by asapersson · 8 years ago
  60. 00d802b Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ ) by ilnik · 8 years ago
  61. 27c46e2 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #4 id:400001 of https://codereview.webrtc.org/2812913002/ ) by ilnik · 8 years ago
  62. 774f6b4 Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
  63. 29dbb19 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2811963002/ ) by ilnik · 8 years ago
  64. 4fa0c4f Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ ) by ilnik · 8 years ago
  65. 5721866 Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ ) by ilnik · 8 years ago
  66. 64e739a Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 8 years ago
  67. 0255acb Change VideoReceiveStream::Stats total_bitrate_bps to include all received packets. by asapersson · 8 years ago
  68. a45102f Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ ) by philipel · 8 years ago
  69. cc452e1 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) by sakal · 8 years ago
  70. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 8 years ago
  71. 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 8 years ago
  72. ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
  73. e5bd702 Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) by philipel · 8 years ago
  74. 27378f3 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ ) by philipel · 8 years ago
  75. 09d6ef0 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2638423003/ ) by philipel · 8 years ago
  76. 50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 8 years ago
  77. 04926b8 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ ) by kjellander · 8 years ago
  78. f20dd00 Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ ) by philipel · 8 years ago
  79. c08c191 Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ ) by philipel · 8 years ago
  80. 0f0763d Make the new jitter buffer the default jitter buffer. by philipel · 8 years ago
  81. a40672a Add UMA stats to bad call detection. by palmkvist · 8 years ago
  82. 6966bd5 ReceiveStatisticsProxy: by asapersson · 8 years ago
  83. 0c43f77 Update video histograms that do not have a minimum lifetime limit before being recorded. by asapersson · 8 years ago
  84. 46c4e3c Add unit tests to ReceiveStatisticsProxy class. by asapersson · 8 years ago
  85. de9e5ff Add stats for frequency offset when converting RTP timestamp to NTP time. by asapersson · 8 years ago
  86. e5ba44e Implement framesDecoded stat in video receive ssrc stats. by sakal · 8 years ago