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gerrit-public.fairphone.software / platform / external / webrtc / 8fdcac3f06ae09fe17bbed02037ef136f086cc00 / video / receive_statistics_proxy.h
  1. 8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
  2. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  3. 81327d5 Move stats for delayed frames to renderer from VCMTiming to ReceiveStatisticsProxy. by Åsa Persson · 6 years ago
  4. 3a79a9a Remove deprecated API methods in video pipeline by Ilya Nikolaevskiy · 6 years ago
  5. 94150ee Implement VideoQualityObserver by Ilya Nikolaevskiy · 6 years ago
  6. 0beed5d Move SampleCounter from ReceiveStatisticsProxy to rtc_base/numerics by Ilya Nikolaevskiy · 6 years ago
  7. 4c8811b Delete some obsolete forward declarations by Niels Möller · 6 years ago
  8. 132e28e Add thread checks to ReceiveStatisticsProxy that reflect design comments. by Tommi · 6 years ago
  9. d397a0d Add dropped frames metric on the receive side by Ilya Nikolaevskiy · 6 years ago
  10. b9b07ea Move stats for decoded frames per second from VCMTiming to ReceiveStatisticsProxy. by Åsa Persson · 7 years ago
  11. 65c3922 Move some more numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  12. ed23be9 Move HistogramPercentileCounter to rtc_base from RecieveStatisticProxy. by Ilya Nikolaevskiy · 7 years ago
  13. daa4f7a Calculate and report to UMA 95th percentile of Interframe Delay by Ilya Nikolaevskiy · 7 years ago
  14. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  15. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  16. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/receive_statistics_proxy.h]
  17. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  18. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  19. 75204c5 Change reporting of timing frames conditions in GetStats on receive side by ilnik · 7 years ago
  20. 6d5b4d6 Piggybacking simulcast id and ALR experiment id into video content type extension. by ilnik · 7 years ago
  21. a79cc28 Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
  22. 3e86e7e Ignore inter-frame delay stats samples when stream is inactive by sprang · 7 years ago
  23. 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
  24. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  25. 2edc684 Report timing frames info in GetStats. by ilnik · 7 years ago
  26. 4257ab2 Add received interframe delay UMA metrics by ilnik · 7 years ago
  27. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  28. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  29. 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
  30. 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
  31. 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
  32. 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
  33. 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
  34. 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
  35. 64e739a Add content type information to Encoded Images and add corresponding RTP extension header. by ilnik · 7 years ago
  36. 0255acb Change VideoReceiveStream::Stats total_bitrate_bps to include all received packets. by asapersson · 7 years ago
  37. 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
  38. 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
  39. e525d6a Revert Make the new jitter buffer the default jitter buffer. by stefan · 7 years ago
  40. 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
  41. ff0e72f Add QP sum stats for received streams. by sakal · 7 years ago
  42. 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
  43. 27378f3 Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:290001 of https://codereview.chromium.org/2652043005/ ) by philipel · 7 years ago
  44. 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
  45. 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
  46. 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
  47. 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
  48. 0f0763d Make the new jitter buffer the default jitter buffer. by philipel · 8 years ago
  49. a40672a Add UMA stats to bad call detection. by palmkvist · 8 years ago
  50. 6966bd5 ReceiveStatisticsProxy: by asapersson · 8 years ago
  51. 349092b Logging basic bad call detection by palmkvist · 8 years ago
  52. 0c43f77 Update video histograms that do not have a minimum lifetime limit before being recorded. by asapersson · 8 years ago
  53. de9e5ff Add stats for frequency offset when converting RTP timestamp to NTP time. by asapersson · 8 years ago
  54. b5f2c3f Rename FecConfig to UlpfecConfig in config.h. by brandtr · 8 years ago
  55. 1490f7a Add histogram for end-to-end delay: "WebRTC.Video.EndToEndDelayInMs" by asapersson · 8 years ago
  56. 4374a09 Only update codec type histogram if lifetime is long enough (10 sec). by asapersson · 8 years ago
  57. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 8 years ago
  58. cfc8e3b Removed all RTP dependencies from ViEChannel and renamed class. by mflodman · 8 years ago
  59. 8688a4e Add histogram stats for jitter buffer delay and current/target delay for received video streams: by asapersson · 8 years ago
  60. a96b60b Move frame_callback.h to common_video/include. by pbos · 8 years ago
  61. a186288 Revert of Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way delay. (patchset #4 id:60001 of https://codereview.webrtc.org/1688143003/ ) by asapersson · 8 years ago
  62. 7ade7b3 Delete class webrtc::VideoRenderer and its header file. by nisse · 8 years ago
  63. f8cdd18 Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs" by asapersson · 8 years ago
  64. 5249599 Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way delay. by asapersson · 8 years ago
  65. 0ab8e81 Move histograms for rtp receive counters to ReceiveStatisticsProxy by sprang · 8 years ago
  66. f75d008 Bitrate controller for VideoToolbox encoder. by tkchin · 8 years ago
  67. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  68. b7d9a97 Expose codec implementation names in stats. by Peter Boström · 9 years ago
  69. 7623ce4 Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ ) by Peter Boström · 9 years ago
  70. 8237abf Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) by kjellander · 9 years ago
  71. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  72. 2557b86 modules/video_coding refactorings by Henrik Kjellander · 9 years ago
  73. 86b0160 Add stats for average QP per frame for VP8 (for received video streams): by asapersson · 9 years ago
  74. f839dcc Add stats for rendered pixels (sqrt(w*h)) per second: by asapersson · 9 years ago
  75. 13c433c Add delay metric (includes network delay (rtt/2) + jitter delay + decode time + render delay): by asapersson · 9 years ago
  76. 6304626 Add a rate tracker that tracks rate over a given interval split up into buckets that accumulate unit counts for their portion of said interval and use this instead of the standard rate tracker so that the values of retrieved frame rate stats are completely independent of the polling rate. by Tim Psiaki · 9 years ago
  77. f42376c Wire up currently-received video codec to stats. by pbos · 9 years ago
  78. 6718e97 Add encode and decode time to histograms stats: by asapersson · 9 years ago
  79. d89920b Add resolution and fps stats to histograms: by asapersson · 9 years ago
  80. d6f1a38 Remove ViEChannel simulcast lock. by Peter Boström · 9 years ago
  81. 300eeb6 Remove VideoEngine interfaces. by Peter Boström · 9 years ago
  82. 45553ae Remove VideoEngine interface usage from new API. by Peter Boström · 9 years ago
  83. f2f8283 Use rtc::CriticalSection in webrtc/video/. by Peter Boström · 9 years ago
  84. 3c391cb Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api. by Åsa Persson · 9 years ago
  85. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 9 years ago
  86. 00b8f6b Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away by kwiberg@webrtc.org · 9 years ago
  87. 1d0fa5d Add RtcpPacketTypeCounter stats to new API. by pbos@webrtc.org · 9 years ago
  88. 5570769 Remove the last getters from VideoReceiveStream stats. by pbos@webrtc.org · 10 years ago
  89. ce4e9a3 Refactor some receive-side stats. by pbos@webrtc.org · 10 years ago
  90. 98c04b3 Get avg_delay_ms from DecoderTiming callback. by pbos@webrtc.org · 10 years ago
  91. 0bae1fa Wire up bandwidth stats to the new API and webrtcvideoengine2. by stefan@webrtc.org · 10 years ago
  92. 38344ed Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
  93. de1429e Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  94. 0931570 Wire up statistics in video receive stream of new API by sprang@webrtc.org · 10 years ago