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gerrit-public.fairphone.software / platform / external / webrtc / fae640003cf028477819309777ecd29a74c25eff / video / receive_statistics_proxy.cc
  1. 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 4 years, 8 months ago
  2. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 9 months ago
  3. e76b3ab Add per frame decode time histograms for 4k/HD and VP9/H264 by Johannes Kron · 4 years, 9 months ago
  4. 608083b Reset QP sum when QP is not reported on decoded frame. by Mirta Dvornicic · 4 years, 10 months ago
  5. caef51e Consolidate FEC book-keeping by Niels Möller · 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. a52e9bd Use StreamStatistician::BitrateReceived to produce total_bitrate_bps for GetStats. by Niels Möller · 5 years ago
  9. 12ebfa6 Delete RtcpStatisticsCallback from ReceiveStatistics by Niels Möller · 5 years ago
  10. 4d7c405 Split out RtcpCnameCallback from RtcpStatisticsCallback by Niels Möller · 5 years ago
  11. 9a9f18a Get WebRTC.Video.ReceivedPacketsLostInPercent from ReceiveStatistics by Niels Möller · 5 years ago
  12. 77d3efc Simplify ReportBlockStats by Niels Möller · 5 years ago
  13. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  14. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
  15. afb8d5c Log average decoded and rendered framerate for a VideoReceiveStream. by Åsa Persson · 5 years ago
  16. c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 5 years ago
  17. dd41da6 Delete unused methods from VCMReceiveStatisticsCallback by Niels Möller · 5 years ago
  18. 9b0b1e0 Delete unused method VCMReceiveStatisticsCallback::OnReceiveRatesUpdated by Niels Möller · 5 years ago
  19. 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 5 years ago
  20. 739baf0 [clang-tidy] Apply performance-for-range-copy fixes. by Mirko Bonadei · 6 years ago
  21. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  22. 278f825 Calculate video quality metrics only for rendered frames. by Sergey Silkin · 6 years ago
  23. 514f084 New statistic added to VideoReceiveStream to determine latency to first decode. by Benjamin Wright · 6 years ago
  24. 3f10ca8 Always record receive timestamps even on when the invalid flag is set. by Benjamin Wright · 6 years ago
  25. f203d73 Correctly slice MediaBitrateRecieved on content type in ReceiveStatisticsProxy by Ilya Nikolaevskiy · 6 years ago
  26. cdc959f Compute video freeze metrics on rendered frames instead of on decoded by Ilya Nikolaevskiy · 6 years ago
  27. 147013a Move call of stat's OnPreDecode to VideoReceiveStream::Decode by Niels Möller · 6 years ago
  28. 8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
  29. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  30. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  31. 81327d5 Move stats for delayed frames to renderer from VCMTiming to ReceiveStatisticsProxy. by Åsa Persson · 6 years ago
  32. 3a79a9a Remove deprecated API methods in video pipeline by Ilya Nikolaevskiy · 6 years ago
  33. 94150ee Implement VideoQualityObserver by Ilya Nikolaevskiy · 6 years ago
  34. 0beed5d Move SampleCounter from ReceiveStatisticsProxy to rtc_base/numerics by Ilya Nikolaevskiy · 6 years ago
  35. 881f168 Make SimpleStringBuilder into a non-template by Karl Wiberg · 6 years ago
  36. fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 6 years ago
  37. 132e28e Add thread checks to ReceiveStatisticsProxy that reflect design comments. by Tommi · 6 years ago
  38. d397a0d Add dropped frames metric on the receive side by Ilya Nikolaevskiy · 6 years ago
  39. 694a36f Only log once per UpdateHistogram call. by Jonas Olsson · 6 years ago
  40. b9b07ea Move stats for decoded frames per second from VCMTiming to ReceiveStatisticsProxy. by Åsa Persson · 7 years ago
  41. cabe383 Moved ALR experiment settings to new experiments folder. by Sebastian Jansson · 7 years ago
  42. 8e07c13 Optional: Use nullopt and implicit construction in /video by Oskar Sundbom · 7 years ago
  43. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  44. b06b358 Update aggregating interval in getStats for receive side. by Ilya Nikolaevskiy · 7 years ago
  45. ed23be9 Move HistogramPercentileCounter to rtc_base from RecieveStatisticProxy. by Ilya Nikolaevskiy · 7 years ago
  46. 3f670e0 Fix potential crash bug in debug builds by Ilya Nikolaevskiy · 7 years ago
  47. daa4f7a Calculate and report to UMA 95th percentile of Interframe Delay by Ilya Nikolaevskiy · 7 years ago
  48. 3b3622f Delete member VideoReceiveStream::Config::Rtp::ulpfec. by nisse · 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[Renamed from webrtc/video/receive_statistics_proxy.cc]
  51. 2e1b40b Implement googContentType GetStats metric reported on receive side. by ilnik · 7 years ago
  52. 75204c5 Change reporting of timing frames conditions in GetStats on receive side by ilnik · 7 years ago
  53. 6d5b4d6 Piggybacking simulcast id and ALR experiment id into video content type extension. by ilnik · 7 years ago
  54. a79cc28 Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
  55. 3e86e7e Ignore inter-frame delay stats samples when stream is inactive by sprang · 7 years ago
  56. 892dab5 Fix incorrect InterframeDelayMaxInMs histogram metrics by sprang · 7 years ago
  57. d083e85 Remove traces from {send,receive}_statistics_proxy.cc by ehmaldonado · 7 years ago
  58. f04afde Report interframe delay sum in old GetStats by ilnik · 7 years ago
  59. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  60. 2edc684 Report timing frames info in GetStats. by ilnik · 7 years ago
  61. 4257ab2 Add received interframe delay UMA metrics by ilnik · 7 years ago
  62. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  63. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  64. c1b5ea9 Add traces for some video receive statistics. by ehmaldonado · 7 years ago
  65. 2077f2f Add some unit tests to ReceiveStatsticsProxy. by asapersson · 7 years ago
  66. 948b275 Update decode/render fps stats when calling VideoReceiveStream::GetStats by sprang · 7 years ago
  67. b99baf8 Only record received key frame histogram stats if a certain number of frames (kMinRequiredSamples) have been received from OnCompleteFrame callback. by asapersson · 7 years ago
  68. 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 · 7 years ago
  69. 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 · 7 years ago
  70. 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 · 7 years ago
  71. 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 · 7 years ago
  72. 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 · 7 years ago
  73. 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 · 7 years ago
  74. 64e739a Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 7 years ago
  75. 0255acb Change VideoReceiveStream::Stats total_bitrate_bps to include all received packets. by asapersson · 7 years ago
  76. a563c21 Increase kMinRequiredSamples (5 -> 200) for updating histogram stats from OnFrameBufferTimingsUpdated callback. by asapersson · 7 years ago
  77. 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 · 7 years ago
  78. cc452e1 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) by sakal · 7 years ago
  79. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 7 years ago
  80. 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 7 years ago
  81. ff0e72f Add QP sum stats for received streams. by sakal · 7 years ago
  82. e5bd702 Reland of Make the new jitter buffer the default jitter buffer. (patchset #2 id:260001 of https://codereview.chromium.org/2656983002/ ) by philipel · 7 years ago
  83. 1474212 Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) by brandtr · 8 years ago
  84. 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
  85. e497495 Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) by kjellander · 8 years ago
  86. fe2bef3 Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. by brandtr · 8 years ago
  87. 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
  88. 50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 8 years ago
  89. 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
  90. 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
  91. 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
  92. 0f0763d Make the new jitter buffer the default jitter buffer. by philipel · 8 years ago
  93. a40672a Add UMA stats to bad call detection. by palmkvist · 8 years ago
  94. c7e7e07 Lower bad call logging severity by palmkvist · 8 years ago
  95. 6966bd5 ReceiveStatisticsProxy: by asapersson · 8 years ago
  96. 349092b Logging basic bad call detection by palmkvist · 8 years ago
  97. 0c43f77 Update video histograms that do not have a minimum lifetime limit before being recorded. by asapersson · 8 years ago
  98. be74270 Calculate JitterBufferDelayInMs in the new jitter buffer. by philipel · 8 years ago
  99. 43cb716 Add ToString method to AggregatedStats and log stats at the end of a call. by asapersson · 8 years ago
  100. de9e5ff Add stats for frequency offset when converting RTP timestamp to NTP time. by asapersson · 8 years ago