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webrtc / src / 379593792082a86f389df9b1b790cc0fe9eb9975 / talk
  1. f49dbfa Close all camera resources when camera error happens. by Alex Glaznev · 10 years ago
  2. 9829af4 Disable VP8 encoder HW acceleration on NVidia devices. by Alex Glaznev · 10 years ago
  3. 352b2d7 Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream). by Åsa Persson · 10 years ago
  4. 4b76c02 Roll chromium_revision 8af41b3..dcb0929 (324854:325030) by Magnus Jedvert · 10 years ago
  5. 3c3f646 Prevent null-stream reconfigs on RTP extensions. by Peter Boström · 10 years ago
  6. e432800 Enable CPU adaptation by default. by Peter Boström · 10 years ago
  7. 56d5028 Remove SignalCaptureStateChange from MediaEngine. by Peter Thatcher · 10 years ago
  8. 575a802 Add an option to update mirror flag in Android video renderer. by Alex Glaznev · 10 years ago
  9. 1b67795 Add i386 to ios fat library build script and use boringssl. by Zeke Chin · 10 years ago
  10. 77f0e3f Remove GetStartCaptureFormat and some related code. by Peter Thatcher · 10 years ago
  11. e7b221f Remove deadlock in WebRtcVideoEngine2. by Peter Boström · 10 years ago
  12. eba964f Revert "Support none multiple of 16 pixels width on android." by Bjorn Volcker · 10 years ago
  13. 99c2fe5 Fix NullVideoEngine's CreateChannel implementation. by Noah Richards · 10 years ago
  14. e4ae8d8 Changes in VideoCapturerAndroid. by Alex Glaznev · 10 years ago
  15. f4acf46 Support none multiple of 16 pixels width on android. by Per · 10 years ago
  16. a125d7d Changes default audio mode in AppRTCDemo to MODE_RINGTONE. by henrika · 10 years ago
  17. 9bfe3daf Cleanup: Remove i420_video_frame.h header. by Thiago Farina · 10 years ago
  18. f6c003e cricket::VideoFrameFactory: Handle if created frame is null by Magnus Jedvert · 10 years ago
  19. 0184057 VideoAdapterTest: Replace FileVideoCapturer with FakeVideoCapturer by Magnus Jedvert · 10 years ago
  20. 76c53d3 Remove ViE interface usage from VideoReceiveStream. by Peter Boström · 10 years ago
  21. 15cf019 Add field-trial flag to disable WebRtcVideoEngine2. by Peter Boström · 10 years ago
  22. 9b3f56e Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection."" by Per · 10 years ago
  23. d61ebda9 Fix the sigslot type of DtlsIdentityStore::WorkerTask. by Jiayang Liu · 10 years ago
  24. ad1f9b6 Remove warning on input frames before config. by Peter Boström · 10 years ago
  25. 9e420af Fix potential race conditions in Android video renderer. by Alex Glaznev · 10 years ago
  26. e41d774 Revert "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection." by Per · 10 years ago
  27. 1d83f1e talk/media/webrtc/webrtcvoiceengine: Delay Agnostic AEC should not override HW-AEC by Bjorn Volcker · 10 years ago
  28. 49a862e Return pending buffers to Java VideoCapturerAndroid if camera is stopping by Per · 10 years ago
  29. 75db861 Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection. by Per · 10 years ago
  30. e095148 Port some fixes in AppRTCDemo. by Alex Glaznev · 10 years ago
  31. ef88309 Cleanup: Forward declare AudioFrame type in voiceprocess.h by Thiago Farina · 10 years ago
  32. 3354419 Zero copy AndroidVideeCapturer. by Per · 10 years ago
  33. 037bad7 ~CaptureManager: DCHECK(capture_states_.empty()) instead of CHECK until we fix not empty bug. by Henrik Boström · 10 years ago
  34. cb76b89 Cleanup: Move json.h into rtc namespace. by Thiago Farina · 10 years ago
  35. 64c1e8c Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 10 years ago
  36. 722ef1f Remove henrike@ from OWNERS by Henrik Kjellander · 10 years ago
  37. 31331cf Revert "Enable CVO by default through webrtc pipeline." by Minyue · 10 years ago
  38. 1b1c15c Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 10 years ago
  39. 4b3c0d6 Use WebRTC API to convert byteorder in srtpfilter. by Jiayang Liu · 10 years ago
  40. 4825356 RTCDataChannel: Unregister data channel observer on dealloc. by Zeke Chin · 10 years ago
  41. 379069f VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const. by Magnus Jedvert · 10 years ago
  42. 23914fe Reject RTP one-byte extension ID 0. by Peter Boström · 10 years ago
  43. 0194d32 Add WebRtcAudioManager to peerconnection_jar library by Alex Glaznev · 10 years ago
  44. 1ecfd55 videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)' by Magnus Jedvert · 10 years ago
  45. dfd53fe Raise streams for SetMaxSendBitrates above 2000k. by Peter Boström · 10 years ago
  46. 53eda3d Add tests for r8811. by Peter Boström · 10 years ago
  47. 75a0255 Handle borked Android cameras gracefully. by Per · 10 years ago
  48. 8ed6a4b Remove unused non-standard capture stats. by Peter Boström · 10 years ago
  49. 3954e1d Remove unused implementations in cricket::VideoFrame by Magnus Jedvert · 10 years ago
  50. 7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 10 years ago
  51. 74d9ed7 Report send codec name in GetStats(). by Peter Boström · 10 years ago
  52. d6f4c25 Reject streams reusing simulcast or RTX SSRCs. by Peter Boström · 10 years ago
  53. e590416 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 10 years ago
  54. 5225dd8 If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size. by Brave Yao · 10 years ago
  55. bf395c1 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android by Bjorn Volcker · 10 years ago
  56. caae5d4 Bye request should use POST not GET by Chuck Hays · 10 years ago
  57. d4362cd Reject StreamParams with RTX SSRCs not in ssrcs. by Peter Boström · 10 years ago
  58. 0e209b0 Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/. by Donald Curtis · 10 years ago
  59. e61c64d Delete NullVideoRenderer by Magnus Jedvert · 10 years ago
  60. 07a4ba5 Simulcast settings for 1080p. Using same bit rates for all 3 modes since only one is used in reality, and the plan is to unify them. by Niklas Enbom · 10 years ago
  61. ac27e20 Delete VideoAdapter::AdaptFrame by Magnus Jedvert · 10 years ago
  62. eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 10 years ago
  63. e815290 Update README instructions for Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
  64. a5f6fb5 Permit single-stream max bitrates above 2000k. by pbos@webrtc.org · 10 years ago
  65. a197a5e Update libsrtp includes in preparation of roll into Chromium. by jiayl@webrtc.org · 10 years ago
  66. 39fc1d3 Disable PeerConnectionClientTest.testLoopbackVp9 by henrik.lundin@webrtc.org · 10 years ago
  67. 0b44b58 Limit disabling of PeerConnectionEndToEndTest.Call to Windows by henrik.lundin@webrtc.org · 10 years ago
  68. 64eb2ff iOS library build script by tkchin@webrtc.org · 10 years ago
  69. 82e8ae4 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest by henrik.lundin@webrtc.org · 10 years ago
  70. e5e92bd Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix) by kjellander@webrtc.org · 10 years ago
  71. cfde27e Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows. by kjellander@webrtc.org · 10 years ago
  72. b789f62 Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..." by tommi@webrtc.org · 10 years ago
  73. 0c34001 Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." by tommi@webrtc.org · 10 years ago
  74. 4ddc938 Support VP8 hardware encoding and decoding on IA devices. by glaznev@webrtc.org · 10 years ago
  75. b9557a9 Fix code to handle crashes for non-VP8. by pbos@webrtc.org · 10 years ago
  76. 66df3cf Set WebRtcVideoEngine2 as the WebRtcMediaEngine. by pbos@webrtc.org · 10 years ago
  77. 8296ec5 Fix heap-use-after-free in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  78. 9f9ea7e Clean up webrtc external capture. by perkj@webrtc.org · 10 years ago
  79. 0c26299 Disabling two flaky tests in libjingle_media_unittest. by tina.legrand@webrtc.org · 10 years ago
  80. 8cc47e9 Objective-C readability review. by tkchin@webrtc.org · 10 years ago
  81. 840da7b Implement Rotation in Android Renderer. by guoweis@webrtc.org · 10 years ago
  82. 143451d Base start bitrate on last observed bitrate. by pbos@webrtc.org · 10 years ago
  83. af612d5 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" by perkj@webrtc.org · 10 years ago
  84. 14ee8cc WebRtcVideoFrame: Support odd resolutions by magjed@webrtc.org · 10 years ago
  85. 3fffd66 Revert "Implement Rotation in Android Renderer." by guoweis@webrtc.org · 10 years ago
  86. 835ec63 Implement Rotation in Android Renderer. by guoweis@webrtc.org · 10 years ago
  87. 52cd828 Allow webrtc external encoder factories to declare encoders have internal camera sources. by pthatcher@webrtc.org · 10 years ago
  88. 2161234 Add new features to AppRTCDemo from private repo. by glaznev@webrtc.org · 10 years ago
  89. a78a94e Fix RateTracker to set an initial reference time when first updated. by perkj@webrtc.org · 10 years ago
  90. ae222b5 Remove dead code in WebRtcVideoEngine2 unittests. by pbos@webrtc.org · 10 years ago
  91. 858024f WebRtcVideoFrame: Initialize members in empty constructor by magjed@webrtc.org · 10 years ago
  92. 592470b Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession. by pthatcher@webrtc.org · 10 years ago
  93. 6ad507a Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE. by pthatcher@webrtc.org · 10 years ago
  94. 4eeef58 Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession. by pthatcher@webrtc.org · 10 years ago
  95. c04a97f Move from BaseSession::GetStats to WebRtcSession::GetTransportStats by pthatcher@webrtc.org · 10 years ago
  96. 3f11823 Disables SW AEC when built-in AEC is enabled by bjornv@webrtc.org · 10 years ago
  97. 2056ee3 Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." by magjed@webrtc.org · 10 years ago
  98. 93d9d65 I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. by hbos@webrtc.org · 10 years ago
  99. 2dc5fa6 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*. by hbos@webrtc.org · 10 years ago
  100. 4b89aa0 Change StatsCollector to use DCHECK instead of ASSERT. by tommi@webrtc.org · 10 years ago