Google Git
Sign in
android / platform / external / webrtc_legacy / HEAD
  1. 410d787 Merge commit \'a4acd9d6bc9b3b033d7d274316e75ee067df8d20\' into HEAD by Bill Yi · 8 years ago
  2. 19b0dfa Merge commit 'a4acd9d6bc9b3b033d7d274316e75ee067df8d20' into HEAD by Bill Yi · 8 years ago
  3. eca4242 Merge "Add missing liblog dependency" am: 6dc914a60e by Dimitry Ivanov · 8 years ago
  4. 6dc914a Merge "Add missing liblog dependency" by Dimitry Ivanov · 8 years ago
  5. 9ea1ad1 Add missing liblog dependency by Dimitry Ivanov · 8 years ago
  6. 4805e82 Merge "Suppress unused-parameter warnings." am: 14a01a403a by Chih-hung Hsieh · 8 years ago
  7. 14a01a4 Merge "Suppress unused-parameter warnings." by Chih-hung Hsieh · 8 years ago
  8. 4e188dd Suppress unused-parameter warnings. by Chih-Hung Hsieh · 8 years ago
  9. b3cb8ab Merge "Merge upstream SHA 04cb763" am: 9a337512d9 by Chih-hung Hsieh · 8 years ago
  10. 9a33751 Merge "Merge upstream SHA 04cb763" by Chih-hung Hsieh · 8 years ago
  11. daef292 Merge upstream SHA 04cb763 by Alex Luebs · 8 years ago
  12. 04cb763 Add tests for verifying transport feedback for audio and video. by Stefan Holmer · 8 years ago
  13. fcfc804 Eliminate defines in talk/ by kjellander · 8 years ago
  14. 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 · 8 years ago
  15. 2734d77 Remove assert which was incorrectly added to TcpPort::OnSentPacket. by Stefan Holmer · 8 years ago
  16. 55674ff Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. by Stefan Holmer · 8 years ago
  17. 31c8d2e Update with new default boringssl no-aes cipher suites. Re-enable tests. by Torbjorn Granlund · 8 years ago
  18. e5e0e57 Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ ) by tommi · 8 years ago
  19. 688e308 Re-land: "Use an explicit identifier in Config" by aluebs · 8 years ago
  20. 7307952 Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. by Stefan Holmer · 8 years ago
  21. 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 · 8 years ago
  22. 35aae2e Remove libfuzzer trybot from default trybot set. by kjellander · 8 years ago
  23. ff2a635 Add ramp-up tests for transport sequence number with and w/o audio. by Stefan Holmer · 8 years ago
  24. 709513d Delete remnants of non-square pixel support from cricket::VideoFrame. by nisse · 8 years ago
  25. beed828 Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop(). by Sergey Ulanov · 8 years ago
  26. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 8 years ago
  27. 8432e1f Re-enable tests that failed under Linux_Msan. by marpan · 8 years ago
  28. fca54f4 Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) by tommi · 8 years ago
  29. 09d944f Roll chromium_revision 346fea9..099be58 (369082:369139) by kjellander · 8 years ago
  30. 306efad Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan by kjellander · 8 years ago
  31. 292e192 Add build_protobuf variable. by kjellander · 8 years ago
  32. a276e73 Clean the code for external denoiser. by jackychen · 8 years ago
  33. 2f7dea1 [rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way by danilchap · 8 years ago
  34. ea8c0f6 Fix capture ntp time issue introduced with r11187. by Stefan Holmer · 8 years ago
  35. 365543d Roll chromium_revision 131167b..346fea9 (368784:369082) by kjellander · 8 years ago
  36. 25249d9 Use an explicit identifier in Config by aluebs · 8 years ago
  37. e591f93 Storing raw audio sink for default audio track. by deadbeef · 8 years ago
  38. 6955870 Convert channel counts to size_t. by Peter Kasting · 8 years ago
  39. 92e677a [rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function by danilchap · 8 years ago
  40. 5584bf4 Make :rtc_base_approved a public dep of :rtc_base. by jbroman · 8 years ago
  41. e84e96e NetEq: Fix a typo in a comment by Henrik Lundin · 8 years ago
  42. 36220ae Slap deprecation notices on Pass methods by kwiberg · 8 years ago
  43. d20e651 Fix test bug introduced in r11101. by Stefan Holmer · 8 years ago
  44. 3e1cfa7 Delete unused method webrtc::VideoRendererInterface::SetSize. by nisse · 8 years ago
  45. 3235a27 Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg. by Henrik Boström · 8 years ago
  46. 2845a02 Remove unused enum RTPDirections. by terelius · 8 years ago
  47. 3842c5c Wire-up BWE feedback for audio receive streams. by Stefan Holmer · 8 years ago
  48. 6183de6 Remove tools/refactoring. by Peter Boström · 8 years ago
  49. 127782b Add default dummy implementation of cricket::VideoRenderer::SetSize, to easy later removal. by nisse · 8 years ago
  50. 16979e3 Update .gitignore by Henrik Kjellander · 8 years ago
  51. 67e94fb Add unit test for stand-alone denoiser and fixed some bugs. by jackychen · 8 years ago
  52. b2328d1 Remove additional channel constraints when Beamforming is enabled in AudioProcessing by aluebs · 8 years ago
  53. e93ad1b Roll chromium_revision 8c958e0..131167b (368561:368784) by kjellander · 8 years ago
  54. 2a34688 Make Beamforming dynamically settable for Android platform builds by aluebs · 8 years ago
  55. 2bc63a1 clang-format audio_device/mac. by andrew · 8 years ago
  56. a7446d2 Change DTLS default from 1.0 to 1.2 for webrtc. by Guo-wei Shieh · 8 years ago
  57. f6c318e Update API for Objective-C RTCMediaSource. by Jon Hjelle · 8 years ago
  58. e799bad Move Objective-C video renderers to webrtc/api/objc. by Jon Hjelle · 8 years ago
  59. 8102879 Update API for Objective-C RTCMediaStreamTrack. by Jon Hjelle · 8 years ago
  60. a2c353f Update API for Objective-C RTCStats. by Jon Hjelle · 8 years ago
  61. 7e8145f [rtp_rtcp] rtcp::Tmmbr moved into own file by danilchap · 8 years ago
  62. 27ed3cc SCTP: Stopped accepting SSRCs higher than max. Seems to fix asan-related crash. by lally · 8 years ago
  63. a9a1d2a H.264: Default flags and pulling in openh264 and ffmpeg. by hbos · 8 years ago
  64. 7823495 Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes. by Jon Hjelle · 8 years ago
  65. fd99dea Roll chromium_revision 42ab10e..8c958e0 (368534:368561) by kjellander · 8 years ago
  66. ef3d805 [rtp_rtcp] rtcp::Tmmbn moved into own file explicetly unchanged. by danilchap · 8 years ago
  67. d36efeb Roll chromium_revision e738b54..42ab10e (368533:368534) by kjellander · 8 years ago
  68. 4de0037 Roll chromium_revision 7d97c94..e738b54 (368514:368533) by kjellander · 8 years ago
  69. 3c05e6c Disable EndToEndTest.TransportSeqNumOnAudioAndVideo for Dr Memory. by kjellander · 8 years ago
  70. daa8749 Revert of Roll chromium_revision 7d97c94..951c006 (368514:368525) (patchset #1 id:1 of https://codereview.webrtc.org/1577573002/ ) by guoweis · 8 years ago
  71. db21f63 fix GN build break on native_client by Guo-wei Shieh · 8 years ago
  72. 6109fc1 Roll chromium_revision 7d97c94..951c006 (368514:368525) by kjellander · 8 years ago
  73. 0697db6 Roll chromium_revision 8a15a7f..7d97c94 (368391:368514) by kjellander · 8 years ago
  74. 684e995 Disable 2 video tests which fail on DrMemoryFull by Guo-wei Shieh · 8 years ago
  75. f475d36 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 8 years ago
  76. 25702cb Misc. small cleanups. by pkasting · 8 years ago
  77. 5de688e Roll chromium_revision ede5d4f..8a15a7f (368310:368391) by kjellander · 8 years ago
  78. 49c454e Cleaning neteq_unittest resource files. by minyue · 8 years ago
  79. f1685c7 Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac by kjellander · 8 years ago
  80. e74eef1 Add CreateSend/ReceiveTransport() methods to CallTest. by stefan · 8 years ago
  81. 37ebcf0 Reland "Add APK targets to build libjingle tests for Android." by phoglund · 8 years ago
  82. b71b4f0 Update attributes to match gclibc's ansidecl.h by kjellander · 8 years ago
  83. 004851c Roll chromium_revision 32569c6..ede5d4f (368258:368310) by kjellander · 8 years ago
  84. e1ca167 Add tracing to NetEqImpl::GetAudio by henrik.lundin · 8 years ago
  85. ec80f03 Check the mic volume only periodically on Mac. by andrew · 8 years ago
  86. fbeb97e Fix clang warning in peerconnection_jni.cc by perkj · 8 years ago
  87. 59bac1a Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest. by asapersson · 8 years ago
  88. 95ab30c Roll chromium_revision 6dd04c2..32569c6 (368115:368258) by kjellander · 8 years ago
  89. a2b1e03 Disable AudioDeviceAPITest.MicrophoneVolumeTests on Linux. by kjellander · 8 years ago
  90. 893505d Adding unit test to ensure TURN server priorities are unique. by Taylor Brandstetter · 8 years ago
  91. e5ba13b Adding a way for a Java RtpSender to set a track without taking ownership. by Taylor Brandstetter · 8 years ago
  92. ced8ec9 Roll chromium_revision bd5949f..6dd04c2 (368055:368115) by kjellander · 8 years ago
  93. bedc17b Fixing integer underflow in FileAudioDevice (webrtc issue 4554) by A.Brauckmann · 8 years ago
  94. 6938793 vp9 tests: Adjust some parameters and re-enable the tests. by Marco · 8 years ago
  95. 6f5ca08 Update API for Objective-C RTCMediaConstraints. by hjon · 8 years ago
  96. 9fea80f Add audio streams to CallTest and a first A/V call test. by Stefan Holmer · 8 years ago
  97. ecd21b4 Add ImplementationName to SimulcastEncoderAdapter. by pbos · 8 years ago
  98. 01f364e Remove always-on options in OveruseFrameDetector. by Peter Boström · 8 years ago
  99. 30166cb iOS stability improvement for device switching, including BT devices by henrika · 8 years ago
  100. 7776e78 Remove unused methods in VideoCodingModule. by Peter Boström · 8 years ago