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gerrit-public.fairphone.software / platform / external / webrtc / 7c03bdc1d343b322939cc5c8a0d2baa5786da060 / pc / rtc_stats_collector_unittest.cc
  1. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 6 years ago
  2. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  3. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/rtcstatscollector_unittest.cc]
  4. 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
  5. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 6 years ago
  6. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 6 years ago
  7. 6c6c9df Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain() by Benjamin Wright · 6 years ago
  8. f25303e Reland: Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
  9. 4905edb Reland: Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  10. 82c71af Revert "Modernize rtc::SSLCertificate" by Niklas Enbom · 6 years ago
  11. 6932fb2 Revert "Reland: Use unique_ptr and ArrayView in SSLFingerprint" by Mirko Bonadei · 6 years ago
  12. 55cd3ac Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
  13. 47f3240 Reland: Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  14. 2b15626 Revert "Use unique_ptr and ArrayView in SSLFingerprint" by Henrik Grunell · 6 years ago
  15. cc21e61 Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  16. 9551375 getStats: add relayProtocol by Philipp Hancke · 6 years ago
  17. 3bc0166 getStats: add kind alias for mediaType by Philipp Hancke · 6 years ago
  18. 6b1985d Reimplement rtc::ToString and rtc::FromString without streams. by Jonas Olsson · 6 years ago
  19. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  20. e12c1fe Removing warning suppression flags from pc/. by Mirko Bonadei · 6 years ago
  21. 66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 6 years ago
  22. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  23. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  24. 5b3541f RTCStatsCollector::GetStatsReport() with optional selector argument. by Henrik Boström · 7 years ago
  25. 13b8bad Final name changing of MediaStreamInterface.label() to id(). by Seth Hampson · 7 years ago
  26. 25e022f Deliver cached stats reports asynchronously. by Taylor Brandstetter · 7 years ago
  27. 87d5a74 Fix crash that occurs if GetStats is called from within OnStatsDelivered by Taylor Brandstetter · 7 years ago
  28. 70473fc Reland "Add hugeFramesSent GetStats metric" by Ilya Nikolaevskiy · 7 years ago
  29. 8ddc2e6 Revert "Add hugeFramesSent GetStats metric" by Max Morin · 7 years ago
  30. f9f71b9 Add hugeFramesSent GetStats metric by Ilya Nikolaevskiy · 7 years ago
  31. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
  32. c392866 Implement certificate chain stats. by Taylor Brandstetter · 7 years ago
  33. 57858b3 Reland "Update RTCStatsCollector to work with RtpTransceivers" by Steve Anton · 7 years ago
  34. ee2388f Revert "Update RTCStatsCollector to work with RtpTransceivers" by Guido Urdaneta · 7 years ago
  35. 56bae8d Update RTCStatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
  36. 5b38731 Use fake PeerConnection for RTCStatsCollector tests by Steve Anton · 7 years ago
  37. 76d2952 Don't crash when sender info has been discarded by lower layers. by Harald Alvestrand · 7 years ago
  38. be5e208 Add FakePeerConnectionBase by Steve Anton · 7 years ago
  39. 2d8609c Move internal PeerConnection methods to PeerConnectionInternal by Steve Anton · 7 years ago
  40. b8e1201 Generate track stats when SSRC=0 by Harald Alvestrand · 7 years ago
  41. a3dab84 Refactor stream stats generation by Harald Alvestrand · 7 years ago
  42. c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
  43. c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
  44. e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
  45. 8906187 Pivot generation of stats to iterate senders/receivers by Harald Alvestrand · 7 years ago
  46. 7411648 Remove SessionStats.proxy_to_transport by Steve Anton · 7 years ago
  47. 593e325 Change RTCStatsCollector to only access channels from signaling thread by Steve Anton · 7 years ago
  48. 719487e Generate signed packets_lost in WebRTC-stats by Harald Alvestrand · 7 years ago
  49. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  50. 37e489c Add network_type to local RTCIceCandidateStats by Gary Liu · 7 years ago
  51. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  52. cbc71b2 Optional: Use nullopt and implicit construction in /pc/rtcstatscollector_unittest.cc by Oskar Sundbom · 7 years ago
  53. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  54. 75737c0 Merge WebRtcSession into PeerConnection by Steve Anton · 7 years ago
  55. ba81867 Prepare WebRtcSession to be merged into PeerConnection by Steve Anton · 7 years ago
  56. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  57. 978b876 Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  58. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  59. bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 7 years ago
  60. 3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  61. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  62. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  63. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  64. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/rtcstatscollector_unittest.cc]
  65. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
  66. 1cc5fc3 Fix places that trigger no-unused-lambda-capture by eladalon · 7 years ago
  67. 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 7 years ago
  68. 35a872c Make RTCStatsReport::ToString() return JSON-parseable string. by ehmaldonado · 7 years ago
  69. e76bd3a Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 7 years ago
  70. 539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 7 years ago
  71. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 7 years ago
  72. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  73. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  74. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  75. f79ade1 Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 7 years ago
  76. d72098a Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 7 years ago
  77. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 7 years ago
  78. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 7 years ago
  79. 5bf9def RTCStatsCollector: Remove closed channels from opened set. by hbos · 8 years ago
  80. a7a9be1 Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect. by hbos · 8 years ago
  81. 13f54b2 Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine. by hbos · 8 years ago
  82. bf8d3e5 RTCIceCandidatePairStats.[total/current]RoundTripTime collected. by hbos · 8 years ago
  83. 92eaec6 RTCIceCandidatePairStats.nominated collected. by hbos · 8 years ago
  84. a51d4f3 Re-land of RTCInboundRTPStreamStats.qpSum collected. by hbos · 8 years ago
  85. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  86. ed02c6d Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ ) by skvlad · 8 years ago
  87. cd195be RTCInboundRTPStreamStats.qpSum collected. by hbos · 8 years ago
  88. 338f78a RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. by hbos · 8 years ago
  89. 3443bb7 RTCRTPStreamStats.ssrc changed type to uint32_t. by hbos · 8 years ago
  90. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
  91. b0ae920 RTCRTPStreamStats.mediaTrackId renamed to trackId. by hbos · 8 years ago
  92. 50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 8 years ago
  93. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/rtcstatscollector_unittest.cc]
  94. f64941f RTCMediaStreamTrackStats.framesDecoded collected. by hbos · 8 years ago
  95. fefe076 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector. by hbos · 8 years ago
  96. 42f6d2f RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 8 years ago
  97. 9e30274 RTCMediaStreamTrackStats collected on a per-attachment basis. by hbos · 8 years ago
  98. c8ee882 Replace use of ASSERT in test code. by nisse · 8 years ago
  99. bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
  100. 160e4a7 RTCMediaStreamTrackStats.kind added and collected. by hbos · 8 years ago