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gerrit-public.fairphone.software / platform / external / webrtc / 2ae140ae7e34bd1e102ca832a14b7b0adbb72d15 / pc / statscollector_unittest.cc
  1. 5b38731 Use fake PeerConnection for RTCStatsCollector tests by Steve Anton · 7 years ago
  2. 3871f6f Rewrite StatsCollector tests to use a fake PeerConnection by Steve Anton · 7 years ago
  3. be5e208 Add FakePeerConnectionBase by Steve Anton · 7 years ago
  4. 75ceef2 Pivot old stats generation to iterate senders/receivers by Harald Alvestrand · 7 years ago
  5. 7411648 Remove SessionStats.proxy_to_transport by Steve Anton · 7 years ago
  6. 7c5597a Remove unused enum (kStatsValueNameEchoCancellationQualityMin). by Gustaf Ullberg · 7 years ago
  7. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  8. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  9. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 7 years ago
  10. ae02609 Add parallel stats interface with optional stats to APM. by Ivo Creusen · 7 years ago
  11. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  12. 75737c0 Merge WebRtcSession into PeerConnection by Steve Anton · 7 years ago
  13. ba81867 Prepare WebRtcSession to be merged into PeerConnection by Steve Anton · 7 years ago
  14. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  15. 978b876 Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  16. bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 7 years ago
  17. 3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  18. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  19. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  20. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/statscollector_unittest.cc]
  21. 0d0b912 Add and modify a few ANA stats. by ivoc · 7 years ago
  22. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  23. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  24. 773be36 Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 7 years ago
  25. 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
  26. f1377f7 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 7 years ago
  27. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  28. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  29. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  30. 42308f6 Fix uploading of available send bitrate statistics. by Alex Narest · 7 years ago
  31. 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
  32. 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
  33. 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
  34. eaabdf6 Delete MediaController class, move Call ownership to PeerConnection. by nisse · 7 years ago
  35. 112b2e9 Switching some interfaces to use std::unique_ptr<>. by deadbeef · 8 years ago
  36. 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
  37. 69fb2cc Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ ) by skvlad · 8 years ago
  38. ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
  39. f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
  40. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/statscollector_unittest.cc]
  41. c8ee882 Replace use of ASSERT in test code. by nisse · 8 years ago
  42. 84abeb1 RTC[In/Out]boundRTPStreamStats.mediaTrackId collected. by hbos · 8 years ago
  43. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  44. ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
  45. f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
  46. df6075a RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 8 years ago
  47. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  48. 49f34fd Relanding: Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  49. 57fd726 Revert of Refactoring that removes P2PTransport and DtlsTransport classes. (patchset #9 id:150001 of https://codereview.webrtc.org/2517883002/ ) by deadbeef · 8 years ago
  50. bd28681 Refactoring that removes P2PTransport and DtlsTransport classes. by deadbeef · 8 years ago
  51. 87da404 Implement qpSum stat for video send ssrc stats. by sakal · 8 years ago
  52. e5ba44e Implement framesDecoded stat in video receive ssrc stats. by sakal · 8 years ago
  53. 43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 8 years ago
  54. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  55. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  56. ac9f876 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ by kwiberg · 8 years ago
  57. 77eab70 Enable the -Wundef warning for clang by kwiberg · 8 years ago
  58. 6348978 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  59. b24b1ce Moving mock classes around so that they may be reused in other unittests by hbos · 8 years ago
  60. 29ff844 Add PeerConnection IsClosed check. by zhihuang · 8 years ago
  61. e9021a3 Propogate network-worker thread split to api by danilchap · 8 years ago
  62. 6ba3b19 Filter out some variables with initial -1 in the stats report. by zhihuang · 8 years ago
  63. 33b01f2 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 8 years ago
  64. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 8 years ago
  65. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 8 years ago
  66. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 8 years ago
  67. b4d01c4 A bunch of interfaces: Return scoped_ptr<SSLCertificate> by kwiberg · 8 years ago
  68. 7d06a8c Add CoreVideoFrameBuffer. by tkchin · 8 years ago
  69. af510af Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests. by nisse · 8 years ago
  70. 51542be Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago
  71. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
  72. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  73. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/statscollector_unittest.cc]
  74. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  75. 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 · 9 years ago
  76. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  77. 726b1f7 Removed dummy "mediastreamsignaling.h" by perkj · 9 years ago
  78. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  79. 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 · 9 years ago
  80. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  81. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  82. d59daf8 Merging BaseSession code into WebRtcSession. by deadbeef · 9 years ago
  83. ab9b2d1 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ ) by deadbeef · 9 years ago
  84. fc648b6 Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ ) by deadbeef · 9 years ago
  85. 97c3929 Moving MediaStreamSignaling logic into PeerConnection. by deadbeef · 9 years ago
  86. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  87. 6caafbe Convert uint16_t to int for WebRTC cipher/crypto suite. by Guo-wei Shieh · 9 years ago
  88. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  89. 27dc29b Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) by guoweis · 9 years ago
  90. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  91. facbbec Remove use of DeviceManager from ChannelManager. by solenberg · 9 years ago
  92. cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 9 years ago
  93. a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 9 years ago
  94. 47ee2f3 TransportController refactoring. by deadbeef · 9 years ago
  95. 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 9 years ago
  96. 9af63f4 TransportController refactoring. by deadbeef · 9 years ago
  97. b071a19 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters. by Fredrik Solenberg · 9 years ago
  98. f3ecdb9 Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer. by Henrik Boström · 9 years ago
  99. d828198 Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer. by Henrik Boström · 9 years ago
  100. 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 9 years ago