Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 75db8612588b4fabdf1b05f4ab145f7737093b45 / talk
  1. 75db861 Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection. by Per · 9 years ago
  2. e095148 Port some fixes in AppRTCDemo. by Alex Glaznev · 9 years ago
  3. ef88309 Cleanup: Forward declare AudioFrame type in voiceprocess.h by Thiago Farina · 9 years ago
  4. 3354419 Zero copy AndroidVideeCapturer. by Per · 9 years ago
  5. 037bad7 ~CaptureManager: DCHECK(capture_states_.empty()) instead of CHECK until we fix not empty bug. by Henrik Boström · 9 years ago
  6. cb76b89 Cleanup: Move json.h into rtc namespace. by Thiago Farina · 9 years ago
  7. 64c1e8c Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 9 years ago
  8. 722ef1f Remove henrike@ from OWNERS by Henrik Kjellander · 9 years ago
  9. 31331cf Revert "Enable CVO by default through webrtc pipeline." by Minyue · 9 years ago
  10. 1b1c15c Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 9 years ago
  11. 4b3c0d6 Use WebRTC API to convert byteorder in srtpfilter. by Jiayang Liu · 9 years ago
  12. 4825356 RTCDataChannel: Unregister data channel observer on dealloc. by Zeke Chin · 9 years ago
  13. 379069f VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const. by Magnus Jedvert · 9 years ago
  14. 23914fe Reject RTP one-byte extension ID 0. by Peter Boström · 9 years ago
  15. 0194d32 Add WebRtcAudioManager to peerconnection_jar library by Alex Glaznev · 9 years ago
  16. 1ecfd55 videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)' by Magnus Jedvert · 9 years ago
  17. dfd53fe Raise streams for SetMaxSendBitrates above 2000k. by Peter Boström · 9 years ago
  18. 53eda3d Add tests for r8811. by Peter Boström · 9 years ago
  19. 75a0255 Handle borked Android cameras gracefully. by Per · 9 years ago
  20. 8ed6a4b Remove unused non-standard capture stats. by Peter Boström · 9 years ago
  21. 3954e1d Remove unused implementations in cricket::VideoFrame by Magnus Jedvert · 9 years ago
  22. 7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 9 years ago
  23. 74d9ed7 Report send codec name in GetStats(). by Peter Boström · 9 years ago
  24. d6f4c25 Reject streams reusing simulcast or RTX SSRCs. by Peter Boström · 9 years ago
  25. e590416 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 9 years ago
  26. 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 · 9 years ago
  27. bf395c1 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android by Bjorn Volcker · 9 years ago
  28. caae5d4 Bye request should use POST not GET by Chuck Hays · 9 years ago
  29. d4362cd Reject StreamParams with RTX SSRCs not in ssrcs. by Peter Boström · 9 years ago
  30. 0e209b0 Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/. by Donald Curtis · 9 years ago
  31. e61c64d Delete NullVideoRenderer by Magnus Jedvert · 9 years ago
  32. 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 · 9 years ago
  33. ac27e20 Delete VideoAdapter::AdaptFrame by Magnus Jedvert · 9 years ago
  34. eebcab5 rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 9 years ago
  35. e815290 Update README instructions for Android AppRTCDemo. by glaznev@webrtc.org · 9 years ago
  36. a5f6fb5 Permit single-stream max bitrates above 2000k. by pbos@webrtc.org · 9 years ago
  37. a197a5e Update libsrtp includes in preparation of roll into Chromium. by jiayl@webrtc.org · 9 years ago
  38. 39fc1d3 Disable PeerConnectionClientTest.testLoopbackVp9 by henrik.lundin@webrtc.org · 9 years ago
  39. 0b44b58 Limit disabling of PeerConnectionEndToEndTest.Call to Windows by henrik.lundin@webrtc.org · 9 years ago
  40. 64eb2ff iOS library build script by tkchin@webrtc.org · 9 years ago
  41. 82e8ae4 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest by henrik.lundin@webrtc.org · 9 years ago
  42. e5e92bd Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix) by kjellander@webrtc.org · 9 years ago
  43. cfde27e Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows. by kjellander@webrtc.org · 9 years ago
  44. b789f62 Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..." by tommi@webrtc.org · 9 years ago
  45. 0c34001 Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." by tommi@webrtc.org · 9 years ago
  46. 4ddc938 Support VP8 hardware encoding and decoding on IA devices. by glaznev@webrtc.org · 9 years ago
  47. b9557a9 Fix code to handle crashes for non-VP8. by pbos@webrtc.org · 9 years ago
  48. 66df3cf Set WebRtcVideoEngine2 as the WebRtcMediaEngine. by pbos@webrtc.org · 9 years ago
  49. 8296ec5 Fix heap-use-after-free in WebRtcVideoEngine2. by pbos@webrtc.org · 9 years ago
  50. 9f9ea7e Clean up webrtc external capture. by perkj@webrtc.org · 9 years ago
  51. 0c26299 Disabling two flaky tests in libjingle_media_unittest. by tina.legrand@webrtc.org · 9 years ago
  52. 8cc47e9 Objective-C readability review. by tkchin@webrtc.org · 9 years ago
  53. 840da7b Implement Rotation in Android Renderer. by guoweis@webrtc.org · 9 years ago
  54. 143451d Base start bitrate on last observed bitrate. by pbos@webrtc.org · 9 years ago
  55. af612d5 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" by perkj@webrtc.org · 9 years ago
  56. 14ee8cc WebRtcVideoFrame: Support odd resolutions by magjed@webrtc.org · 9 years ago
  57. 3fffd66 Revert "Implement Rotation in Android Renderer." by guoweis@webrtc.org · 9 years ago
  58. 835ec63 Implement Rotation in Android Renderer. by guoweis@webrtc.org · 9 years ago
  59. 52cd828 Allow webrtc external encoder factories to declare encoders have internal camera sources. by pthatcher@webrtc.org · 9 years ago
  60. 2161234 Add new features to AppRTCDemo from private repo. by glaznev@webrtc.org · 9 years ago
  61. a78a94e Fix RateTracker to set an initial reference time when first updated. by perkj@webrtc.org · 9 years ago
  62. ae222b5 Remove dead code in WebRtcVideoEngine2 unittests. by pbos@webrtc.org · 9 years ago
  63. 858024f WebRtcVideoFrame: Initialize members in empty constructor by magjed@webrtc.org · 9 years ago
  64. 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 · 9 years ago
  65. 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 · 9 years ago
  66. 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 · 9 years ago
  67. c04a97f Move from BaseSession::GetStats to WebRtcSession::GetTransportStats by pthatcher@webrtc.org · 9 years ago
  68. 3f11823 Disables SW AEC when built-in AEC is enabled by bjornv@webrtc.org · 9 years ago
  69. 2056ee3 Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." by magjed@webrtc.org · 9 years ago
  70. 93d9d65 I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. by hbos@webrtc.org · 9 years ago
  71. 2dc5fa6 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*. by hbos@webrtc.org · 9 years ago
  72. 4b89aa0 Change StatsCollector to use DCHECK instead of ASSERT. by tommi@webrtc.org · 9 years ago
  73. eed2fca Roll chromium_revision 00e438c..8d51d96 (320241:320682) by kjellander@webrtc.org · 9 years ago
  74. 2d25b44 Check associated payload type when negotiate RTX codecs. by changbin.shao@webrtc.org · 9 years ago
  75. c29f7f3 Disable assert for nr of threads in PeerConnectionTest.java. by tommi@webrtc.org · 9 years ago
  76. f1f558c Fix AppRTCDemo and AppRTCDemoTest builds. by glaznev@webrtc.org · 9 years ago
  77. d83f4ef Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. by jiayl@webrtc.org · 9 years ago
  78. b01c707 Use a NULL session in unit tests that don't actually use the session. by pthatcher@webrtc.org · 9 years ago
  79. b4aac13 Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well. by pthatcher@webrtc.org · 9 years ago
  80. 990a00c Remove unused transport code. by pthatcher@webrtc.org · 9 years ago
  81. 9b2e114 Supporting Opus DTX in Voice Engine. by minyue@webrtc.org · 9 years ago
  82. 503a9e8 Make AppRTCDemoTest pass without Internet connection. by kjellander@webrtc.org · 9 years ago
  83. 8372888 Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns." by jiayl@webrtc.org · 9 years ago
  84. 3d3c005 Fix Android peer connection client instrumentation tests. by glaznev@webrtc.org · 9 years ago
  85. fde1de9 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. by jiayl@webrtc.org · 9 years ago
  86. 00c509a Add concept of whether video renderer supports rotation. by guoweis@webrtc.org · 9 years ago
  87. 04cd698 Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns." by jiayl@webrtc.org · 9 years ago
  88. fdd1057 Add CVO support to Vie layer. by guoweis@webrtc.org · 9 years ago
  89. 4f85288 Socket options are only applied when first setting TransportChannelImpl. by guoweis@webrtc.org · 9 years ago
  90. 93604da Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. by jiayl@webrtc.org · 9 years ago
  91. d390029 Use a variant for storing stats values in StatsCollector code. by tommi@webrtc.org · 9 years ago
  92. 75b7f17 Temporary interface change to StatsReport::Id. by tommi@webrtc.org · 9 years ago
  93. afdd5dd Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"" by magjed@webrtc.org · 9 years ago
  94. e9413c6 Revert 8689 "Fix an issue in DtlsIdentityStore when the store is..." by mflodman@webrtc.org · 9 years ago
  95. 2a3942a Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. by jiayl@webrtc.org · 9 years ago
  96. 8c5ea8a Fix temporal layer log string. by decurtis@webrtc.org · 9 years ago
  97. ae1a078 Convert AppRTCDemo and AppRTCDemoTest to proper GYP target. by glaznev@webrtc.org · 9 years ago
  98. b218ff5 Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame" by magjed@webrtc.org · 9 years ago
  99. 370a72c Remove frame copy from cricket::VideoFrame to I420VideoFrame by magjed@webrtc.org · 9 years ago
  100. e77c9c8 Build WebRtcMediaEngine2 outside of Chromium. by pbos@webrtc.org · 9 years ago