Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 3b50f9f9ceb17d98964407277e13658bfbed4541 / pc / rtc_stats_integrationtest.cc
  1. 0237106 Expose video freeze metrics in GetStats. by Sergey Silkin · 7 years ago
  2. 64b626b Use Abseil container algorithms in pc/ by Steve Anton · 7 years ago
  3. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 7 years ago
  4. 0acffb5 Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`. by Chen Xing · 7 years ago
  5. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 7 years ago
  6. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 7 years ago[Renamed from pc/rtcstats_integrationtest.cc]
  7. 68586e8 Replace starts_with and ends_with with Abseil by Steve Anton · 7 years ago
  8. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 7 years ago
  9. 352ce5c Expose delayed packet outage as a cumulative metric of samples in the new getStats API. by Jakob Ivarsson · 7 years ago
  10. 59cfd35 Address vptr race condition while PeerConnection is destructed. by Yves Gerey · 7 years ago
  11. 8af8896 Expose jitter buffer flushes metric in new getStats api. by Ruslan Burakov · 7 years ago
  12. b5bb513 Disable RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection by Yves Gerey · 7 years ago
  13. 9551375 getStats: add relayProtocol by Philipp Hancke · 7 years ago
  14. 3bc0166 getStats: add kind alias for mediaType by Philipp Hancke · 7 years ago
  15. f06f923 Delete almost all use of MediaConstraintsInterface in the PeerConnection API by Niels Möller · 7 years ago
  16. e12c1fe Removing warning suppression flags from pc/. by Mirko Bonadei · 7 years ago
  17. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  18. 2d02e08 Delete deprecated CreateAudioSource method, with constraints. by Niels Möller · 7 years ago
  19. 25cfeb9 Fix possible race between the stats collector and transport controller by Steve Anton · 7 years ago
  20. 1df1bf8 PeerConnectionInterface::GetStats() with selector argument added. by Henrik Boström · 7 years ago
  21. b619936 Stats traversal algorithm added. by Henrik Boström · 7 years ago
  22. 70473fc Reland "Add hugeFramesSent GetStats metric" by Ilya Nikolaevskiy · 7 years ago
  23. 8ddc2e6 Revert "Add hugeFramesSent GetStats metric" by Max Morin · 7 years ago
  24. f9f71b9 Add hugeFramesSent GetStats metric by Ilya Nikolaevskiy · 7 years ago
  25. 719487e Generate signed packets_lost in WebRTC-stats by Harald Alvestrand · 8 years ago
  26. 37e489c Add network_type to local RTCIceCandidateStats by Gary Liu · 8 years ago
  27. c61ce0d Fixing some clang-tidy findings. by Mirko Bonadei · 8 years ago
  28. 36b29d1 Enable cpplint in pc/ by Steve Anton · 8 years ago
  29. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 8 years ago
  30. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  31. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  32. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/pc/rtcstats_integrationtest.cc]
  33. 8ab0fd8 Reland of Trace the stats report as JSON instead of each stat separately. (patchset #1 id:1 of https://codereview.webrtc.org/3001683002/ ) by ehmaldonado · 8 years ago
  34. 2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 8 years ago
  35. 3439c89 Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ ) by mbonadei · 8 years ago
  36. 80c6578 Trace the stats report as JSON instead of each stat separately. by ehmaldonado · 8 years ago
  37. 35a872c Make RTCStatsReport::ToString() return JSON-parseable string. by ehmaldonado · 8 years ago
  38. 80c829f Enable tracing on rtcstats_integrationtest.cc by ehmaldonado · 8 years ago
  39. e725159 Reland of Make the default ctor of rtc::Thread, protected by tommi · 8 years ago
  40. 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 · 8 years ago
  41. a117b04 Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ ) by charujain · 8 years ago
  42. a8a3515 Make the default ctor of rtc::Thread, protected. by tommi · 8 years ago
  43. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  44. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  45. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  46. 86c40a1 Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent. by deadbeef · 8 years ago
  47. 6488ea4 Remove temporary include of builtin_audio_encoder_factory.h. by ossu · 8 years ago
  48. 98e186c Remove VirtualSocketServer's dependency on PhysicalSocketServer. by deadbeef · 8 years ago
  49. 9e5b11e Test CreatePeerConnectionFactory() with a forwarding mock AudioDecoderFactory by kwiberg · 8 years ago
  50. a7a9be1 Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect. by hbos · 8 years ago
  51. 13f54b2 Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine. by hbos · 8 years ago
  52. bf8d3e5 RTCIceCandidatePairStats.[total/current]RoundTripTime collected. by hbos · 8 years ago
  53. 3fd31fe Fix TSAN race in webrtc::voe::Channel. by hbos · 8 years ago
  54. 92eaec6 RTCIceCandidatePairStats.nominated collected. by hbos · 8 years ago
  55. 1498def Disable RTCStatsIntegrationTest on TSAN bots. by philipel · 8 years ago
  56. 5fec128 Add QP for libvpx VP8 decoder. by sakal · 8 years ago
  57. a51d4f3 Re-land of RTCInboundRTPStreamStats.qpSum collected. by hbos · 8 years ago
  58. ed02c6d Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ ) by skvlad · 9 years ago
  59. cd195be RTCInboundRTPStreamStats.qpSum collected. by hbos · 9 years ago
  60. 338f78a RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. by hbos · 9 years ago
  61. b0ae920 RTCRTPStreamStats.mediaTrackId renamed to trackId. by hbos · 9 years ago
  62. 50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 9 years ago
  63. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 9 years ago[Renamed (99%) from webrtc/api/rtcstats_integrationtest.cc]
  64. f64941f RTCMediaStreamTrackStats.framesDecoded collected. by hbos · 9 years ago
  65. fefe076 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector. by hbos · 9 years ago
  66. 2d4d653 Fix msan flake in rtcstats_integrationtest.cc. by hbos · 9 years ago
  67. 42f6d2f RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 9 years ago
  68. 9e30274 RTCMediaStreamTrackStats collected on a per-attachment basis. by hbos · 9 years ago
  69. 160e4a7 RTCMediaStreamTrackStats.kind added and collected. by hbos · 9 years ago
  70. 7064d59 RTCTransportStats.dtlsState replaces .activeConnection by hbos · 9 years ago
  71. 84abeb1 RTC[In/Out]boundRTPStreamStats.mediaTrackId collected. by hbos · 9 years ago
  72. bb37751 RTCMediaStreamTrackStats.ssrcIds removed. by hbos · 9 years ago
  73. b4e426e RTCIceCandidateStats.transportId added and collected. by hbos · 9 years ago
  74. 6769c49 RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added. by hbos · 9 years ago
  75. d17a5a7 RTCIceCandidateStats.deleted = false added. by hbos · 9 years ago
  76. 06495bc RTCIceCandidatePairStats.[state/priority] added, ConnectionInfo updated. by hbos · 9 years ago
  77. 2335119 Improve rtcstats_integrationtest.cc by sanity checking values. by hbos · 9 years ago
  78. c3a2b7f RTCIceCandidateStats.isRemote added and collected. by hbos · 9 years ago
  79. 40610e2 Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. by deadbeef · 9 years ago
  80. b78306a Fix segfault when PeerConnection is destroyed during stats collection. by hbos · 9 years ago
  81. 3168c7a Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time. by hbos · 9 years ago
  82. 7bf5369 RTCStatsIntegrationTest: TestMemberIsIDReference on all defined IDs. by hbos · 9 years ago
  83. e10e6d1 RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values. by hbos · 9 years ago
  84. e448dd5 RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector by hbos · 9 years ago
  85. 02cd4d6 RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector. by hbos · 9 years ago
  86. d82f512 RTCIceCandidatePairStats.requestsReceived defined by RTCStatsCollector. by hbos · 9 years ago
  87. 0583b28 Collecting RTCIceCandidatePairStats.transport_id and improved unittests. by hbos · 9 years ago
  88. db346a7 RTCStatsIntegrationTest added. by hbos · 9 years ago