1. cc9669c Cleanup shared memory handling in DesktopCapturer interface. by sergeyu · 8 years ago
  2. c815c60 Roll chromium_revision 3a90ecf..a6aefb7 (374096:374428) by kjellander · 8 years ago
  3. fa639f0 Surface the noise estimate of the NS to be used by other components by Alejandro Luebs · 8 years ago
  4. 78ddd73 Update path for audioproc_debug proto output. by kjellander · 8 years ago
  5. 4bba35f Switch third_party/gflags to use updated GitHub repo. by kjellander · 8 years ago
  6. 09fef9e [rtp_rtcp] Added Sender Report Request rtcp packet. by danilchap · 8 years ago
  7. dfb769d Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete by perkj · 8 years ago
  8. 0715a83 Avoid OpenH264 encoder bug for #threads > 1 on Mac and Chromium+Sandbox. by hbos · 8 years ago
  9. 097d549 Added thread annotations to FifoBuffer. by jbauch · 8 years ago
  10. e594213 Fix div-by-0 in NetEq's StatisticsCalculator by henrik.lundin · 8 years ago
  11. fd2be27 Fuzzer tests for AudioDecoder's DecodeRedundant and IncomingPacket by henrik.lundin · 8 years ago
  12. 7ae5e52 Revert of Analyze support in gyp_webrtc (patchset #1 id:1 of https://codereview.webrtc.org/1369683004/ ) by kjellander · 8 years ago
  13. d2a2296 Enable cpplint for webrtc/modules/pacing and fix all uncovered cpplint errors. by jbauch · 8 years ago
  14. cd0e475 Create QuicSession by mikescarlett · 8 years ago
  15. 456801d Add perkj+magjed to webrtc/media/OWNERS by kjellander · 8 years ago
  16. c0ae305 Fix null-pointer dereference in RTPSenderVideo. by Peter Boström · 8 years ago
  17. 58c664c Clean up of CongestionController. by Stefan Holmer · 8 years ago
  18. d1d66ba Remove ViEChannel calls for VideoReceiveStream. by Peter Boström · 8 years ago
  19. 2945153 Roll chromium_revision 8da2495..3a90ecf (374076:374096) by kjellander · 8 years ago
  20. 7336eeb [rtp_rtcp] rtcp::Tmmbn cleaned and got Parse function by danilchap · 8 years ago
  21. 62756ee Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. by hbos · 8 years ago
  22. 47b6263 Remove Java PC support. This cl removes none Android Java support. by perkj · 8 years ago
  23. f6b5509 Fix GYP and GN references that are invalid in Chromium builds. by kjellander · 8 years ago
  24. 1afca73 Change to WebRTC license in webrtc/media by kjellander · 8 years ago
  25. 66a1401 Roll chromium_revision 3a7cbe0..8da2495 (374049:374076) by kjellander · 8 years ago
  26. a81f6a3 Revert of Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. (patchset #1 id:1 of https://codereview.webrtc.org/1674103002/ ) by hbos · 8 years ago
  27. 10b9dd7 Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. by hbos · 8 years ago
  28. c37b59f Roll chromium_revision 9127267..3a7cbe0 (374043:374049) by kjellander · 8 years ago
  29. f9f84b2 Roll chromium_revision 70700a1..9127267 (374041:374043) by kjellander · 8 years ago
  30. 39be561 Roll chromium_revision f0cfd18..70700a1 (374026:374041) by kjellander · 8 years ago
  31. cdc4451 Roll chromium_revision 3c45587..f0cfd18 (373863:374026) by kjellander · 8 years ago
  32. e796f96 check v4l frame rate capability with bitwise method. by Weiyong Yao · 8 years ago
  33. fd6706a Log Android HW decoder delay time statistics. by glaznev · 8 years ago
  34. 1c24a6d Set use_gtk=0 as default for Chromium builds. by kjellander · 8 years ago
  35. 210cf01 Roll chromium_revision 6e376b8..3c45587 (373731:373863) by kjellander · 8 years ago
  36. c09525a Revert of Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. (patchset #1 id:1 of https://codereview.webrtc.org/1660403004/ ) by hbos · 8 years ago
  37. 50fca62 Remove fake cricket::VideoCapturer devices. by Peter Boström · 8 years ago
  38. 7cd94f6 Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. by hbos · 8 years ago
  39. 900f975 H264: Improve FFmpeg decoder performance by using I420BufferPool. by hbos · 8 years ago
  40. c6e16e3 Use a delayed encoder in GetStats test. by Peter Boström · 8 years ago
  41. f751bf8 Set VideoReceiveStream members in init list. by Peter Boström · 8 years ago
  42. f174e3a [rtp_rtcp] rtcp::Tmmbr cleaned and got Parse function by danilchap · 8 years ago
  43. 48fa271 Made implicit casts in the echo canceller explicit. by peah · 8 years ago
  44. 1d04ac6 Untangle ViEChannel and ViEEncoder. by Peter Boström · 8 years ago
  45. e449915 Measure encoding time on encode callbacks. by Peter Boström · 8 years ago
  46. 8e8908a Delete FrameInput method and FrameInputWrapper class. by nisse · 8 years ago
  47. 25d1f28 Fix race between Thread ctor/dtor and MessageQueueManager registrations. by jbauch · 8 years ago
  48. 988d31e Move gtest_prod_util.h out of webrtc/test tree. by kjellander · 8 years ago
  49. a96e2d7 Move talk/media to webrtc/media by kjellander · 8 years ago
  50. a713a40 Roll chromium_revision 4c670a4..6e376b8 (373575:373731) by kjellander · 8 years ago
  51. b647aca Roll chromium_revision fbab684..4c670a4 (373504:373575) by kjellander · 8 years ago
  52. ae95ff3 Add more logging and fix PTS overflow for HW decoder. by glaznev · 8 years ago
  53. a92d6be rtcp::TmmbItem designed to replace RTCPUtility::RTCPPacketRTPFBTMMBRItem (for creating and parsing rtcp TMMBR/TMMBN packets) by danilchap · 8 years ago
  54. 20834ca Adds a nullptr check to prevent a rare crash when starting or stopping an RtcEventLog. by ivoc · 8 years ago
  55. 15ba624 Revert of Rename iOS test specs to match buildbot names. (patchset #1 id:1 of https://codereview.webrtc.org/1665783002/ ) by kjellander@webrtc.org · 8 years ago
  56. daa672d Roll chromium_revision 28e68f8..fbab684 (373442:373504) by kjellander · 8 years ago
  57. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 8 years ago
  58. 2ddb8bd Avoid undefined behavior in vp8 screenshare_layers by sprang · 8 years ago
  59. 08582ff Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. by nisse · 8 years ago
  60. 8cb910d Delete backwards compatibility cruft from cricket::VideoFrame and VideoSourceInterface. by nisse · 8 years ago
  61. c2148a5 Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics). by asapersson · 8 years ago
  62. 9031d63 Remove the network with empty name or NONE connection type from the network list. by honghaiz · 8 years ago
  63. fc5fc1e Roll chromium_revision 609aa24..28e68f8 (373145:373442) by kjellander · 8 years ago
  64. f2a2bf4 Stay writable after partial socket writes. by jbauch · 8 years ago
  65. 14d024d Do not notify networkconnect if the connection type is known. by Honghai Zhang · 8 years ago
  66. 45b683f Call static method getConnectionType using the class name. by Honghai Zhang · 8 years ago
  67. 5c35cf9 Re-enable RestartingSendStreamPreservesRtpState. by danilchap · 8 years ago
  68. cedff02 Remove dead code from WebRtcVideoEngine2. by Peter Boström · 8 years ago
  69. e03ac51 Implement NullVideoDecoder to avoid crash on unsupported decoders. by jbauch · 8 years ago
  70. 9dc5928 Ability to disable the effects of |rtc_use_h264| with DisableRtcUseH264. by hbos · 8 years ago
  71. 86512b4 Rename iOS test specs to match buildbot names. by kjellander · 8 years ago
  72. 1088001 Support multiple rtx codecs. by Stefan Holmer · 8 years ago
  73. abe095b Roll chromium_revision c6076f2..609aa24 (372974:373145) by kjellander · 8 years ago
  74. d983c3c Add SEI to the list of packets counted as keyframe. by noahric · 8 years ago
  75. d1fb26d Add iOS tracing. by tkchin · 8 years ago
  76. 8e85a3f iOS buildbot configurations. by kjellander · 8 years ago
  77. 7f77749 Disable flaky test WebRtcSessionTest.TestRtxRemovedByCreateAnswer on Win and Mac. by honghaiz · 8 years ago
  78. 27a3485 Fixing a DCHECK failure on unknown connection type from OS. by honghaiz · 8 years ago
  79. 2ab8157 Remove implicit downcast in producer_fec_fuzzer.cc. by Peter Boström · 8 years ago
  80. c3a0983 Roll chromium_revision a8e5140..c6076f2 (372922:372974) incl. update to Opus v.1.1.2 by kjellander · 8 years ago
  81. a7ad7c3 Get the adapter type information from Android OS. by honghaiz · 8 years ago
  82. ae695e9 Refactor RtpSender and SSRCDatabase. by tommi · 8 years ago
  83. 040b79f Add helper macros for calling a histogram with different names. by asapersson · 8 years ago
  84. ed3277b Deprecate VideoDecoder::Reset() and remove calls. by Peter Boström · 8 years ago
  85. ce23bee Remove SendStreamFormat and ViewRequests. by Peter Boström · 8 years ago
  86. d467a91 Roll chromium_revision f41a54b..a8e5140 (372876:372922) by kjellander · 8 years ago
  87. c61635c PRESUBMIT: Exclude supplement.gypi from _CheckNoSourcesAboveGyp check. by kjellander · 8 years ago
  88. c5a39c2 H264: Thread-safe InitializeFFmpeg. Flag to control if InitializeFFmpeg should be called. by hbos · 8 years ago
  89. 799379e Let a minimum time interval pass (one bucket size) after initialization before reporting rates (to avoid rates being based on too short time intervals). by asapersson · 8 years ago
  90. 6fd26b6 Roll chromium_revision f6e3d46..f41a54b (372710:372876) by kjellander · 8 years ago
  91. c463e20 Reset TURN port NONCE when a new socket is created. by honghaiz · 8 years ago
  92. 9429148 Extra logging for HW codec. by glaznev · 8 years ago
  93. 3668cf0 Roll chromium_revision 65f9b34..f6e3d46 (372637:372710) by kjellander · 8 years ago
  94. a6c39d9 Remove unimplemented VideoChannel code. by Peter Boström · 8 years ago
  95. 6f7557e Disable useless BWE tests. by stefan · 8 years ago
  96. e37a2d1 Reland "Removing webrtc::AudioFrame::energy_." by minyue · 8 years ago
  97. d8de115 Remove mutable from rtc::CriticalSections. by pbos · 8 years ago
  98. 34877ee Revert of Added validation between RTP and RTCP timestamps (patchset #7 id:120001 of https://codereview.webrtc.org/1633843003/ ) by danilchap · 8 years ago
  99. 74451a5 Prevent zero division in VCMJitterEstimator. by Peter Boström · 8 years ago
  100. b46c333 Roll chromium_revision 126e210..65f9b34 (372588:372637) by kjellander · 8 years ago