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gerrit-public.fairphone.software / platform / external / webrtc / 1836fd6257a692959b3b49ba99ef587ad9995871
  1. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  2. 701d628 Moved the AGC render sample queue into the audio processing module by peah · 8 years ago
  3. d8872c5 Removed the file resources/audioproc.aecdump.sha1 file which is no longer used. by peah · 8 years ago
  4. a062460 Several subcomponents inside APM copy render audio from by peah · 8 years ago
  5. 67c8bc4 RTCStats equality operator added. by hbos · 8 years ago
  6. 01bbc3c Reland of Deflake ChangingNetworkRoute test. by stefan · 8 years ago
  7. 77f5953 Revert of Deflake ChangingNetworkRoute test. (patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ ) by ehmaldonado · 8 years ago
  8. 6711820 Deflake ChangingNetworkRoute test. by stefan · 8 years ago
  9. cc34833 Remove now unused code in RtpHeaderExtensionMap by danilchap · 8 years ago
  10. 611f267 Make WebRTC compatible with OpenH264 v1.6. by sprang · 8 years ago
  11. af1ae31 Remove dead dependencies on xmllite and xmpp. by kjellander · 8 years ago
  12. dbf6705 codec_unittest.cc: Fix TEST vs TEST_F mismatch by Magnus Jedvert · 8 years ago
  13. 06c8e1e Revert of H264 codec: Check profile-level-id when matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ ) by Magnus Jedvert · 8 years ago
  14. fcba8fe Delete left-over file profiler_unittest.cc. by nisse · 8 years ago
  15. 74097fd Delete unused file screencastid.h. by nisse · 8 years ago
  16. e58d73d Fix more swarming test failures by using the fake clock or longer timeout. by honghaiz · 8 years ago
  17. a6b8298 Use relative names in GN to make Chromium happy by kwiberg · 8 years ago
  18. 4a18f16 Update XServerPixelBuffer to handle errors returned from XGetImage(). by sergeyu · 8 years ago
  19. da2bf4e Stop using old AudioCodingModule::RegisterReceiveCodec overloads by kwiberg · 8 years ago
  20. 88b7074 Remove unused function implementations from FakeWebRtcVoiceEngine. by solenberg · 8 years ago
  21. fb70b45 Preventing TURN redirects to loopback addresses. by deadbeef · 8 years ago
  22. 838cdb3 Revert of Fix chromium-style warnings. (patchset #1 id:1 of https://codereview.webrtc.org/2400993002/ ) by terelius · 8 years ago
  23. 5d79a7c rtcstats_objects.h updated with TODOs about stats not being collected by hbos · 8 years ago
  24. a6f495c Simplifying audio network adaptor by moving receiver frame length range to ctor. by minyue · 8 years ago
  25. a73f6c9 NetEq now works with packets as values, rather than pointers. by ossu · 8 years ago
  26. d312713 Roll chromium_revision 1362287708..9b5bb47fa0 (426760:426837) + roll Android SDK to N by ehmaldonado · 8 years ago
  27. 86b92e0 Drop VP8 frames older than the last sync frame in the RtpFrameReferenceFinder. by philipel · 8 years ago
  28. 1655e45 Elimiteted race condition in the AudioMixer. by aleloi · 8 years ago
  29. 2206c95 Revert of Fix some chromium style warnings in remote_bitrate_estimator.h (patchset #1 id:1 of https://codereview.webrtc.org/2387113008/ ) by terelius · 8 years ago
  30. 0140408 Add tests and fix thread annotations by danilchap · 8 years ago
  31. b60d196 Eliminate left shift of negative value by using multiplication instead by kwiberg · 8 years ago
  32. 2fa7c67 RTCTransportStats[1] added, supporting all members. by hbos · 8 years ago
  33. 5de3a7e Remove unused variable from delay based BWE. by terelius · 8 years ago
  34. 509eadd Fix chromium-style warnings. by terelius · 8 years ago
  35. c22bcf4 Fix some chromium style warnings in remote_bitrate_estimator.h by terelius · 8 years ago
  36. d7ce668 Roll chromium_revision f9e01d4887..1362287708 (426685:426760) by buildbot · 8 years ago
  37. a3cac05 GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn by kjellander · 8 years ago
  38. 43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 8 years ago
  39. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  40. d3c4008 Delete always-zero ByteBufferWriter::start_. by nisse · 8 years ago
  41. 61c053e Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ ) by nisse · 8 years ago
  42. ebf5240 Allow using Java classes that don't require JNI in Chromium. by sakal · 8 years ago
  43. 66712b0 Revert of Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. (patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ ) by nisse · 8 years ago
  44. 151572b Delete unused class AudioSourceWithMixStatus. by nisse · 8 years ago
  45. 25445d3 Integrate FlexfecReceiveStream with Call. by brandtr · 8 years ago
  46. 764e364 Several subcomponents inside APM copy render audio from by peah · 8 years ago
  47. 8974973 Added the missing ReadQueuedRenderData() call to the AECM bitexactness test by peah · 8 years ago
  48. 12986c4 Added the missing ReadQueuedRenderData() call to the gain controller bitexactness test by peah · 8 years ago
  49. da38293 Added the missing ReadQueuedRenderData() call to the AEC bitexactness test by peah · 8 years ago
  50. 894c400 Android VideoFileRenderer: Wait for posted frames in release() by Magnus Jedvert · 8 years ago
  51. 8c63a82 Add a placeholder stat for logging the estimated residual echo likelihood. by ivoc · 8 years ago
  52. 3355f6d Avoids invalid copy of audio buffer to task queue. by henrika · 8 years ago
  53. c4d2dc4 Delete DataLog abstraction, which was almost unused. by nisse · 8 years ago
  54. dda1e60 Reland of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2441493003/ ) by nisse · 8 years ago
  55. bd16341 Roll chromium_revision 4c4977aa05..f9e01d4887 (426117:426685) by buildbot · 8 years ago
  56. 84fbf9e SUCCEEDED macro is misused by zijiehe · 8 years ago
  57. bdb8df8 BringSelectedWindowToFront should bring the window to front instead of only focusing it by zijiehe · 8 years ago
  58. 97abf24 Use variadic templates instead of pump for RefCountedObject by danilchap · 8 years ago
  59. 6c27849 Move audio frame memory handling inside AudioMixer. by aleloi · 8 years ago
  60. 920d30b Replaced thread checker with race checker in AudioMixer. by aleloi · 8 years ago
  61. 161a586 Fix some flaky tests by using longer timeout and/or fake clock. by Honghai Zhang · 8 years ago
  62. b9eaeba Return nullptr from RTCCertificate::FromPEM on failure. by jbroman · 8 years ago
  63. 58000a0 Move shared_desktop_frame.cc to webrtc/modules/desktop_capture:primitives by Sergey Ulanov · 8 years ago
  64. 142f019 Append second nack list in same compound rtcp packet instead of replace by danilchap · 8 years ago
  65. aed581a Made AudioReceiveStream a mixer participant. by aleloi · 8 years ago
  66. 5f70d3b Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest. by sakal · 8 years ago
  67. 201dfe9 Split audio mixer into interface and implementation. by aleloi · 8 years ago
  68. 76648da Add FlexfecReceiveStream. by brandtr · 8 years ago
  69. 057b8d9 Remove all traces of Dr Memory. by Henrik Kjellander · 8 years ago
  70. 69034df Make GN build screenshare_loopback by palmkvist · 8 years ago
  71. 5a87245 iOS: Optimize video scaling and cropping by magjed · 8 years ago
  72. 7a97344 Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream. by minyue · 8 years ago
  73. 1cb4823 Android YuvConverter: Use OpenGL Framebuffer instead of EGL pixel buffer by magjed · 8 years ago
  74. 9ab8a18 Android: Extend functionality of EglRenderer by magjed · 8 years ago
  75. ca20e7c Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ ) by nisse · 8 years ago
  76. c1f8ecb Remove check for numberOfCameras from AppRTC Mobile PeerConnectionClient. by sakal · 8 years ago
  77. be4aff7 Suppress deprecation warning in CallFragment. by sakal · 8 years ago
  78. aff9ff0 Create .git-blame-ignore-revs and add Java format CL to it. by sakal · 8 years ago
  79. e33c5d9 Added a level controller initialization value to MediaConstraints. by aleloi · 8 years ago
  80. 647915f Add loopback option and improve UX of AppRTCMobile for Mac. by denicija · 8 years ago
  81. 725e212 Prevent stripping of C interfaces in framework by kthelgason · 8 years ago
  82. e037060 Add to rtc::Optional equality/unequality comparisions with object by danilchap · 8 years ago
  83. a34e796 Delete unused file mediacommon.h. by nisse · 8 years ago
  84. 55928fe QualityScaler reset bugfix by kthelgason · 8 years ago
  85. 0489e49 Change RefCountedObject to use perfect forwarding. by perkj · 8 years ago
  86. 79f0bf3 A variable in ScreenCapturerWinDirectx has a bad name by zijiehe · 8 years ago
  87. 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
  88. cc6817e Move current bitstream parser to more appropriate directory. by kthelgason · 8 years ago
  89. 577bc19 Android: Move YuvConverter to its own file by Magnus Jedvert · 8 years ago
  90. b6f1fb5 Delete RTPSender::BuildRtpHeader function and all dependencies by danilchap · 8 years ago
  91. 061ea0d Remove VideoCodec resolution validation. by Per · 8 years ago
  92. e3e411a Removed perkj@ from video WATCHLIST by Per · 8 years ago
  93. 73c5d4a Include ScreenCapturerAndroid in libjingle_peerconnection_java.jar by sakal · 8 years ago
  94. 0934785 Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ ) by nisse · 8 years ago
  95. 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
  96. 249beee Remove DesktopRegion parameter from DesktopCapturer::Capture by zijiehe · 8 years ago
  97. e8295fc Roll chromium_revision eb9b71b64b..4c4977aa05 (426008:426117) by buildbot · 8 years ago
  98. 6a4607e Deflaky ScreenCapturerTest by zijiehe · 8 years ago
  99. 1eb1293 Handle BW drop in ALR region and initiate probing by Irfan Sheriff · 8 years ago
  100. a9c7cfa Prepare for introduction of rtc::PacketTransportInterface. by johan · 8 years ago