Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 2edc6845ac12a7216aeb84d9ad9da8260e2bb434
  1. 2edc684 Report timing frames info in GetStats. by ilnik · 7 years ago
  2. 5b7fc8c A few simplifications to CodecDatabase and VCMGenericDecoder. by tommi · 7 years ago
  3. 7025244 Roll chromium_revision f45f1f992e..c33c6bfd24 (484285:484321) by buildbot · 7 years ago
  4. 6aa9511 Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable by bdodson · 7 years ago
  5. e4f63a1 Roll chromium_revision c01b31617b..f45f1f992e (484252:484285) by buildbot · 7 years ago
  6. b16a01f Revert "Reland "Adding ANA config event to debug dump."" by minyue-webrtc · 7 years ago
  7. 63d146b NetEq: Rectify the implementation of PacketBuffer::DiscardOldPackets by henrik.lundin · 7 years ago
  8. 440ea8c Roll chromium_revision 3fe2409358..c01b31617b (484231:484252) by buildbot · 7 years ago
  9. 191113a Added implementation of four functions in the BBR congestion controller: by gnish · 7 years ago
  10. bc0c4f5 Roll chromium_revision 6da2ebcead..3fe2409358 (484119:484231) by buildbot · 7 years ago
  11. fae474c Implement packet discard rate in NetEq. by minyue-webrtc · 7 years ago
  12. 889d965 Fix issue with zero rtt reports when using FlexFEC and add perf test. by stefan · 7 years ago
  13. 070efc0 Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat by henrika · 7 years ago
  14. abee2d8 Roll chromium_revision 2fe6dc66f8..6da2ebcead (484092:484119) by buildbot · 7 years ago
  15. 0fc6d87 Roll chromium_revision cf58257d56..2fe6dc66f8 (483646:484092) by buildbot · 7 years ago
  16. f720704 Added philipel@webrtc.org to webrtc/modules/remote_bitrate_estimator/OWNERS. by philipel · 7 years ago
  17. cb576c5 Fixes build issue based on usage of Android O specific API by henrika · 7 years ago
  18. c43f68e Fix do not unregister bluetooth receiver if it was not registered by Gustavo Garcia · 7 years ago
  19. cc8856c Remove unused static VideoEncoder functions by magjed · 7 years ago
  20. f612998 Override bots to use libstdc++ on Linux by oprypin · 7 years ago
  21. 8eadead Adds support for USB audio devices in AppRTCMobile on Android. by henrika · 7 years ago
  22. a9521e2 Reduce send rate to 50% if overusing before we have an acknowledged bitrate. by terelius · 7 years ago
  23. 2c3161c Changed default value for the duration of the echo in echocanceller 3 by peah · 7 years ago
  24. 0d7f04d Reland of Add received audio/video call duration metrics based on packets. by saza · 7 years ago
  25. 38fecaf Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ ) by ehmaldonado · 7 years ago
  26. d3588cf Improved low-level echo handling in echo canceller 3 by peah · 7 years ago
  27. ed56680 Remove webrtc/tools by ehmaldonado · 7 years ago
  28. 382f21c Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ ) by saza · 7 years ago
  29. 7467492 Add received audio/video call duration metrics based on packets. by saza · 7 years ago
  30. 2a2b297 Add underscore at end of Call members' names by eladalon · 7 years ago
  31. 4235d78 Disabling flaky complexity tests for the audio processing module. by peah · 7 years ago
  32. 7ab7fd6 Fix gmock warnings emanating from FlexfecReceiveStreamTest by eladalon · 7 years ago
  33. 7c7796b Register FlexFEC SSRC to receive RTCP on sending side. by brandtr · 7 years ago
  34. 48587f9 Changing AudioConferenceMixer logging to base/logging.h by Alex Loiko · 7 years ago
  35. 4257ab2 Add received interframe delay UMA metrics by ilnik · 7 years ago
  36. ad515c4 fix comment length by jansson · 7 years ago
  37. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  38. dca1e09 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" by Henrik Kjellander · 7 years ago
  39. c8fa692 Update includes for webrtc/{base => rtc_base} rename (1/3) by kjellander · 7 years ago
  40. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  41. 89c4a7e Wire up experiment for improved screenshare bwe. by sprang · 7 years ago
  42. e96c45b Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)" by kjellander · 7 years ago
  43. 224e659 Reland of "VideoFrameBuffer: Remove deprecated functions" by Magnus Jedvert · 7 years ago
  44. e75d96b Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ ) by terelius · 7 years ago
  45. 2d54784 Reland "Adding ANA config event to debug dump." by minyue-webrtc · 7 years ago
  46. 17675ce Enable the injection of an APM into a peerconnection by peah · 7 years ago
  47. fa85678 Fix FecTest.FlexfecTest flakiness caused by seq. num. wraparound. by brandtr · 7 years ago
  48. 697a590 Added the ability to adjust the AEC3 performance for large rooms by peah · 7 years ago
  49. 4583db4 Enable -Wunused-function warning everywhere. by Henrik Kjellander · 7 years ago
  50. 1009cfc More gracefully handle rtp timestamp jumps in the rtp to ntp estimator. by stefan · 7 years ago
  51. 9addbeb Remove RtpDemuxer tweak for preventing multiple RSID inspections by eladalon · 7 years ago
  52. 49085ef Improves audio-routing in combination with BT in AppRTCMobile on Android. by henrika · 7 years ago
  53. 0072511 Revert "Update includes for webrtc/{base => rtc_base} rename (3/3)" by Henrik Kjellander · 7 years ago
  54. f1c5ebf Update includes for webrtc/{base => rtc_base} rename (3/3) by kjellander · 7 years ago
  55. 96d74bb Opus implementation of the AudioDecoderFactoryTemplate API by kwiberg · 7 years ago
  56. 3aba2d1 Fix android video_quality_loopback_test by ehmaldonado · 7 years ago
  57. d76b753 Disable AudioDeviceTest.StartStopRecording on iOS by henrika · 7 years ago
  58. 96da011 Opus implementation of the AudioEncoderFactoryTemplate API by kwiberg · 7 years ago
  59. 41bafb2 Update PRESUBMIT.py for webrtc/{tools => rtc_tools} rename. by kjellander · 7 years ago
  60. 9aed31c Temporarily removed the analog gain change detection in AEC3 by Per Åhgren · 7 years ago
  61. 8f9ce1d Corrected the limit on the allowed API jitter in AEC3 by peah · 7 years ago
  62. d2b63cf Move webrtc/{tools => rtc_tools} by kjellander · 7 years ago
  63. cb8f045 Fix receiving FlexFEC in video_loopback. by brandtr · 7 years ago
  64. 5f8b04d Higher logging severity for RED packets in UlpfecReceiverImpl. by brandtr · 7 years ago
  65. 2800be3 Roll chromium_revision 7aa4e8bf36..cf58257d56 (483375:483646) by buildbot · 7 years ago
  66. 1129df2 Always ResetSenderCongestionControlObjects before RegisterEtc... by ossu · 7 years ago
  67. 88af8b4 Fix -Wcomment warning in webrtcsdp.cc by kjellander · 7 years ago
  68. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  69. c9be3d5 Roll chromium_revision 3691cc167a..7aa4e8bf36 (483339:483375) by buildbot · 7 years ago
  70. 26afe21 Properly export the symbols of video frame-buffer classes for link-time by VladimirTechMan · 7 years ago
  71. 06b47c5 Listen for Wifi-Direct networks and include them in the network list by bdodson · 7 years ago
  72. 8cf398d Roll chromium_revision 53b56ec80b..3691cc167a (483312:483339) by buildbot · 7 years ago
  73. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 7 years ago
  74. 3dbfac3 Fix two simple type mismatches thay may cause compilation issues on win. by sprang · 7 years ago
  75. 960fd5b Adding a presubmit check for .proto files EOF newline by Mirko Bonadei · 7 years ago
  76. f1e3483 Revert "VideoFrameBuffer: Remove deprecated functions" by Magnus Jedvert · 7 years ago
  77. bc8feda Delete SignalThread class. by nisse · 7 years ago
  78. 428c9e2 VideoFrameBuffer: Remove deprecated functions by Magnus Jedvert · 7 years ago
  79. 57ca81a Actually use virtual network in OrtcFactory unit test. by deadbeef · 7 years ago
  80. 8a90f87 Add SetCodecSettings method for configuring VideoCodec settings. by asapersson · 7 years ago
  81. 17432ec Add magjed@ as owner of webrtc/api/video/ by Magnus Jedvert · 7 years ago
  82. 26b1f92 Roll chromium_revision 9dd69e9f64..53b56ec80b (483005:483312) by buildbot · 7 years ago
  83. d726a3f Reland of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #1 id:1 of https://codereview.webrtc.org/2919313005/ ) by brandtr · 7 years ago
  84. e435019 Don't disable FEC if timing frames are disabled. by ilnik · 7 years ago
  85. 8c1ee7b Simplifies StartStopRecording test on iOS. by henrika · 7 years ago
  86. 8d08a92 Do not copy I420 frames in the decoder when not necessary. by Sami Kalliomäki · 7 years ago
  87. b14fad4 Adding newline at the end of .proto files by Mirko Bonadei · 7 years ago
  88. f4efb6f Reland "Move webrtc/{base => rtc_base} (stub headers) by Henrik Kjellander · 7 years ago
  89. c036276 Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002) by Henrik Kjellander · 7 years ago
  90. ec78f1c Revert "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002) by Henrik Kjellander · 7 years ago
  91. a4c113a Support building WebRTC without audio and video for IOS. by zhihuang · 7 years ago
  92. 9588682 Update memcheck suppression for HttpServer.SignalsCloseAfterForcedCloseAll by Henrik Kjellander · 7 years ago
  93. 9b808e7 Update TSan suppressions for base->rtc_base rename by Henrik Kjellander · 7 years ago
  94. 6776518 Move webrtc/{base => rtc_base} by Henrik Kjellander · 7 years ago
  95. e0eb35d Roll chromium_revision 8af690d4cd..9dd69e9f64 (482924:483005) by buildbot · 7 years ago
  96. 86c40a1 Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent. by deadbeef · 7 years ago
  97. c3e3e60 nit: Rename RtpDemuxer::sink_ to RtpDemuxer::ssrc_sinks_ by eladalon · 7 years ago
  98. 9f789a4 LowCutFilter::BiqueadFilter::Process: Fix UBSan fuzzer bug by Alex Loiko · 7 years ago
  99. d6e9466 No compliation-flag-dependent members in CriticalSection by eladalon · 7 years ago
  100. 3d0e7bb Improved thread checking scheme for iOS. by henrika · 7 years ago