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gerrit-public.fairphone.software / platform / external / webrtc / 9922016ee44c70eb1c14f647cc4a5ade58a0018c
  1. aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  2. 5f70d3b Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest. by sakal · 8 years ago
  3. 201dfe9 Split audio mixer into interface and implementation. by aleloi · 8 years ago
  4. 76648da Add FlexfecReceiveStream. by brandtr · 8 years ago
  5. 057b8d9 Remove all traces of Dr Memory. by Henrik Kjellander · 8 years ago
  6. 69034df Make GN build screenshare_loopback by palmkvist · 8 years ago
  7. 5a87245 iOS: Optimize video scaling and cropping by magjed · 8 years ago
  8. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  9. 1cb4823 Android YuvConverter: Use OpenGL Framebuffer instead of EGL pixel buffer by magjed · 8 years ago
  10. 9ab8a18 Android: Extend functionality of EglRenderer by magjed · 8 years ago
  11. ca20e7c Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ ) by nisse · 8 years ago
  12. c1f8ecb Remove check for numberOfCameras from AppRTC Mobile PeerConnectionClient. by sakal · 8 years ago
  13. be4aff7 Suppress deprecation warning in CallFragment. by sakal · 8 years ago
  14. aff9ff0 Create .git-blame-ignore-revs and add Java format CL to it. by sakal · 8 years ago
  15. e33c5d9 Added a level controller initialization value to MediaConstraints. by aleloi · 8 years ago
  16. 647915f Add loopback option and improve UX of AppRTCMobile for Mac. by denicija · 8 years ago
  17. 725e212 Prevent stripping of C interfaces in framework by kthelgason · 8 years ago
  18. e037060 Add to rtc::Optional equality/unequality comparisions with object by danilchap · 8 years ago
  19. a34e796 Delete unused file mediacommon.h. by nisse · 8 years ago
  20. 55928fe QualityScaler reset bugfix by kthelgason · 8 years ago
  21. 0489e49 Change RefCountedObject to use perfect forwarding. by perkj · 8 years ago
  22. 79f0bf3 A variable in ScreenCapturerWinDirectx has a bad name by zijiehe · 8 years ago
  23. f04f14e Revert of Move bitstream parser to more appropriate directory. (patchset #4 id:60001 of https://codereview.webrtc.org/2370853005/ ) by kthelgason · 8 years ago
  24. cc6817e Move current bitstream parser to more appropriate directory. by kthelgason · 8 years ago
  25. 577bc19 Android: Move YuvConverter to its own file by Magnus Jedvert · 8 years ago
  26. b6f1fb5 Delete RTPSender::BuildRtpHeader function and all dependencies by danilchap · 8 years ago
  27. 061ea0d Remove VideoCodec resolution validation. by Per · 8 years ago
  28. e3e411a Removed perkj@ from video WATCHLIST by Per · 8 years ago
  29. 73c5d4a Include ScreenCapturerAndroid in libjingle_peerconnection_java.jar by sakal · 8 years ago
  30. 0934785 Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ ) by nisse · 8 years ago
  31. 4e52386 Reland of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #1 id:1 of https://codereview.webrtc.org/2427733002/ ) by brandtr · 8 years ago
  32. 249beee Remove DesktopRegion parameter from DesktopCapturer::Capture by zijiehe · 8 years ago
  33. e8295fc Roll chromium_revision eb9b71b64b..4c4977aa05 (426008:426117) by buildbot · 8 years ago
  34. 6a4607e Deflaky ScreenCapturerTest by zijiehe · 8 years ago
  35. 1eb1293 Handle BW drop in ALR region and initiate probing by Irfan Sheriff · 8 years ago
  36. a9c7cfa Prepare for introduction of rtc::PacketTransportInterface. by johan · 8 years ago
  37. 1203066 Compilerwarning possible loss of data in file port.h by bertholdherrmann08 · 8 years ago
  38. cc555c5 RTCDataChannelStats[1] added, supporting all stats members. by hbos · 8 years ago
  39. 1394c7b Fix for flaky test: EndToEndTest.VerifyHistogramStatsWithRtx by asapersson · 8 years ago
  40. 0b7be9c Roll chromium_revision c8b7ee41e0..eb9b71b64b (425645:426008) by buildbot · 8 years ago
  41. 9960bb1 Call OnTransportFeedback just when feedback_observer exist. by michaelt · 8 years ago
  42. 53fe19d Set min and max rate on caller and on callee side. by michaelt · 8 years ago
  43. 64e1a32 Second try to get "Support for video file instead of camera and output video out to file" accepted by mandermo · 8 years ago
  44. 67a8c98 Revert of Support for video file instead of camera and output video out to file (patchset #17 id:320001 of https://codereview.webrtc.org/2273573003/ ) by kjellander · 8 years ago
  45. f33970b Add unittest for I420Buffer::Rotate. by nisse · 8 years ago
  46. 6ed592d Rename variables to reflect that DelayBasedBwe lives on the send side rather than receive side. by terelius · 8 years ago
  47. 5588a13 Now uses rtc::Buffer in AudioDeviceBuffer. by henrika · 8 years ago
  48. 4466699 Support for video file instead of camera and output video out to file by mandermo · 8 years ago
  49. 9e83c97 Add rtc::Optional::emplace by danilchap · 8 years ago
  50. 7a37761 Removed RTPHeader from NetEq's Packet struct. by ossu · 8 years ago
  51. 553024a During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger. by aleloi · 8 years ago
  52. e405d9b Add a fuzzer for FlexfecReceiver. by brandtr · 8 years ago
  53. e6b5829 Extends how AppRTCMobile handles audio focus on Android by henrika · 8 years ago
  54. 91718a1 Roll script: Update after SVN support was dropped from depot tools by kjellander · 8 years ago
  55. 5c63989 Import build/config/clang/clang.gni in webrtc/base/BUILD.gn by ehmaldonado · 8 years ago
  56. 862d74d Revert of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #2 id:60001 of https://codereview.webrtc.org/2390823009/ ) by honghaiz · 8 years ago
  57. c4fd23c Add rtc::Optional::reset by danilchap · 8 years ago
  58. 9b9910d Roll chromium_revision 3d5a0fb164..c8b7ee41e0 (425604:425645) by buildbot · 8 years ago
  59. 7e76560 Enable logging to console in DirectRTCClientTest. by sakal · 8 years ago
  60. 2f255d8 Replace const -> constexpr for rtcp Packet Type by danilchap · 8 years ago
  61. c1f40b7 Remove RtcpPacket dependency on rtcp_utility by danilchap · 8 years ago
  62. 27c3d5b Restore thread name consistency for webrtc/p2p/ . by johan · 8 years ago
  63. 883ad66 Removed the deprecated audioproc executable by peah · 8 years ago
  64. e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  65. 671534e Roll chromium_revision 90b4cf429c..3d5a0fb164 (425384:425604) by buildbot · 8 years ago
  66. 9c4b4b4 Add path for recovered packets from internal::Call to RtpStreamReceiver. by brandtr · 8 years ago
  67. e5ddf52 Delete unused file webrtcvideochannelfactory.h. by nisse · 8 years ago
  68. 285e558 Removed suppressions for the data race inside the APM that is now fixed. by peah · 8 years ago
  69. 2c4c422 Roll chromium_revision a3c4a78675..90b4cf429c (425286:425384) by buildbot · 8 years ago
  70. 0d4b129 Roll chromium_revision 00384b2217..a3c4a78675 (425234:425286) by buildbot · 8 years ago
  71. 8f7cc7e This CL corrects the emptying of the render queues for the by peah · 8 years ago
  72. 5d2e58c Roll chromium_revision 61fb879aaf..00384b2217 (425083:425234) by buildbot · 8 years ago
  73. 91902cb Remove DesktopRegion parameter from DesktopCapturer::Capture. by zijiehe · 8 years ago
  74. 794d535 Roll chromium_revision 50c7b3ce18..61fb879aaf (424992:425083) by buildbot · 8 years ago
  75. 9ae585d Cleanup of voice_engine includes. by aleloi · 8 years ago
  76. 3283cf9 Add asyncstuntcpsocket_unittest.cc to rtc_unittests by kjellander · 8 years ago
  77. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  78. b593bc0 Suggest myself as owner of api/ by solenberg · 8 years ago
  79. 81b8a07 Roll chromium_revision 2cabef4e7d..50c7b3ce18 (424936:424992) by buildbot · 8 years ago
  80. 0d83857 NetEq: Convert AverageIAT from int to float calculations by henrik.lundin · 8 years ago
  81. c9ec875 NetEq: Remove special case for Merge without Expand by henrik.lundin · 8 years ago
  82. 722b0dc Revert of Android audio playout now supports non-call media streams (patchset #3 id:10004 of https://codereview.webrtc.org/2411263003/ ) by henrika · 8 years ago
  83. dd7a1cf Landmine due to corrupt .pdb files on Windows. by Henrik Kjellander · 8 years ago
  84. da3303f Revert of Remove tools dir from root webrtc target (patchset #1 id:1 of https://codereview.webrtc.org/2412353004/ ) by kjellander · 8 years ago
  85. 91a5759 Roll chromium_revision 316b880c55..2cabef4e7d (421519:424936) by Henrik Kjellander · 8 years ago
  86. 163b1a2 Remove tools dir from root webrtc target by charujain · 8 years ago
  87. 614f68f Remove duplicate entry in webrtc .gn file exec_script_whitelist by fbarchard · 8 years ago
  88. db158f9 Fix experiment name in BitrateControllerTest. by Stefan Holmer · 8 years ago
  89. 77c663d Give FeedbackTimeout experiment the correct name. by stefan · 8 years ago
  90. 12a39f4 Don't crash on unexpected stap-a or fu-a. by stefan · 8 years ago
  91. 75c8fb4 DataChannelInterface default impl of [messages/bytes]_[sent/received]. by hbos · 8 years ago
  92. 84ffdee DataChannel[Interface]::[message/bytes]_[sent/received]() added. by hbos · 8 years ago
  93. 73fdc31 Several fixes to screen_capturer_mac. by erikchen · 8 years ago
  94. e280cde Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz. by ossu · 8 years ago
  95. 872f614 Android audio playout now supports non-call media streams. by henrika · 8 years ago
  96. 116ec6d Implemented further mixer interface change suggestions from https://codereview.webrtc.org/2386383003/ by aleloi · 8 years ago
  97. 7e30432 Hooking up audio network adaptor to VoE. by minyue · 8 years ago
  98. 917d4e1 Removed the legacy behavior of stopping playout when setting new receive codecs. by solenberg · 8 years ago
  99. e97974d Cleanup of the mixer interface. by aleloi · 8 years ago
  100. 73a28ee The AudioProcessing class is used as an interface by peah · 8 years ago