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webrtc / src / 5b361730d0776fba4c6f1324a9dc1efcc148cdba
  1. 5b36173 Support re-entrant calls to MessageQueueManager::Clear. by jbauch · 8 years ago
  2. 876088a Roll chromium_revision baaa9eae93..4b357464fd (484696:484824) by buildbot · 8 years ago
  3. 4a494ff desktop_capture: crop border in window_capture on Win8/10 by braveyao · 8 years ago
  4. f07e6b4 Roll chromium_revision 2e0945b687..baaa9eae93 (484611:484696) by buildbot · 8 years ago
  5. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  6. ea39dfa Roll chromium_revision c33c6bfd24..2e0945b687 (484321:484611) by buildbot · 8 years ago
  7. 9e3f1e4 Fixed a miscalculation of sent bitrate caused by mixup of time units by Sebastian Jansson · 8 years ago
  8. d66072b Moving asm code out of common_audio_c sources list by mbonadei · 8 years ago
  9. 3b03476 Remove MAIN_NIB_FILE from Info.plist because the substitution is broken by oprypin · 8 years ago
  10. a449107 Let NetEq reset the AudioFrame during muted state by henrik.lundin · 8 years ago
  11. 02569ad Update screen simulcast config by sprang · 8 years ago
  12. 168794c Implement RTP keepalive in native stack. by sprang · 8 years ago
  13. 5c0d703 Moving asm code out of isac_fix_c sources list by mbonadei · 8 years ago
  14. 05db21d Reland of move webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2973493002/ ) by ehmaldonado · 8 years ago
  15. 2edc684 Report timing frames info in GetStats. by ilnik · 8 years ago
  16. 5b7fc8c A few simplifications to CodecDatabase and VCMGenericDecoder. by tommi · 8 years ago
  17. 7025244 Roll chromium_revision f45f1f992e..c33c6bfd24 (484285:484321) by buildbot · 8 years ago
  18. 6aa9511 Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable by bdodson · 8 years ago
  19. e4f63a1 Roll chromium_revision c01b31617b..f45f1f992e (484252:484285) by buildbot · 8 years ago
  20. b16a01f Revert "Reland "Adding ANA config event to debug dump."" by minyue-webrtc · 8 years ago
  21. 63d146b NetEq: Rectify the implementation of PacketBuffer::DiscardOldPackets by henrik.lundin · 8 years ago
  22. 440ea8c Roll chromium_revision 3fe2409358..c01b31617b (484231:484252) by buildbot · 8 years ago
  23. 191113a Added implementation of four functions in the BBR congestion controller: by gnish · 8 years ago
  24. bc0c4f5 Roll chromium_revision 6da2ebcead..3fe2409358 (484119:484231) by buildbot · 8 years ago
  25. fae474c Implement packet discard rate in NetEq. by minyue-webrtc · 8 years ago
  26. 889d965 Fix issue with zero rtt reports when using FlexFEC and add perf test. by stefan · 8 years ago
  27. 070efc0 Improves WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches UMA stat by henrika · 8 years ago
  28. abee2d8 Roll chromium_revision 2fe6dc66f8..6da2ebcead (484092:484119) by buildbot · 8 years ago
  29. 0fc6d87 Roll chromium_revision cf58257d56..2fe6dc66f8 (483646:484092) by buildbot · 8 years ago
  30. f720704 Added philipel@webrtc.org to webrtc/modules/remote_bitrate_estimator/OWNERS. by philipel · 8 years ago
  31. cb576c5 Fixes build issue based on usage of Android O specific API by henrika · 8 years ago
  32. c43f68e Fix do not unregister bluetooth receiver if it was not registered by Gustavo Garcia · 8 years ago
  33. cc8856c Remove unused static VideoEncoder functions by magjed · 8 years ago
  34. f612998 Override bots to use libstdc++ on Linux by oprypin · 8 years ago
  35. 8eadead Adds support for USB audio devices in AppRTCMobile on Android. by henrika · 8 years ago
  36. a9521e24 Reduce send rate to 50% if overusing before we have an acknowledged bitrate. by terelius · 8 years ago
  37. 2c3161c Changed default value for the duration of the echo in echocanceller 3 by peah · 8 years ago
  38. 0d7f04d Reland of Add received audio/video call duration metrics based on packets. by saza · 8 years ago
  39. 38fecaf Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ ) by ehmaldonado · 8 years ago
  40. d3588cf Improved low-level echo handling in echo canceller 3 by peah · 8 years ago
  41. ed56680 Remove webrtc/tools by ehmaldonado · 8 years ago
  42. 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 · 8 years ago
  43. 7467492 Add received audio/video call duration metrics based on packets. by saza · 8 years ago
  44. 2a2b297 Add underscore at end of Call members' names by eladalon · 8 years ago
  45. 4235d78 Disabling flaky complexity tests for the audio processing module. by peah · 8 years ago
  46. 7ab7fd6 Fix gmock warnings emanating from FlexfecReceiveStreamTest by eladalon · 8 years ago
  47. 7c7796b Register FlexFEC SSRC to receive RTCP on sending side. by brandtr · 8 years ago
  48. 48587f9 Changing AudioConferenceMixer logging to base/logging.h by Alex Loiko · 8 years ago
  49. 4257ab2 Add received interframe delay UMA metrics by ilnik · 8 years ago
  50. ad515c4 fix comment length by jansson · 8 years ago
  51. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  52. dca1e09 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" by Henrik Kjellander · 8 years ago
  53. c8fa692 Update includes for webrtc/{base => rtc_base} rename (1/3) by kjellander · 8 years ago
  54. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  55. 89c4a7e Wire up experiment for improved screenshare bwe. by sprang · 8 years ago
  56. e96c45b Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)" by kjellander · 8 years ago
  57. 224e659 Reland of "VideoFrameBuffer: Remove deprecated functions" by Magnus Jedvert · 8 years ago
  58. 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 · 8 years ago
  59. 2d54784 Reland "Adding ANA config event to debug dump." by minyue-webrtc · 8 years ago
  60. 17675ce Enable the injection of an APM into a peerconnection by peah · 8 years ago
  61. fa85678 Fix FecTest.FlexfecTest flakiness caused by seq. num. wraparound. by brandtr · 8 years ago
  62. 697a590 Added the ability to adjust the AEC3 performance for large rooms by peah · 8 years ago
  63. 4583db4 Enable -Wunused-function warning everywhere. by Henrik Kjellander · 8 years ago
  64. 1009cfc More gracefully handle rtp timestamp jumps in the rtp to ntp estimator. by stefan · 8 years ago
  65. 9addbeb Remove RtpDemuxer tweak for preventing multiple RSID inspections by eladalon · 8 years ago
  66. 49085ef Improves audio-routing in combination with BT in AppRTCMobile on Android. by henrika · 8 years ago
  67. 0072511 Revert "Update includes for webrtc/{base => rtc_base} rename (3/3)" by Henrik Kjellander · 8 years ago
  68. f1c5ebf Update includes for webrtc/{base => rtc_base} rename (3/3) by kjellander · 8 years ago
  69. 96d74bb Opus implementation of the AudioDecoderFactoryTemplate API by kwiberg · 8 years ago
  70. 3aba2d1a Fix android video_quality_loopback_test by ehmaldonado · 8 years ago
  71. d76b753 Disable AudioDeviceTest.StartStopRecording on iOS by henrika · 8 years ago
  72. 96da011 Opus implementation of the AudioEncoderFactoryTemplate API by kwiberg · 8 years ago
  73. 41bafb2 Update PRESUBMIT.py for webrtc/{tools => rtc_tools} rename. by kjellander · 8 years ago
  74. 9aed31c Temporarily removed the analog gain change detection in AEC3 by Per Åhgren · 8 years ago
  75. 8f9ce1d Corrected the limit on the allowed API jitter in AEC3 by peah · 8 years ago
  76. d2b63cf Move webrtc/{tools => rtc_tools} by kjellander · 8 years ago
  77. cb8f045 Fix receiving FlexFEC in video_loopback. by brandtr · 8 years ago
  78. 5f8b04d Higher logging severity for RED packets in UlpfecReceiverImpl. by brandtr · 8 years ago
  79. 2800be3 Roll chromium_revision 7aa4e8bf36..cf58257d56 (483375:483646) by buildbot · 8 years ago
  80. 1129df2 Always ResetSenderCongestionControlObjects before RegisterEtc... by ossu · 8 years ago
  81. 88af8b4 Fix -Wcomment warning in webrtcsdp.cc by kjellander · 8 years ago
  82. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  83. c9be3d5 Roll chromium_revision 3691cc167a..7aa4e8bf36 (483339:483375) by buildbot · 8 years ago
  84. 26afe21 Properly export the symbols of video frame-buffer classes for link-time by VladimirTechMan · 8 years ago
  85. 06b47c5 Listen for Wifi-Direct networks and include them in the network list by bdodson · 8 years ago
  86. 8cf398d Roll chromium_revision 53b56ec80b..3691cc167a (483312:483339) by buildbot · 8 years ago
  87. a9cc40b Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  88. 3dbfac3 Fix two simple type mismatches thay may cause compilation issues on win. by sprang · 8 years ago
  89. 960fd5b Adding a presubmit check for .proto files EOF newline by Mirko Bonadei · 8 years ago
  90. f1e3483 Revert "VideoFrameBuffer: Remove deprecated functions" by Magnus Jedvert · 8 years ago
  91. bc8feda Delete SignalThread class. by nisse · 8 years ago
  92. 428c9e2 VideoFrameBuffer: Remove deprecated functions by Magnus Jedvert · 8 years ago
  93. 57ca81a Actually use virtual network in OrtcFactory unit test. by deadbeef · 8 years ago
  94. 8a90f87 Add SetCodecSettings method for configuring VideoCodec settings. by asapersson · 8 years ago
  95. 17432ec Add magjed@ as owner of webrtc/api/video/ by Magnus Jedvert · 8 years ago
  96. 26b1f92 Roll chromium_revision 9dd69e9f64..53b56ec80b (483005:483312) by buildbot · 8 years ago
  97. 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 · 8 years ago
  98. e435019 Don't disable FEC if timing frames are disabled. by ilnik · 8 years ago
  99. 8c1ee7b Simplifies StartStopRecording test on iOS. by henrika · 8 years ago
  100. 8d08a92 Do not copy I420 frames in the decoder when not necessary. by Sami Kalliomäki · 8 years ago