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gerrit-public.fairphone.software / platform / external / webrtc / c2a0eb2699f7871c876a487105070f84c38a3dd0 / pc / statscollector_unittest.cc
  1. 978b876 Move clients of WebRtcSession to use PeerConnection by Steve Anton · 8 years ago
  2. bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 8 years ago
  3. 3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 8 years ago
  4. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 8 years ago
  5. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  6. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/pc/statscollector_unittest.cc]
  7. 0d0b912 Add and modify a few ANA stats. by ivoc · 8 years ago
  8. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 8 years ago
  9. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 8 years ago
  10. 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 8 years ago
  11. 539d104 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 8 years ago
  12. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 8 years ago
  13. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  14. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  15. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  16. 42308f6 Fix uploading of available send bitrate statistics. by Alex Narest · 8 years ago
  17. 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 · 8 years ago
  18. 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 · 8 years ago
  19. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 8 years ago
  20. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  21. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  22. cc452e1 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ ) by sakal · 8 years ago
  23. 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 9 years ago
  24. ff0e72f Add QP sum stats for received streams. by sakal · 9 years ago
  25. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 9 years ago
  26. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 9 years ago[Renamed (99%) from webrtc/api/statscollector_unittest.cc]
  27. c8ee882 Replace use of ASSERT in test code. by nisse · 9 years ago
  28. 84abeb1 RTC[In/Out]boundRTPStreamStats.mediaTrackId collected. by hbos · 9 years ago
  29. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 9 years ago
  30. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 9 years ago
  31. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 9 years ago
  32. df6075a RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 9 years ago
  33. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 9 years ago
  34. 49f34fd Relanding: Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 9 years ago
  35. 57fd726 Revert of Refactoring that removes P2PTransport and DtlsTransport classes. (patchset #9 id:150001 of https://codereview.webrtc.org/2517883002/ ) by deadbeef · 9 years ago
  36. bd28681 Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 9 years ago
  37. 87da404 Implement qpSum stat for video send ssrc stats. by sakal · 9 years ago
  38. e5ba44e Implement framesDecoded stat in video receive ssrc stats. by sakal · 9 years ago
  39. 43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 9 years ago
  40. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 9 years ago
  41. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 9 years ago
  42. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 9 years ago
  43. 77eab70 Enable the -Wundef warning for clang by kwiberg · 9 years ago
  44. 6348978 Add new decoding statistics for muted output by henrik.lundin · 9 years ago
  45. b24b1ce Moving mock classes around so that they may be reused in other unittests by hbos · 9 years ago
  46. 29ff844 Add PeerConnection IsClosed check. by zhihuang · 9 years ago
  47. e9021a3 Propogate network-worker thread split to api by danilchap · 9 years ago
  48. 6ba3b19 Filter out some variables with initial -1 in the stats report. by zhihuang · 9 years ago
  49. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
  50. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 9 years ago
  51. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 9 years ago
  52. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  53. b4d01c4 A bunch of interfaces: Return scoped_ptr<SSLCertificate> by kwiberg · 9 years ago
  54. 7d06a8c Add CoreVideoFrameBuffer. by tkchin · 9 years ago
  55. af510af Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests. by nisse · 9 years ago
  56. 51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago
  57. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
  58. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  59. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/statscollector_unittest.cc]
  60. a96e2d7 Move talk/media to webrtc/media by kjellander · 10 years ago
  61. bec70ab https://github.com/w3c/webrtc-stats/pull/10/files added mediaType to the tracks. The closest in the current stats is the ssrc type. by fippo · 10 years ago
  62. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 10 years ago
  63. 726b1f7 Removed dummy "mediastreamsignaling.h" by perkj · 10 years ago
  64. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 10 years ago
  65. 318166b Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) by guoweis · 10 years ago
  66. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 10 years ago
  67. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 10 years ago
  68. d59daf8 Merging BaseSession code into WebRtcSession. by deadbeef · 10 years ago
  69. ab9b2d1 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ ) by deadbeef · 10 years ago
  70. fc648b6 Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ ) by deadbeef · 10 years ago
  71. 97c3929 Moving MediaStreamSignaling logic into PeerConnection. by deadbeef · 10 years ago
  72. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 10 years ago
  73. 6caafbe Convert uint16_t to int for WebRTC cipher/crypto suite. by Guo-wei Shieh · 10 years ago
  74. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 10 years ago
  75. 27dc29b Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) by guoweis · 10 years ago
  76. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 10 years ago
  77. facbbec Remove use of DeviceManager from ChannelManager. by solenberg · 10 years ago
  78. cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 10 years ago
  79. a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 10 years ago
  80. 47ee2f3 TransportController refactoring. by deadbeef · 10 years ago
  81. 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 10 years ago
  82. 9af63f4 TransportController refactoring. by deadbeef · 10 years ago
  83. b071a19 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters. by Fredrik Solenberg · 10 years ago
  84. f3ecdb9 Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer. by Henrik Boström · 10 years ago
  85. d828198 Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer. by Henrik Boström · 10 years ago
  86. 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 10 years ago
  87. be24c94 Set / verify stats report timestamps. by jbauch · 10 years ago
  88. 8e6fd46 Route time-stretching metrics through libjingle by Henrik Lundin · 10 years ago
  89. 7fb711f Remove unused voice channel argument from cricket::VideoChannel ctor and corresponding field in class. by Fredrik Solenberg · 10 years ago
  90. 7c027b6 Enable more Clang warnings for talk/ by Henrik Kjellander · 10 years ago
  91. c04a97f Move from BaseSession::GetStats to WebRtcSession::GetTransportStats by pthatcher@webrtc.org · 10 years ago
  92. b01c707 Use a NULL session in unit tests that don't actually use the session. by pthatcher@webrtc.org · 10 years ago
  93. d390029 Use a variant for storing stats values in StatsCollector code. by tommi@webrtc.org · 10 years ago
  94. 92f4018 Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do. by tommi@webrtc.org · 10 years ago
  95. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  96. 058b1f1 Remove GetReceiveBandwidthEstimatorStats. by pbos@webrtc.org · 10 years ago
  97. 7bea1ff Expose negotiated ciphers through stats API. by pthatcher@webrtc.org · 10 years ago
  98. 1ed6224 Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info." by pbos@webrtc.org · 10 years ago
  99. 8ad05b7 Remove dead stats from Video{Sender,Receiver}Info. by pbos@webrtc.org · 10 years ago
  100. 652bc37 Adding two new stats to StatsReport. by minyue@webrtc.org · 10 years ago