Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 7aba0297e65c3413c88a93c78435dc369e86fe46
  1. 7aba029 Make use of new APM statistics interface. by ivoc · 8 years ago
  2. 25b57ce Update header formatters to FlexFEC draft 03. by brandtr · 8 years ago
  3. 7c2f174 Roll chromium_revision f1e2718a3f..5c396eba99 (431838:431855) by buildbot · 8 years ago
  4. 445fb8f Use correct define in H264 end-to-end tests. by brandtr · 8 years ago
  5. 8e75a52 Explicitly use RTX for RED in VideoQualityTest and video_loopback. by brandtr · 8 years ago
  6. a65704b Expose RtpCodecParameters to VideoMediaInfo stats. by hbos · 8 years ago
  7. 82ebe02 Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed]. by hbos · 8 years ago
  8. 6f0b9fd Allowing resetting of AudioNetworkAdaptor in AudioSendStream. by minyue · 8 years ago
  9. 218f436 Roll chromium_revision 1a6cf4da7c..f1e2718a3f (431807:431838) by kjellander · 8 years ago
  10. 8c2ec19 Revert of Temporarily remove ios_api_framework from the commit queue. (patchset #2 id:40001 of https://codereview.webrtc.org/2341113004/ ) by kjellander · 8 years ago
  11. 931e975 Roll chromium_revision fd046d62bd..1a6cf4da7c (431804:431807) by buildbot · 8 years ago
  12. 59695e5 Roll chromium_revision b3671bcbd9..fd046d62bd (431798:431804) by buildbot · 8 years ago
  13. e5d3e7e Roll chromium_revision 7a2fab5553..b3671bcbd9 (431793:431798) by buildbot · 8 years ago
  14. d73cbbf Roll chromium_revision 4762e7cc7d..7a2fab5553 (431782:431793) by buildbot · 8 years ago
  15. f823ede Negotiate H264 profiles in SDP by magjed · 8 years ago
  16. 13ceeea Revert of H.264 packetization mode 0 (try 2) (patchset #27 id:520001 of https://codereview.webrtc.org/2337453002/ ) by magjed · 8 years ago
  17. b73a751 Roll chromium_revision 13d2c57f10..4762e7cc7d (431754:431782) by buildbot · 8 years ago
  18. e7d17f7 Roll chromium_revision 51641c7dde..13d2c57f10 (431606:431754) by buildbot · 8 years ago
  19. 372719b Remove screen_capturer_mock_objects.h by zijiehe · 8 years ago
  20. 3074096 Crash in DirectX capturer by zijiehe · 8 years ago
  21. c9a6e4a CroppingWindowCapturer::CreateCapturer() function to replace raw pointer version by zijiehe · 8 years ago
  22. b7510d3 Roll chromium_revision 4cee38b9a3..51641c7dde (431537:431606) by buildbot · 8 years ago
  23. 74f9d98 Remove unused warning suppression by kthelgason · 8 years ago
  24. 2814598 Revert of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2494683006/ ) by hta · 8 years ago
  25. c63fb3a Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491933002/ ) by hta · 8 years ago
  26. b05fa24 Optimize FindCodecById and ReferencedCodecsMatch by magjed · 8 years ago
  27. acd935b Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ ) by nisse · 8 years ago
  28. e6f98c7 Remove RED/RTX workaround from sender/receiver and VideoEngine2. by brandtr · 8 years ago
  29. ea51d08 Roll chromium_revision 09f4d43e57..4cee38b9a3 (431473:431537) by buildbot · 8 years ago
  30. 0b3a638 Ensures that AudioDeviceBuffer::StopPeriodicLogging works as intended. by henrika · 8 years ago
  31. 2a615fc Reduce taking locks in RTPSenderVideo::SendVideo by danilchap · 8 years ago
  32. e918a13 MB: Add new Win8 and Win10 bots. by Henrik Kjellander · 8 years ago
  33. 98903d2 Remove ScreenCapturer and WindowCapturer by zijiehe · 8 years ago
  34. 3bba101 Implement H.264 packetization mode 0. by hta · 8 years ago
  35. e95f5bf Roll chromium_revision 3a46cc72d5..09f4d43e57 (431304:431473) by buildbot · 8 years ago
  36. a3095d0 Re-enable the P2PTransportChannelMultihomedTest.TestBasic by zhihuang · 8 years ago
  37. bd162f7 Roll chromium_revision 65cb3abb07..3a46cc72d5 (431231:431304) by buildbot · 8 years ago
  38. e3fe4a7 Update VideoFrameBuffer-related methods to not use references to scoped_refptr. by nisse · 8 years ago
  39. 7fe6db9 Revert of Declare VideoCodec.codec_specific_info private (patchset #5 id:80001 of https://codereview.webrtc.org/2452963002/ ) by hta · 8 years ago
  40. 1369c83 Revert of Issue 2434073003: Extract bitrate allocation ... (patchset #4 id:60001 of https://codereview.webrtc.org/2488833004/ ) by sprang · 8 years ago
  41. ef6cbae Add UMA histogram for Echo likelihood. by ivoc · 8 years ago
  42. 44c7ecf Added a public GN config to compile mock headers. by aleloi · 8 years ago
  43. 647bf43 Reland of Issue 2434073003: Extract bitrate allocation ... by sprang · 8 years ago
  44. 9d1315a Add UMA stats for the resolution camera is started in. by sakal · 8 years ago
  45. d64bf6f MB: Add Mac Asan (swarming) to FYI waterfall. by ehmaldonado · 8 years ago
  46. aae4fa8 Set videoCapturerStopped when PeerConnectionClient stopped by hekra01 · 8 years ago
  47. 917a4ee Replace SequencedTaskChecker in RTPSenderVideo by kthelgason · 8 years ago
  48. 1b35d4c Don't copy loop variables when we don't have to by kwiberg · 8 years ago
  49. dbdb3f1 Wire up FlexfecSender in RTPSender and add unit tests. by brandtr · 8 years ago
  50. 131bc49 Wire up FlexfecSender in RTPSenderVideo. by brandtr · 8 years ago
  51. ef3f768 Roll chromium_revision 6be6a794e7..65cb3abb07 (431208:431231) by buildbot · 8 years ago
  52. 3cf8ece Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ ) by magjed · 8 years ago
  53. 7927386 Declare VideoCodec.codec_specific_info private by hta · 8 years ago
  54. 59be5f7 Add helper functions for negotiating H264 profile level id by magjed · 8 years ago
  55. 5ec208f Use bitcode by default for ios framework build. by kthelgason · 8 years ago
  56. 5de52fd Created a mocked AudioTransport. by aleloi · 8 years ago
  57. 37e4ad5 Roll chromium_revision cc82e2a1fa..6be6a794e7 (428909:431208) by buildbot · 8 years ago
  58. 51e9608 AudioDeviceBuffer now uses 16-bit buffers by henrika · 8 years ago
  59. 9f71ec5 Stop caching supported codecs in WebRtcVideoEngine2 by magjed · 8 years ago
  60. 3045589 Remove references of ScreenCapturer and WindowCapturer by zijiehe · 8 years ago
  61. 6bedebf First step in providing a UdpTransportChannel. by johan · 8 years ago
  62. cc2d1c6 Bump mac deployment target to 10.9 by kthelgason · 8 years ago
  63. ccf1021 Add screenshare support to AppRTCMobile. by sakal · 8 years ago
  64. 2256e04 Rename media constraints model and store. by denicija · 8 years ago
  65. 4bc98d4 Revert of Extract bitrate allocation of spatial/temporal layers out of codec impl. (patchset #17 id:320001 of https://codereview.webrtc.org/2434073003/ ) by sprang · 8 years ago
  66. 6ba58d6 Reland of Simplify and extend RtpHeaderExtensionMap (patchset #1 id:1 of https://codereview.webrtc.org/2484863007/ ) by danilchap · 8 years ago
  67. 16e3caa Removed unused forward declaration. by aleloi · 8 years ago
  68. 8f46c67 Extract bitrate allocation of spatial/temporal layers out of codec impl. by sprang · 8 years ago
  69. 592baaf Add build flag to enable bitcode on ios by kthelgason · 8 years ago
  70. b04b5c2 Reland of Add bitrate section to settings view controller. (patchset #1 id:1 of https://codereview.webrtc.org/2488653002/ ) by denicija · 8 years ago
  71. d22f6bc Add sakal@ as owner of webrtc/examples/androidapp/ by magjed · 8 years ago
  72. faa7689 Remove cyclic dependency in gyp build by kthelgason · 8 years ago
  73. 28ec6bd Make releaseEglSurface in EglRenderer asynchronous. by sakal · 8 years ago
  74. cbe7435 Reland of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #1 id:1 of https://codereview.webrtc.org/2483273002/ ) by Kári Tristan Helgason · 8 years ago
  75. 7252a00 Ping the premier connection on each network with higher priority. by honghaiz · 8 years ago
  76. d924ff3 iOS: Fix improved BWE field trial. by tkchin · 8 years ago
  77. 669d69b Use rtc::PacketTransportInterface in WebrtcSession unit test. by johan · 8 years ago
  78. bc38b4d Revert of Simplify and extend RtpHeaderExtensionMap (patchset #12 id:260001 of https://codereview.webrtc.org/2452293004/ ) by danilchap · 8 years ago
  79. e083909 Remove evil defines out of shared_x_display.h by zijiehe · 8 years ago
  80. d1d26fb Simplify and extend RtpHeaderExtensionMap by danilchap · 8 years ago
  81. d69ad84 Revert of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #2 id:20001 of https://codereview.webrtc.org/2463313002/ ) by kwiberg · 8 years ago
  82. 671c0ab GN: Add //build/toolchain/toolchain.gni to exec_script_whitelist by Henrik Kjellander · 8 years ago
  83. f94ba46 Update BWE_TEST_LOGGING_PLOT output format, and fix plot_dynamics.py script. by terelius · 8 years ago
  84. 579729d GN: Build tools and examples by default. by ehmaldonado · 8 years ago
  85. e239779 Reenable P2P test case and remove outdated TODO by kthelgason · 8 years ago
  86. 9561183 Changed mixing to be done at the minimal possible frequency. by aleloi · 8 years ago
  87. 37d7a22 GN: Don't build tests by default. by ehmaldonado · 8 years ago
  88. 09bc128 RTCMediaStream[Track]Stats added. by hbos · 8 years ago
  89. 8c375de Expose bit rate property in ARDAppClient and set max bitrate for video RTCRtcSender. by denicija · 8 years ago
  90. 56827ef Remove GetPreferredTryMasters from PRESUBMIT. by kjellander · 8 years ago
  91. 4fcc4d8 Expand CQ dry run permission to people with tryjob access. by tandrii · 8 years ago
  92. 6eaa558 Disabling NOTREACHED which we're hitting flakily in browser tests. by phoglund · 8 years ago
  93. 81da488 Added audio mixer and removed audio device module in AudioState::Config. by aleloi · 8 years ago
  94. 40532a1 Revert of Add bitrate section to settings view controller. (patchset #1 id:1 of https://codereview.webrtc.org/2473783003/ ) by denicija · 8 years ago
  95. 5d78e8d Remove audio from BitrateEstimatorTest. by aleloi · 8 years ago
  96. eee9c0e GN: Move xmllite_xmpp_unittests into webrtc/libjingle by kjellander · 8 years ago
  97. a92704e Add function for getting supported H264 level from max resolution and fps by magjed · 8 years ago
  98. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  99. b5b0b98 Allow webrtc::test::VcmCapturer to be updated with its current sink. by philipel · 8 years ago
  100. b6e857d Remove deprecated methods from CameraEnumerationAndroid. by sakal · 8 years ago