1. 5963c7c Count disabled due to low bw streams or layers as bw limited quality in GetStats by Ilya Nikolaevskiy · 4 years, 10 months ago
  2. 7c06777 Cleanup includes in modules/include/module_common_types.h by Danil Chapovalov · 4 years, 10 months ago
  3. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 4 years, 11 months ago
  4. 4d7c405 Split out RtcpCnameCallback from RtcpStatisticsCallback by Niels Möller · 5 years ago
  5. ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
  6. 87e3f9d [video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
  7. 9fe1834 Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video. by Henrik Boström · 5 years ago
  8. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  9. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  10. e2fd86a Move encoder metadata into EncoderInfo struct. by Erik Språng · 6 years ago
  11. d3b8c63 Reland "Add spatial index to EncodedImage." by Niels Möller · 6 years ago
  12. 5a998d7 Revert "Add spatial index to EncodedImage." by Niels Moller · 6 years ago
  13. da0898d Add spatial index to EncodedImage. by Niels Möller · 6 years ago
  14. 8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
  15. 213618e New api function CreateVideoStreamEncoder. by Niels Möller · 6 years ago
  16. dbdb3a0 Refactoring PayloadRouter. by Stefan Holmer · 6 years ago
  17. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  18. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  19. 97e0488 Delete unused stats for preferred_bitrate. by Niels Möller · 6 years ago
  20. 70473fc Reland "Add hugeFramesSent GetStats metric" by Ilya Nikolaevskiy · 6 years ago
  21. 8ddc2e6 Revert "Add hugeFramesSent GetStats metric" by Max Morin · 6 years ago
  22. f9f71b9 Add hugeFramesSent GetStats metric by Ilya Nikolaevskiy · 6 years ago
  23. 875841d Exclude initial adapt downs in stats for quality adapt changes per minute. by Åsa Persson · 7 years ago
  24. aa329e7 Reland: googBandwidthLimitedResolution stat is not always set depending on configuration. by Åsa Persson · 7 years ago
  25. 62e9ebe Revert "googBandwidthLimitedResolution stat is not always set depending on configuration." by Guido Urdaneta · 7 years ago
  26. 59283e4 googBandwidthLimitedResolution stat is not always set depending on configuration. by Åsa Persson · 7 years ago
  27. c3ed630 Add stats googHasEnteredLowResolution. by Åsa Persson · 7 years ago
  28. 45bbc8a Change forced software encoder fallback for VP8 to be only based on resolution and not bitrate. by Åsa Persson · 7 years ago
  29. d79314f Reland "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  30. 1c1a681 Revert "Add fine grained dropped video frames counters on sending side" by Ilya Nikolaevskiy · 7 years ago
  31. 4b1a363 Add fine grained dropped video frames counters on sending side by Ilya Nikolaevskiy · 7 years ago
  32. 0122e84 Reland "Remove sent framerate and bitrate calculations from MediaOptimization." by Åsa Persson · 7 years ago
  33. ca0ed63 Revert "Remove sent framerate and bitrate calculations from MediaOptimization." by Åsa Persson · 7 years ago
  34. af721b7 Remove sent framerate and bitrate calculations from MediaOptimization. by Åsa Persson · 7 years ago
  35. 8d75ac7 Add stats for forced software encoder fallback for VP8. by asapersson · 7 years ago
  36. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  37. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  38. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/video/send_statistics_proxy.h]
  39. a37de39 Update thread annotiation macros to use RTC_ prefix by danilchap · 7 years ago
  40. 5212700 Removing dependencies on stub headers within WebRTC. by mbonadei · 7 years ago
  41. cc3d442 Rename ViEEncoder to VideoStreamEncoder by mflodman · 7 years ago
  42. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  43. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  44. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  45. 09f0561 Update adaptation stats to support degradations in both resolution and framerate. by asapersson · 7 years ago
  46. 0944a80 Update stats for cpu/quality adaptation changes to excluded time when video is suspended. by asapersson · 7 years ago
  47. 6eca98b Add histogram stats for number of cpu/quality adapt changes per minute for sent video streams: by asapersson · 7 years ago
  48. c5d62e2 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ ) by sprang · 7 years ago
  49. 36e9eb4 Do not report quality limited resolution stats when quality scaler is disabled. by asapersson · 7 years ago
  50. f9ed235 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ ) by lliuu · 7 years ago
  51. 3ea3c77 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ ) by sprang · 7 years ago
  52. 8b45b11 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ ) by skvlad · 7 years ago
  53. 72acf25 Add framerate to VideoSinkWants and ability to signal on overuse by sprang · 7 years ago
  54. 93e1e23 Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call. by asapersson · 7 years ago
  55. bc5d921 Rename base/analytics/ to base/numerics/ by terelius · 8 years ago
  56. 66d4b37 Move histogram for number of pause events to per stream: by asapersson · 8 years ago
  57. 0cd27ba Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ ) by kthelgason · 8 years ago
  58. fe04bd4 Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ ) by kthelgason · 8 years ago
  59. 0c8c538 Properly report number of quality downscales in stats. by kthelgason · 8 years ago
  60. 876222f Move usage of QualityScaler to ViEEncoder. by kthelgason · 8 years ago
  61. 320e45a Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call. by asapersson · 8 years ago
  62. 69b627d Move smoothing filter to common audio and exp_filter to base/analytics. by minyue · 8 years ago
  63. 3c3aef4 Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ ) by minyue · 8 years ago
  64. 223641f Reland "Move smoothing filter to common audio". by minyue · 8 years ago
  65. 827cab3 Add qp counter for H264 in SendStatisticsProxy. by asapersson · 8 years ago
  66. 803d97f Let ViEEncoder express resolution requests as Sinkwants. by perkj · 8 years ago
  67. a48ddb7 Add VideoSendStream::Stats::prefered_media_bitrate_bps by Per · 8 years ago
  68. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 8 years ago
  69. 275afc5 Add codec name to CodecSpecificInfo and get the codec name stats from there instead. by perkj · 8 years ago
  70. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 8 years ago
  71. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 8 years ago
  72. 4374a09 Only update codec type histogram if lifetime is long enough (10 sec). by asapersson · 8 years ago
  73. cd349d9 Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ ) by sprang · 8 years ago
  74. a49f110 Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ ) by aluebs · 8 years ago
  75. 05ce4ae Reland Issue 2061423003: Refactor NACK bitrate allocation by Erik Språng · 8 years ago
  76. e5dd441 Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ ) by sprang · 8 years ago
  77. 5fc59e8 Refactor NACK bitrate allocation by Erik Språng · 8 years ago
  78. f5b2e51 Fix stats for encoder target bitrate when target rate is zero. by perkj · 8 years ago
  79. 69b332d Move logic for calculating needed bitrate overhead used by NACK and FEC to VideoSender. by Per · 8 years ago
  80. 376b192 Remove VideoCodingModule::VCMPacketizationCallback by perkj · 8 years ago
  81. 02b3d27 Reland of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #1 id:1 of https://codereview.webrtc.org/1903193002/ ) by kjellander · 8 years ago
  82. a261e61 Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ ) by kjellander · 8 years ago
  83. f5d55aa Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. by perkj · 8 years ago
  84. 5265fed Add histogram stats for average QP per frame for VP9 (for sent video streams): by asapersson · 8 years ago
  85. 118ef00 Add histogram stats for average QP per frame for VP8 (for sent video streams): by asapersson · 8 years ago
  86. 27f982b Replace scoped_ptr with unique_ptr in webrtc/video/ by kwiberg · 8 years ago
  87. 22c2b48 Move RTP stats histograms from VieChannel to SendStatisticsProxy. by Erik Språng · 8 years ago
  88. 07fb9be Move RTCP histograms from vie_channel to video channel stats proxies. by sprang · 8 years ago
  89. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 8 years ago
  90. e449915 Measure encoding time on encode callbacks. by Peter Boström · 8 years ago
  91. 5ad935c Remove mutable from rtc::CriticalSection members. by pbos · 9 years ago
  92. b7d9a97 Expose codec implementation names in stats. by Peter Boström · 9 years ago
  93. d1590b2 Lint clean video/ and add lint presubmit check. by mflodman · 9 years ago
  94. 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
  95. 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
  96. 03ef053 Merge webrtc/video_engine/ into webrtc/video/ by Peter Boström · 9 years ago
  97. 1aa420b Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead. by asapersson · 9 years ago
  98. b4a1ae5 Add separate send-side UMA stats for screenshare and video. by sprang · 9 years ago
  99. 2557b86 modules/video_coding refactorings by Henrik Kjellander · 9 years ago
  100. f040b23 Add histograms for send-side delay stats for a sent video stream: by asapersson · 9 years ago