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gerrit-public.fairphone.software / platform / external / webrtc / da38293e51ba2a058ca75880d6f719d6e8b0a93b
  1. da38293 Added the missing ReadQueuedRenderData() call to the AEC bitexactness test by peah · 8 years ago
  2. 894c400 Android VideoFileRenderer: Wait for posted frames in release() by Magnus Jedvert · 8 years ago
  3. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  4. 3355f6d Avoids invalid copy of audio buffer to task queue. by henrika · 8 years ago
  5. c4d2dc4 Delete DataLog abstraction, which was almost unused. by nisse · 8 years ago
  6. dda1e60 Reland of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2441493003/ ) by nisse · 8 years ago
  7. bd16341 Roll chromium_revision 4c4977aa05..f9e01d4887 (426117:426685) by buildbot · 8 years ago
  8. 84fbf9e SUCCEEDED macro is misused by zijiehe · 8 years ago
  9. bdb8df8 BringSelectedWindowToFront should bring the window to front instead of only focusing it by zijiehe · 8 years ago
  10. 97abf24 Use variadic templates instead of pump for RefCountedObject by danilchap · 8 years ago
  11. 6c27849 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  12. 920d30b Replaced thread checker with race checker in AudioMixer. by aleloi · 8 years ago
  13. 161a586 Fix some flaky tests by using longer timeout and/or fake clock. by Honghai Zhang · 8 years ago
  14. b9eaeba Return nullptr from RTCCertificate::FromPEM on failure. by jbroman · 8 years ago
  15. 58000a0 Move shared_desktop_frame.cc to webrtc/modules/desktop_capture:primitives by Sergey Ulanov · 8 years ago
  16. 142f019 Append second nack list in same compound rtcp packet instead of replace by danilchap · 8 years ago
  17. aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  18. 5f70d3b Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest. by sakal · 8 years ago
  19. 201dfe9 Split audio mixer into interface and implementation. by aleloi · 8 years ago
  20. 76648da Add FlexfecReceiveStream. by brandtr · 8 years ago
  21. 057b8d9 Remove all traces of Dr Memory. by Henrik Kjellander · 8 years ago
  22. 69034df Make GN build screenshare_loopback by palmkvist · 8 years ago
  23. 5a87245 iOS: Optimize video scaling and cropping by magjed · 8 years ago
  24. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  25. 1cb4823 Android YuvConverter: Use OpenGL Framebuffer instead of EGL pixel buffer by magjed · 8 years ago
  26. 9ab8a18 Android: Extend functionality of EglRenderer by magjed · 8 years ago
  27. ca20e7c Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ ) by nisse · 8 years ago
  28. c1f8ecb Remove check for numberOfCameras from AppRTC Mobile PeerConnectionClient. by sakal · 8 years ago
  29. be4aff7 Suppress deprecation warning in CallFragment. by sakal · 8 years ago
  30. aff9ff0 Create .git-blame-ignore-revs and add Java format CL to it. by sakal · 8 years ago
  31. e33c5d9 Added a level controller initialization value to MediaConstraints. by aleloi · 8 years ago
  32. 647915f Add loopback option and improve UX of AppRTCMobile for Mac. by denicija · 8 years ago
  33. 725e212 Prevent stripping of C interfaces in framework by kthelgason · 8 years ago
  34. e037060 Add to rtc::Optional equality/unequality comparisions with object by danilchap · 8 years ago
  35. a34e796 Delete unused file mediacommon.h. by nisse · 8 years ago
  36. 55928fe QualityScaler reset bugfix by kthelgason · 8 years ago
  37. 0489e49 Change RefCountedObject to use perfect forwarding. by perkj · 8 years ago
  38. 79f0bf3 A variable in ScreenCapturerWinDirectx has a bad name by zijiehe · 8 years ago
  39. 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
  40. cc6817e Move current bitstream parser to more appropriate directory. by kthelgason · 8 years ago
  41. 577bc19 Android: Move YuvConverter to its own file by Magnus Jedvert · 8 years ago
  42. b6f1fb5 Delete RTPSender::BuildRtpHeader function and all dependencies by danilchap · 8 years ago
  43. 061ea0d Remove VideoCodec resolution validation. by Per · 8 years ago
  44. e3e411a Removed perkj@ from video WATCHLIST by Per · 8 years ago
  45. 73c5d4a Include ScreenCapturerAndroid in libjingle_peerconnection_java.jar by sakal · 8 years ago
  46. 0934785 Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ ) by nisse · 8 years ago
  47. 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
  48. 249beee Remove DesktopRegion parameter from DesktopCapturer::Capture by zijiehe · 8 years ago
  49. e8295fc Roll chromium_revision eb9b71b64b..4c4977aa05 (426008:426117) by buildbot · 8 years ago
  50. 6a4607e Deflaky ScreenCapturerTest by zijiehe · 8 years ago
  51. 1eb1293 Handle BW drop in ALR region and initiate probing by Irfan Sheriff · 8 years ago
  52. a9c7cfa Prepare for introduction of rtc::PacketTransportInterface. by johan · 8 years ago
  53. 1203066 Compilerwarning possible loss of data in file port.h by bertholdherrmann08 · 8 years ago
  54. cc555c5 RTCDataChannelStats[1] added, supporting all stats members. by hbos · 8 years ago
  55. 1394c7b Fix for flaky test: EndToEndTest.VerifyHistogramStatsWithRtx by asapersson · 8 years ago
  56. 0b7be9c Roll chromium_revision c8b7ee41e0..eb9b71b64b (425645:426008) by buildbot · 8 years ago
  57. 9960bb1 Call OnTransportFeedback just when feedback_observer exist. by michaelt · 8 years ago
  58. 53fe19d Set min and max rate on caller and on callee side. by michaelt · 8 years ago
  59. 64e1a32 Second try to get "Support for video file instead of camera and output video out to file" accepted by mandermo · 8 years ago
  60. 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
  61. f33970b Add unittest for I420Buffer::Rotate. by nisse · 8 years ago
  62. 6ed592d Rename variables to reflect that DelayBasedBwe lives on the send side rather than receive side. by terelius · 8 years ago
  63. 5588a13 Now uses rtc::Buffer in AudioDeviceBuffer. by henrika · 8 years ago
  64. 4466699 Support for video file instead of camera and output video out to file by mandermo · 8 years ago
  65. 9e83c97 Add rtc::Optional::emplace by danilchap · 8 years ago
  66. 7a37761 Removed RTPHeader from NetEq's Packet struct. by ossu · 8 years ago
  67. 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
  68. e405d9b Add a fuzzer for FlexfecReceiver. by brandtr · 8 years ago
  69. e6b5829 Extends how AppRTCMobile handles audio focus on Android by henrika · 8 years ago
  70. 91718a1 Roll script: Update after SVN support was dropped from depot tools by kjellander · 8 years ago
  71. 5c63989 Import build/config/clang/clang.gni in webrtc/base/BUILD.gn by ehmaldonado · 8 years ago
  72. 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
  73. c4fd23c Add rtc::Optional::reset by danilchap · 8 years ago
  74. 9b9910d Roll chromium_revision 3d5a0fb164..c8b7ee41e0 (425604:425645) by buildbot · 8 years ago
  75. 7e76560 Enable logging to console in DirectRTCClientTest. by sakal · 8 years ago
  76. 2f255d8 Replace const -> constexpr for rtcp Packet Type by danilchap · 8 years ago
  77. c1f40b7 Remove RtcpPacket dependency on rtcp_utility by danilchap · 8 years ago
  78. 27c3d5b Restore thread name consistency for webrtc/p2p/ . by johan · 8 years ago
  79. 883ad66 Removed the deprecated audioproc executable by peah · 8 years ago
  80. e40a7ee GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  81. 671534e Roll chromium_revision 90b4cf429c..3d5a0fb164 (425384:425604) by buildbot · 8 years ago
  82. 9c4b4b4 Add path for recovered packets from internal::Call to RtpStreamReceiver. by brandtr · 8 years ago
  83. e5ddf52 Delete unused file webrtcvideochannelfactory.h. by nisse · 8 years ago
  84. 285e558 Removed suppressions for the data race inside the APM that is now fixed. by peah · 8 years ago
  85. 2c4c422 Roll chromium_revision a3c4a78675..90b4cf429c (425286:425384) by buildbot · 8 years ago
  86. 0d4b129 Roll chromium_revision 00384b2217..a3c4a78675 (425234:425286) by buildbot · 8 years ago
  87. 8f7cc7e This CL corrects the emptying of the render queues for the by peah · 8 years ago
  88. 5d2e58c Roll chromium_revision 61fb879aaf..00384b2217 (425083:425234) by buildbot · 8 years ago
  89. 91902cb Remove DesktopRegion parameter from DesktopCapturer::Capture. by zijiehe · 8 years ago
  90. 794d535 Roll chromium_revision 50c7b3ce18..61fb879aaf (424992:425083) by buildbot · 8 years ago
  91. 9ae585d Cleanup of voice_engine includes. by aleloi · 8 years ago
  92. 3283cf9 Add asyncstuntcpsocket_unittest.cc to rtc_unittests by kjellander · 8 years ago
  93. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  94. b593bc0 Suggest myself as owner of api/ by solenberg · 8 years ago
  95. 81b8a07 Roll chromium_revision 2cabef4e7d..50c7b3ce18 (424936:424992) by buildbot · 8 years ago
  96. 0d83857 NetEq: Convert AverageIAT from int to float calculations by henrik.lundin · 8 years ago
  97. c9ec875 NetEq: Remove special case for Merge without Expand by henrik.lundin · 8 years ago
  98. 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
  99. dd7a1cf Landmine due to corrupt .pdb files on Windows. by Henrik Kjellander · 8 years ago
  100. 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