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gerrit-public.fairphone.software / platform / external / webrtc / a46bd4b9c745e057b7a8d502bb9a914958642f4d / pc / statscollector.cc
  1. 70473fc Reland "Add hugeFramesSent GetStats metric" by Ilya Nikolaevskiy · 7 years ago
  2. 8ddc2e6 Revert "Add hugeFramesSent GetStats metric" by Max Morin · 7 years ago
  3. f9f71b9 Add hugeFramesSent GetStats metric by Ilya Nikolaevskiy · 7 years ago
  4. 4ff5443 Fix bugs in collecting STUN candidate stats and configuring STUN by Qingsi Wang · 7 years ago
  5. c392866 Implement certificate chain stats. by Taylor Brandstetter · 7 years ago
  6. 72a43a1 Collect packet loss and RTT stats of STUN binding requests. by Qingsi Wang · 7 years ago
  7. afb0bb7 Remove PeerConnection voice_channel/video_channel methods by Steve Anton · 7 years ago
  8. b886711 Prepare StatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
  9. 5dfde18 Change PeerConnection stats interface to be more flexible by Steve Anton · 7 years ago
  10. 2d8609c Move internal PeerConnection methods to PeerConnectionInternal by Steve Anton · 7 years ago
  11. 75ceef2 Pivot old stats generation to iterate senders/receivers by Harald Alvestrand · 7 years ago
  12. 7411648 Remove SessionStats.proxy_to_transport by Steve Anton · 7 years ago
  13. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  14. ae02609 Add parallel stats interface with optional stats to APM. by Ivo Creusen · 7 years ago
  15. c3ed630 Add stats googHasEnteredLowResolution. by Åsa Persson · 7 years ago
  16. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  17. 75737c0 Merge WebRtcSession into PeerConnection by Steve Anton · 7 years ago
  18. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  19. d5585ca Move almost all references from WebRtcSession to PeerConnection by Steve Anton · 7 years ago
  20. 978b876 Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  21. bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 7 years ago
  22. 3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  23. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  24. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/statscollector.cc]
  25. 0d0b912 Add and modify a few ANA stats. by ivoc · 7 years ago
  26. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  27. 50864a8 Add reporting of googContentType via GetStats on send side by ilnik · 7 years ago
  28. 2e1b40b Implement googContentType GetStats metric reported on receive side. by ilnik · 7 years ago
  29. 0e320ec Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 7 years ago
  30. a79cc28 Report max interframe delay over window insdead of interframe delay sum by ilnik · 7 years ago
  31. 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
  32. f04afde Report interframe delay sum in old GetStats by ilnik · 8 years ago
  33. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  34. 2edc684 Report timing frames info in GetStats. by ilnik · 8 years ago
  35. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  36. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  37. 42308f6 Fix uploading of available send bitrate statistics. by Alex Narest · 8 years ago
  38. 19b3a55 Fixing incorrect use of erase/remove idiom. by deadbeef · 8 years ago
  39. 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
  40. 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
  41. 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
  42. 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
  43. 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
  44. ff0e72f Add QP sum stats for received streams. by sakal · 8 years ago
  45. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/statscollector.cc]
  46. 04a057b Add missing if-clause for residual_echo_likelihood_recent_max by henrik.lundin · 8 years ago
  47. 4e477a1 Added a new echo likelihood stat that reports the maximum value from a previous time period. by ivoc · 8 years ago
  48. eb4ca4e Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  49. df6075a RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 8 years ago
  50. ba7e71b remove googViewLimitedResolution stat by philipp.hancke · 8 years ago
  51. e9e94c3 Return false if PeerConnection::GetStats() is called on invalid tracks by zhihuang · 8 years ago
  52. 87da404 Implement qpSum stat for video send ssrc stats. by sakal · 8 years ago
  53. e5ba44e Implement framesDecoded stat in video receive ssrc stats. by sakal · 8 years ago
  54. 43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 8 years ago
  55. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  56. 5377bc7 Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere by kwiberg · 8 years ago
  57. 8f90106 Revert of Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere (patchset #2 id:20001 of https://codereview.webrtc.org/2384693002/ ) by guidou · 8 years ago
  58. ab0b929 Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere by kwiberg · 8 years ago
  59. 6348978 Add new decoding statistics for muted output by henrik.lundin · 8 years ago
  60. cdf37a9 Delete Timing class, timing.h, and update all users. by nisse · 8 years ago
  61. e29352b Refactor certificate stats collection, added SSLCertificateStats. by hbos · 8 years ago
  62. 5ecf16c Add Stats to Stun ping. by zhihuang · 9 years ago
  63. 6ba3b19 Filter out some variables with initial -1 in the stats report. by zhihuang · 9 years ago
  64. 555604a Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  65. 2a8a78c Add AEC filter divergence metric to StatsCollector. by Minyue · 9 years ago
  66. b4d01c4 A bunch of interfaces: Return scoped_ptr<SSLCertificate> by kwiberg · 9 years ago
  67. fcc640f Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector, by nisse · 9 years ago
  68. f5d4786 SSLCertificate::GetChain: Return scoped_ptr by kwiberg · 9 years ago
  69. 03d6d57 Late initialize MediaController, for less resource i.e. ProcessThread, usage by PeerConnection. by solenberg · 9 years ago
  70. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
  71. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  72. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (99%) from talk/app/webrtc/statscollector.cc]
  73. 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
  74. b7d9a97 Expose codec implementation names in stats. by Peter Boström · 9 years ago
  75. 2fe1cb0 Don't overwrite audio stats when they're not available. by andrew · 9 years ago
  76. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  77. 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
  78. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  79. d59daf8 Merging BaseSession code into WebRtcSession. by deadbeef · 9 years ago
  80. ab9b2d1 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ ) by deadbeef · 9 years ago
  81. fc648b6 Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ ) by deadbeef · 9 years ago
  82. 97c3929 Moving MediaStreamSignaling logic into PeerConnection. by deadbeef · 9 years ago
  83. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  84. 6caafbe Convert uint16_t to int for WebRTC cipher/crypto suite. by Guo-wei Shieh · 9 years ago
  85. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  86. 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
  87. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  88. cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 9 years ago
  89. a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 9 years ago
  90. 47ee2f3 TransportController refactoring. by deadbeef · 9 years ago
  91. 04ac81f Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet). by Peter Thatcher · 9 years ago
  92. 275a2f1 Revert of Replace readable with receiving where receiving means receiving anything (stun ping, response or da… (patchset #7 id:340001 of https://codereview.webrtc.org/1351673003/ ) by tommi · 9 years ago
  93. ae16f85 Replace readable with receiving where receiving means receiving anything (stun ping, response or data packet). by honghaiz · 9 years ago
  94. 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 9 years ago
  95. 9af63f4 TransportController refactoring. by deadbeef · 9 years ago
  96. 91d6ede Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 9 years ago
  97. f3ecdb9 Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer. by Henrik Boström · 9 years ago
  98. f42376c Wire up currently-received video codec to stats. by pbos · 9 years ago
  99. d828198 Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer. by Henrik Boström · 9 years ago
  100. be24c94 Set / verify stats report timestamps. by jbauch · 10 years ago