1. e618cc9 Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API by Artem Titov · 4 years, 6 months ago
  2. 72d6915 Populate sdp_fmtp_line and channels of RTCCodecStats by Johannes Kron · 4 years, 7 months ago
  3. 0c626af Use newer version of TimeDelta and TimeStamp factories in webrtc by Danil Chapovalov · 4 years, 7 months ago
  4. 910cdc2 Add a round-trip test that stats.toJson output is parseable by Harald Alvestrand · 4 years, 8 months ago
  5. 4f40fa5 Implement RTCOutboundRtpStreamStats::remoteId. by Henrik Boström · 4 years, 8 months ago
  6. 00376e1 Add totalInterFrameDelay to RTCInboundRTPStreamStats by Johannes Kron · 4 years, 9 months ago
  7. 5cb7807 Implement crypto stats on DTLS transport by Harald Alvestrand · 4 years, 10 months ago
  8. fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 10 months ago
  9. ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 4 years, 11 months ago
  10. ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 4 years, 10 months ago
  11. fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 4 years, 11 months ago
  12. 1b57541 Always pass arguments to INSTANTIATE_TEST_SUITE_P. by Mirko Bonadei · 5 years ago
  13. cc62b16 Add qualityLimitationResolutionChanges stat by Evan Shrubsole · 5 years ago
  14. 149dc72 Add support for RTCTransportStats.selectedCandidatePairChanges by Jonas Oreland · 5 years ago
  15. 0c141c5 Fix frames dropped statistics by Johannes Kron · 5 years ago
  16. 6b43086 Reland "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Boström · 5 years ago
  17. df625f4 Revert "[GetStats] Expose video codec implementation in standardized metrics." by Henrik Andreassson · 5 years ago
  18. 2b9fa09 [GetStats] Expose video codec implementation in standardized metrics. by Henrik Boström · 5 years ago
  19. 928e7a3 Make ID of datachannel stats not depend on dc.id by Harald Alvestrand · 5 years ago
  20. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  21. d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
  22. bfd343b Add totalDecodeTime to RTCInboundRTPStreamStats by Johannes Kron · 5 years ago
  23. 2efae77 Add RTCStats for keyFramesEncoded, keyFramesDecoded. by Rasmus Brandt · 5 years ago
  24. 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago
  25. 8605fbf [getStats] Make remote-inbound-rtp.ssrc match outbound-rtp.ssrc. by Henrik Boström · 5 years ago
  26. 6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
  27. ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
  28. 883eefc Implement RTCRemoteInboundRtpStreamStats for both audio and video. by Henrik Boström · 5 years ago
  29. 646fda0 Implement RTCMediaSourceStats and friends in standard getStats(). by Henrik Boström · 5 years ago
  30. 23aff9b Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget. by Henrik Boström · 5 years ago
  31. 9fe1834 Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video. by Henrik Boström · 5 years ago
  32. 8d8ffdb Expose new audio stats on the API by Ivo Creusen · 5 years ago
  33. 44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 5 years ago
  34. cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 5 years ago
  35. 01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 5 years ago
  36. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 5 years ago
  37. 2e06926 Implement RTC[In/Out]boundRtpStreamStats.contentType. by Henrik Boström · 5 years ago
  38. f71362f Wire up RTCOutboundRtpStreamStats.totalEncodeTime. by Henrik Boström · 5 years ago
  39. 232b3fd Expose relative packet arrival delay metric in stats API. by Jakob Ivarsson · 5 years ago
  40. 40b030e Reland "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone." by Henrik Boström · 5 years ago
  41. c4dd730 Fix -Wextra-semi warnings. by Mirko Bonadei · 5 years ago
  42. ca890ee Revert "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone." by Mirko Bonadei · 6 years ago
  43. 05d43c6 Fix getStats() freeze bug affecting Chromium but not WebRTC standalone. by Henrik Boström · 6 years ago
  44. 1c9c9fc Replace replace_substrs with Abseil by Steve Anton · 6 years ago
  45. 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 6 years ago
  46. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  47. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  48. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/rtcstatscollector_unittest.cc]
  49. 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
  50. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  51. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  52. 6c6c9df Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain() by Benjamin Wright · 6 years ago
  53. f25303e Reland: Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
  54. 4905edb Reland: Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  55. 82c71af Revert "Modernize rtc::SSLCertificate" by Niklas Enbom · 6 years ago
  56. 6932fb2 Revert "Reland: Use unique_ptr and ArrayView in SSLFingerprint" by Mirko Bonadei · 6 years ago
  57. 55cd3ac Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
  58. 47f3240 Reland: Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  59. 2b15626 Revert "Use unique_ptr and ArrayView in SSLFingerprint" by Henrik Grunell · 6 years ago
  60. cc21e61 Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  61. 9551375 getStats: add relayProtocol by Philipp Hancke · 6 years ago
  62. 3bc0166 getStats: add kind alias for mediaType by Philipp Hancke · 6 years ago
  63. 6b1985d Reimplement rtc::ToString and rtc::FromString without streams. by Jonas Olsson · 6 years ago
  64. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  65. e12c1fe Removing warning suppression flags from pc/. by Mirko Bonadei · 6 years ago
  66. 66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 6 years ago
  67. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  68. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  69. 5b3541f RTCStatsCollector::GetStatsReport() with optional selector argument. by Henrik Boström · 6 years ago
  70. 13b8bad Final name changing of MediaStreamInterface.label() to id(). by Seth Hampson · 6 years ago
  71. 25e022f Deliver cached stats reports asynchronously. by Taylor Brandstetter · 6 years ago
  72. 87d5a74 Fix crash that occurs if GetStats is called from within OnStatsDelivered by Taylor Brandstetter · 6 years ago
  73. 70473fc Reland "Add hugeFramesSent GetStats metric" by Ilya Nikolaevskiy · 6 years ago
  74. 8ddc2e6 Revert "Add hugeFramesSent GetStats metric" by Max Morin · 6 years ago
  75. f9f71b9 Add hugeFramesSent GetStats metric by Ilya Nikolaevskiy · 6 years ago
  76. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 6 years ago
  77. c392866 Implement certificate chain stats. by Taylor Brandstetter · 6 years ago
  78. 57858b3 Reland "Update RTCStatsCollector to work with RtpTransceivers" by Steve Anton · 7 years ago
  79. ee2388f Revert "Update RTCStatsCollector to work with RtpTransceivers" by Guido Urdaneta · 7 years ago
  80. 56bae8d Update RTCStatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
  81. 5b38731 Use fake PeerConnection for RTCStatsCollector tests by Steve Anton · 7 years ago
  82. 76d2952 Don't crash when sender info has been discarded by lower layers. by Harald Alvestrand · 7 years ago
  83. be5e208 Add FakePeerConnectionBase by Steve Anton · 7 years ago
  84. 2d8609c Move internal PeerConnection methods to PeerConnectionInternal by Steve Anton · 7 years ago
  85. b8e1201 Generate track stats when SSRC=0 by Harald Alvestrand · 7 years ago
  86. a3dab84 Refactor stream stats generation by Harald Alvestrand · 7 years ago
  87. c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
  88. c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
  89. e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
  90. 8906187 Pivot generation of stats to iterate senders/receivers by Harald Alvestrand · 7 years ago
  91. 7411648 Remove SessionStats.proxy_to_transport by Steve Anton · 7 years ago
  92. 593e325 Change RTCStatsCollector to only access channels from signaling thread by Steve Anton · 7 years ago
  93. 719487e Generate signed packets_lost in WebRTC-stats by Harald Alvestrand · 7 years ago
  94. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  95. 37e489c Add network_type to local RTCIceCandidateStats by Gary Liu · 7 years ago
  96. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  97. cbc71b2 Optional: Use nullopt and implicit construction in /pc/rtcstatscollector_unittest.cc by Oskar Sundbom · 7 years ago
  98. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  99. 75737c0 Merge WebRtcSession into PeerConnection by Steve Anton · 7 years ago
  100. ba81867 Prepare WebRtcSession to be merged into PeerConnection by Steve Anton · 7 years ago