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gerrit-public.fairphone.software / platform / external / webrtc / af9e6637c0fd154a6469a07913e233aad2b4d8f0
  1. af9e663 Make rtc::CriticalSection lockable from f() const. by Peter Boström · 9 years ago
  2. 3c16978 Remove cast to LocalAudioSource from AudioRtpSender. by Tommi · 9 years ago
  3. 32be07b Remove RentACodec::GetEncoderStack by kwiberg · 9 years ago
  4. 693a114 Add stefan@webrtc.org to webrtc/test/OWNERS. by Peter Boström · 9 years ago
  5. 3313ec9 Enable transport seq num extension on receive channel to suppress log warning. by stefan · 9 years ago
  6. d664836 Added EncodedImage::GetBufferPaddingBytes. by hbos · 9 years ago
  7. 429c345 Fixes a bug which incorrectly logs incoming RTCP as outgoing. by terelius · 9 years ago
  8. b304e26 Roll chromium_revision 1728ddf..4623ce8 (370595:370665) by kjellander · 9 years ago
  9. 1f150b3 Add new NetEq resources to modules_unittests.isolate. by kjellander · 9 years ago
  10. 902c03e rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni. by hbos · 9 years ago
  11. 0b98cf7 Delete CaptureRenderAdapter::VideoRenderInfo struct, it is unused since the recent deletion of SetSize. by nisse · 9 years ago
  12. 5082c83 Make type and constructors in EglBase14 public. by noahric · 9 years ago
  13. becf9ee Roll chromium_revision 6a04368..1728ddf (370362:370595) by kjellander · 9 years ago
  14. d9f641e Reallocate encoded buffer size if needed. Initially set to the input image size. by asapersson · 9 years ago
  15. d26fadb Delete GetRenderer method, used only by the tests. by nisse · 9 years ago
  16. 057ecf0 Making WebRtcSession fire a destroyed signal. by deadbeef · 9 years ago
  17. da99da8 Update API for Objective-C RTCPeerConnectionFactory. by Jon Hjelle · 9 years ago
  18. 065aacc Move RTCVideoSource to webrtc/api/objc. by Jon Hjelle · 9 years ago
  19. d8dccd5 uses standard types instead of RTCPUtility type to store data. by danilchap · 9 years ago
  20. 72c08ed Reenables several NetEq unittests on android. by ivoc · 9 years ago
  21. 32f8154 Support REMB in combination with send-side BWE. by stefan · 9 years ago
  22. a5dec16 Name SimulcastEncoderApdater on InitEncode. by Peter Boström · 9 years ago
  23. a2b4c40 Roll chromium_revision 15d94b7..6a04368 (370289:370362) by kjellander · 9 years ago
  24. 9090e0b Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding. by Tommi · 9 years ago
  25. 84df580 Switch to rtc::CriticalSection in IncomingVideoStream and remove one lock. by tommi · 9 years ago
  26. e849332 Remove ConditionVariableWrapper. by Tommi · 9 years ago
  27. 63cb434 Switch use of CriticalSectionWrapper -> rtc::CriticalSection in call/ by tommi · 9 years ago
  28. 1d61a51 Send key frame if time difference between incoming frames exceeds a certain limit. by asapersson · 9 years ago
  29. 436ff31 Update exclude files for renamed test by kjellander · 9 years ago
  30. a927dcf Roll chromium_revision 542b77a..15d94b7 (370158:370289) by kjellander · 9 years ago
  31. f0b8a37 Allow disabling denoiser when it is enabled. by jackychen · 9 years ago
  32. 3a6bf2d Enable full screen windows to be shown in window picker for mac. Before this patch a full screen window can be shared if sharing is started before the window is entered into full screen mode, but not if it's already in full screen. by niklas.enbom · 9 years ago
  33. 95c8b40 Roll chromium_revision f527e86..542b77a (370073:370158) by kjellander · 9 years ago
  34. f01ea4f Remove use of ConditionVariableWrapper and CriticalSectionWrapper from UdpSocket2Windows. by Tommi · 9 years ago
  35. cd255cc Remove unused ConditionVariableWrapper on POSIX platforms by tommi · 9 years ago
  36. 7b971e7 Remove extra_options from VideoCodec. by Peter Boström · 9 years ago
  37. ee5a309 Make CriticalSectionWrapper non-virtual. by Tommi · 9 years ago
  38. dd45eb6 Remove use-after-free when quality tests stall. by Peter Boström · 9 years ago
  39. 8a2c31d Make it possible to run peerconnection_unittests on Android. by perkj · 9 years ago
  40. 0edb05b Declare that rent_a_codec depends on the audio codecs by kwiberg · 9 years ago
  41. 73674f8 Replace hardcoded constant in video capture with macro. by kjellander · 9 years ago
  42. 3c85cad Roll chromium_revision 7a4fb8d..f527e86 (370025:370073) by kjellander · 9 years ago
  43. 61046eb Rename RWLockGeneric to RWLockWinXP to more accurately reflect when it's used. by tommi · 9 years ago
  44. 3860c7f Fix parsing of CLANG_REVISON from tools/clang/scripts/update.py by kjellander · 9 years ago
  45. c4c8485 Deleted renderer-related SetSize methods, and all uses. by nisse · 9 years ago
  46. 81354f5 Added mute logic to VideoTrackRenderers. by nisse · 9 years ago
  47. 8d6fab8 Remove two dead 'using' instances. by Peter Boström · 9 years ago
  48. 2067826 Remove dependency on ConditionVariableWrapper and CriticalSectionWrapper in UdpSocketPosix. by Tommi · 9 years ago
  49. 233bfd2 Move keyframe requests outside encoder mutex. by Peter Boström · 9 years ago
  50. 49c7402 Roll chromium_revision ad2f344..7a4fb8d (370010:370025) by kjellander · 9 years ago
  51. aff4b70 Simplify the implementation of LoggingTest. by tommi · 9 years ago
  52. f8c2bac Add a gyp/gn variable for whether to use iLBC or not by kwiberg · 9 years ago
  53. f5a3a93 Add 5-argument wrapper WebRtcVideoFrame::InitToBlack by Niels Möller · 9 years ago
  54. d142067 Roll chromium_revision 1c9621e..ad2f344 (369979:370010) by kjellander · 9 years ago
  55. 34ed2b9 [rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function by danilchap · 9 years ago
  56. 8b1e431 Delete remnants of non-square pixel support from cricket::VideoFrame. by nisse · 9 years ago
  57. 33c1dca Roll chromium_revision 89ca041..1c9621e (369966:369979) by kjellander · 9 years ago
  58. 9d2a3c5 Roll chromium_revision 4b805fe..89ca041 (369965:369966) by kjellander · 9 years ago
  59. e110e5c Roll chromium_revision 6058a7b..4b805fe (369961:369965) by kjellander · 9 years ago
  60. d7db862 Roll chromium_revision 9e8fb7a..6058a7b (369957:369961) by kjellander · 9 years ago
  61. c1cf0d3 Roll chromium_revision 0a79aa1..9e8fb7a (369950:369957) by kjellander · 9 years ago
  62. 011df0a Roll chromium_revision 553c2cb..0a79aa1 (369932:369950) by kjellander · 9 years ago
  63. f624a22 Roll chromium_revision 46fd746..553c2cb (369797:369932) by kjellander · 9 years ago
  64. 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
  65. 56271ed fix bug 5430 by guoweis · 9 years ago
  66. f4decb5 Add QP statistics logging to Android HW encoder. by glaznev · 9 years ago
  67. 305ca25 Roll chromium_revision ff895e2..46fd746 (369726:369797) by kjellander · 9 years ago
  68. 884f585 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  69. 1567d0b [rtp_rtcp] rtcp::Sdes moved into own file by Danil Chapovalov · 9 years ago
  70. 79a5a83 Adapt to boringssl's new defaults. by torbjorng · 9 years ago
  71. 2c13297 [rtp_rtcp] rtcp::Rpsi moved into own file by Danil Chapovalov · 9 years ago
  72. 256e5b2 Cleaning/Parsing will be done in the https://codereview.webrtc.org/1557593002/ by Danil Chapovalov · 9 years ago
  73. a132197 Roll chromium_revision 6e188de..ff895e2 (369712:369726) by kjellander · 9 years ago
  74. 5679da1 [rtp_rtcp] rtcp::Fir moved into own file by Danil Chapovalov · 9 years ago
  75. a5eba6c [rtp_rtcp] rtcp::Remb moved into own file by Danil Chapovalov · 9 years ago
  76. d66b44d Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 9 years ago
  77. 74e8df81 Roll chromium_revision 9946592..6e188de (369667:369712) by kjellander · 9 years ago
  78. 0f7d293 Revert changes to default option setting in https://codereview.webrtc.org/1500633002/ by solenberg · 9 years ago
  79. 5602f65 setup_links.py fix so that FFmpeg compiles on windows. by hbos · 9 years ago
  80. 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
  81. 301830f Roll chromium_revision 099be58..9946592 (369139:369667) by kjellander · 9 years ago
  82. dc305db Add ApplyPacketOptions() by Sergey Ulanov · 9 years ago
  83. 20ac434 Fix a test bot failure. by Honghai Zhang · 9 years ago
  84. e1f9d83 Adding AddTrack/RemoveTrack to native PeerConnection API. by deadbeef · 9 years ago
  85. d9e62f5 Fixed sending Rtp packets with non zero csrcs and certain extensions. by danilchap · 9 years ago
  86. 67b1e1a Put options as the argument of the java PeerConnectionFactory constructor. by honghaiz · 9 years ago
  87. 5d332ac Fix expectation bug in the RTPSender unit test. by terelius · 9 years ago
  88. 04cb763 Add tests for verifying transport feedback for audio and video. by Stefan Holmer · 9 years ago
  89. fcfc804 Eliminate defines in talk/ by kjellander · 9 years ago
  90. 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
  91. 2734d77 Remove assert which was incorrectly added to TcpPort::OnSentPacket. by Stefan Holmer · 9 years ago
  92. 55674ff Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. by Stefan Holmer · 9 years ago
  93. 31c8d2e Update with new default boringssl no-aes cipher suites. Re-enable tests. by Torbjorn Granlund · 9 years ago
  94. 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
  95. 688e308 Re-land: "Use an explicit identifier in Config" by aluebs · 9 years ago
  96. 7307952 Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. by Stefan Holmer · 9 years ago
  97. 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
  98. 35aae2e Remove libfuzzer trybot from default trybot set. by kjellander · 9 years ago
  99. ff2a635 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 9 years ago
  100. 709513d Delete remnants of non-square pixel support from cricket::VideoFrame. by nisse · 9 years ago