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gerrit-public.fairphone.software / platform / external / webrtc / 9d2a3c51d9bfb5988e707dc99e12475e796b77fc
  1. 9d2a3c5 Roll chromium_revision 4b805fe..89ca041 (369965:369966) by kjellander · 9 years ago
  2. e110e5c Roll chromium_revision 6058a7b..4b805fe (369961:369965) by kjellander · 9 years ago
  3. d7db862 Roll chromium_revision 9e8fb7a..6058a7b (369957:369961) by kjellander · 9 years ago
  4. c1cf0d3 Roll chromium_revision 0a79aa1..9e8fb7a (369950:369957) by kjellander · 9 years ago
  5. 011df0a Roll chromium_revision 553c2cb..0a79aa1 (369932:369950) by kjellander · 9 years ago
  6. f624a22 Roll chromium_revision 46fd746..553c2cb (369797:369932) by kjellander · 9 years ago
  7. cec0a08 Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set. by honghaiz · 9 years ago
  8. 56271ed fix bug 5430 by guoweis · 9 years ago
  9. f4decb5 Add QP statistics logging to Android HW encoder. by glaznev · 9 years ago
  10. 305ca25 Roll chromium_revision ff895e2..46fd746 (369726:369797) by kjellander · 9 years ago
  11. 884f585 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  12. 1567d0b [rtp_rtcp] rtcp::Sdes moved into own file by Danil Chapovalov · 9 years ago
  13. 79a5a83 Adapt to boringssl's new defaults. by torbjorng · 9 years ago
  14. 2c13297 [rtp_rtcp] rtcp::Rpsi moved into own file by Danil Chapovalov · 9 years ago
  15. 256e5b2 Cleaning/Parsing will be done in the https://codereview.webrtc.org/1557593002/ by Danil Chapovalov · 9 years ago
  16. a132197 Roll chromium_revision 6e188de..ff895e2 (369712:369726) by kjellander · 9 years ago
  17. 5679da1 [rtp_rtcp] rtcp::Fir moved into own file by Danil Chapovalov · 9 years ago
  18. a5eba6c [rtp_rtcp] rtcp::Remb moved into own file by Danil Chapovalov · 9 years ago
  19. d66b44d Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 9 years ago
  20. 74e8df81 Roll chromium_revision 9946592..6e188de (369667:369712) by kjellander · 9 years ago
  21. 0f7d293 Revert changes to default option setting in https://codereview.webrtc.org/1500633002/ by solenberg · 9 years ago
  22. 5602f65 setup_links.py fix so that FFmpeg compiles on windows. by hbos · 9 years ago
  23. 6a59ad3 Revert of Remove libfuzzer trybot from default trybot set. (patchset #1 id:1 of https://codereview.webrtc.org/1585963002/ ) by kjellander · 9 years ago
  24. 301830f Roll chromium_revision 099be58..9946592 (369139:369667) by kjellander · 9 years ago
  25. dc305db Add ApplyPacketOptions() by Sergey Ulanov · 9 years ago
  26. 20ac434 Fix a test bot failure. by Honghai Zhang · 9 years ago
  27. e1f9d83 Adding AddTrack/RemoveTrack to native PeerConnection API. by deadbeef · 9 years ago
  28. d9e62f5 Fixed sending Rtp packets with non zero csrcs and certain extensions. by danilchap · 9 years ago
  29. 67b1e1a Put options as the argument of the java PeerConnectionFactory constructor. by honghaiz · 9 years ago
  30. 5d332ac Fix expectation bug in the RTPSender unit test. by terelius · 9 years ago
  31. 04cb763 Add tests for verifying transport feedback for audio and video. by Stefan Holmer · 9 years ago
  32. fcfc804 Eliminate defines in talk/ by kjellander · 9 years ago
  33. 3542013 Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) by sprang · 9 years ago
  34. 2734d77 Remove assert which was incorrectly added to TcpPort::OnSentPacket. by Stefan Holmer · 9 years ago
  35. 55674ff Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. by Stefan Holmer · 9 years ago
  36. 31c8d2e Update with new default boringssl no-aes cipher suites. Re-enable tests. by Torbjorn Granlund · 9 years ago
  37. e5e0e57 Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ ) by tommi · 9 years ago
  38. 688e308 Re-land: "Use an explicit identifier in Config" by aluebs · 9 years ago
  39. 7307952 Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. by Stefan Holmer · 9 years ago
  40. 268493a Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ ) by nisse · 9 years ago
  41. 35aae2e Remove libfuzzer trybot from default trybot set. by kjellander · 9 years ago
  42. ff2a635 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 9 years ago
  43. 709513d Delete remnants of non-square pixel support from cricket::VideoFrame. by nisse · 9 years ago
  44. beed828 Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop(). by Sergey Ulanov · 9 years ago
  45. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  46. 8432e1f Re-enable tests that failed under Linux_Msan. by marpan · 9 years ago
  47. fca54f4 Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) by tommi · 9 years ago
  48. 09d944f Roll chromium_revision 346fea9..099be58 (369082:369139) by kjellander · 9 years ago
  49. 306efad Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan by kjellander · 9 years ago
  50. 292e192 Add build_protobuf variable. by kjellander · 9 years ago
  51. a276e73 Clean the code for external denoiser. by jackychen · 9 years ago
  52. 2f7dea1 [rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way by danilchap · 9 years ago
  53. ea8c0f6 Fix capture ntp time issue introduced with r11187. by Stefan Holmer · 9 years ago
  54. 365543d Roll chromium_revision 131167b..346fea9 (368784:369082) by kjellander · 9 years ago
  55. 25249d9 Use an explicit identifier in Config by aluebs · 9 years ago
  56. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  57. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  58. 92e677a [rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function by danilchap · 9 years ago
  59. 5584bf4 Make :rtc_base_approved a public dep of :rtc_base. by jbroman · 9 years ago
  60. e84e96e NetEq: Fix a typo in a comment by Henrik Lundin · 9 years ago
  61. 36220ae Slap deprecation notices on Pass methods by kwiberg · 9 years ago
  62. d20e651 Fix test bug introduced in r11101. by Stefan Holmer · 9 years ago
  63. 3e1cfa7 Delete unused method webrtc::VideoRendererInterface::SetSize. by nisse · 9 years ago
  64. 3235a27 Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg. by Henrik Boström · 9 years ago
  65. 2845a02 Remove unused enum RTPDirections. by terelius · 9 years ago
  66. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 9 years ago
  67. 6183de6 Remove tools/refactoring. by Peter Boström · 9 years ago
  68. 127782b Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal. by nisse · 9 years ago
  69. 16979e3 Update .gitignore by Henrik Kjellander · 9 years ago
  70. 67e94fb Add unit test for stand-alone denoiser and fixed some bugs. by jackychen · 9 years ago
  71. b2328d1 Remove additional channel constraints when Beamforming is enabled in AudioProcessing by aluebs · 9 years ago
  72. e93ad1b Roll chromium_revision 8c958e0..131167b (368561:368784) by kjellander · 9 years ago
  73. 2a34688 Make Beamforming dynamically settable for Android platform builds by aluebs · 9 years ago
  74. 2bc63a1 clang-format audio_device/mac. by andrew · 9 years ago
  75. a7446d2 Change DTLS default from 1.0 to 1.2 for webrtc. by Guo-wei Shieh · 9 years ago
  76. f6c318e Update API for Objective-C RTCMediaSource. by Jon Hjelle · 9 years ago
  77. e799bad Move Objective-C video renderers to webrtc/api/objc. by Jon Hjelle · 9 years ago
  78. 8102879 Update API for Objective-C RTCMediaStreamTrack. by Jon Hjelle · 9 years ago
  79. a2c353f Update API for Objective-C RTCStats. by Jon Hjelle · 9 years ago
  80. 7e8145f [rtp_rtcp] rtcp::Tmmbr moved into own file by danilchap · 9 years ago
  81. 27ed3cc SCTP: Stopped accepting SSRCs higher than max. Seems to fix asan-related crash. by lally · 9 years ago
  82. a9a1d2a H.264: Default flags and pulling in openh264 and ffmpeg. by hbos · 9 years ago
  83. 7823495 Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes. by Jon Hjelle · 9 years ago
  84. fd99dea Roll chromium_revision 42ab10e..8c958e0 (368534:368561) by kjellander · 9 years ago
  85. ef3d805 [rtp_rtcp] rtcp::Tmmbn moved into own file explicetly unchanged. by danilchap · 9 years ago
  86. d36efeb Roll chromium_revision e738b54..42ab10e (368533:368534) by kjellander · 9 years ago
  87. 4de0037 Roll chromium_revision 7d97c94..e738b54 (368514:368533) by kjellander · 9 years ago
  88. 3c05e6c Disable EndToEndTest.TransportSeqNumOnAudioAndVideo for Dr Memory. by kjellander · 9 years ago
  89. daa8749 Revert of Roll chromium_revision 7d97c94..951c006 (368514:368525) (patchset #1 id:1 of https://codereview.webrtc.org/1577573002/ ) by guoweis · 9 years ago
  90. db21f63 fix GN build break on native_client by Guo-wei Shieh · 9 years ago
  91. 6109fc1 Roll chromium_revision 7d97c94..951c006 (368514:368525) by kjellander · 9 years ago
  92. 0697db6 Roll chromium_revision 8a15a7f..7d97c94 (368391:368514) by kjellander · 9 years ago
  93. 684e995 Disable 2 video tests which fail on DrMemoryFull by Guo-wei Shieh · 9 years ago
  94. f475d36 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 9 years ago
  95. 25702cb Misc. small cleanups. by pkasting · 9 years ago
  96. 5de688e Roll chromium_revision ede5d4f..8a15a7f (368310:368391) by kjellander · 9 years ago
  97. 49c454e Cleaning neteq_unittest resource files. by minyue · 9 years ago
  98. f1685c7 Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac by kjellander · 9 years ago
  99. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 9 years ago
  100. 37ebcf0 Reland "Add APK targets to build libjingle tests for Android." by phoglund · 9 years ago