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gerrit-public.fairphone.software / platform / external / webrtc / aa0bbab8ec359f50dc7d6a468d3a8bac01b39fc8
  1. aa0bbab Fix build failure by jmarusic@webrtc.org · 10 years ago
  2. a4bef3e AcmReceiver: use std::map instead of an array to keep the list of decoders by jmarusic@webrtc.org · 10 years ago
  3. 3335a4f Prevent asserting on unset start bitrate. by pbos@webrtc.org · 10 years ago
  4. 50ed0d9 Roll chromium_revision 6311617..da9a1c0 (321517:321718) by kjellander@webrtc.org · 10 years ago
  5. e5e92bd Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix) by kjellander@webrtc.org · 10 years ago
  6. cfde27e Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows. by kjellander@webrtc.org · 10 years ago
  7. 38492c5 Re-land 8810 "- Add a SetPriority method to ThreadWr..." by tommi@webrtc.org · 10 years ago
  8. 90a1cb4 Revert 8810 "- Add a SetPriority method to ThreadWrapper" by tommi@webrtc.org · 10 years ago
  9. b789f62 Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..." by tommi@webrtc.org · 10 years ago
  10. 0c34001 Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." by tommi@webrtc.org · 10 years ago
  11. 346a64b Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default. by braveyao@webrtc.org · 10 years ago
  12. 4553941 Document the 'int' return value of Resampler methods. by wtc@chromium.org · 10 years ago
  13. 3200a64 Minor fix for MIPS Android build. by andrew@webrtc.org · 10 years ago
  14. 4ddc938 Support VP8 hardware encoding and decoding on IA devices. by glaznev@webrtc.org · 10 years ago
  15. b9557a9 Fix code to handle crashes for non-VP8. by pbos@webrtc.org · 10 years ago
  16. b6817d7 - Add a SetPriority method to ThreadWrapper by tommi@webrtc.org · 10 years ago
  17. 66df3cf Set WebRtcVideoEngine2 as the WebRtcMediaEngine. by pbos@webrtc.org · 10 years ago
  18. 8296ec5 Fix heap-use-after-free in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  19. a3209a2 Release buffer pool in Vp8DecoderImpl::Release(). by pbos@webrtc.org · 10 years ago
  20. 8904290 Make screenshare target bitrate experiment always on by pbos@webrtc.org · 10 years ago
  21. d9c5024 Roll chromium_revision bd49b12..6311617 (320783:321517) by kjellander@webrtc.org · 10 years ago
  22. 9f9ea7e Clean up webrtc external capture. by perkj@webrtc.org · 10 years ago
  23. 443ad40 Remove FullStackTest frame pointer handles. by pbos@webrtc.org · 10 years ago
  24. 6231fb6 Prevent crashes when copying a zero-size frame. by pbos@webrtc.org · 10 years ago
  25. 6069032 Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  26. 4ab23d0 Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  27. bd8c865 Remove build-time beamformer flags. by andrew@webrtc.org · 10 years ago
  28. 04c5098 Add the Ooura FFT to RealFourier. by andrew@webrtc.org · 10 years ago
  29. ba86031 Whitespace change to trigger new Git pollers (2). by kjellander@webrtc.org · 10 years ago
  30. cf3fb9b Whitespace change to trigger new Git pollers. by kjellander@webrtc.org · 10 years ago
  31. 80d9aee Adds full-duplex unit test to AudioDeviceTest on Android by henrika@webrtc.org · 10 years ago
  32. 361981f Use scoped_ptr for ThreadWrapper::CreateThread. by tommi@webrtc.org · 10 years ago
  33. c7d5a73 Disable flaky test on DrMemory bots by tina.legrand@webrtc.org · 10 years ago
  34. 27c0be9 Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper. by tommi@webrtc.org · 10 years ago
  35. 0c26299 Disabling two flaky tests in libjingle_media_unittest. by tina.legrand@webrtc.org · 10 years ago
  36. 17c64d1 Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame" by magjed@webrtc.org · 10 years ago
  37. c7157da Use atomic operations for setting/reading the trace filter. by tommi@webrtc.org · 10 years ago
  38. 9afaee7 Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal() by jmarusic@webrtc.org · 10 years ago
  39. d21406d Remove command-line tool 'video_coding_test'. by pbos@webrtc.org · 10 years ago
  40. c4709a2 Split C++ class from macro overrides to fix Chromium build by tommi@webrtc.org · 10 years ago
  41. 5506a93 Expose ViECaptureImpl::DisconnectCaptureDevice() to JNI of WebRTCDemo and call it before releasing camera to deregister the corresponding framecallback. Also stop camera after stop remote rendering as the correct termination order. by braveyao@webrtc.org · 10 years ago
  42. 8cc47e9 Objective-C readability review. by tkchin@webrtc.org · 10 years ago
  43. 2a8a46d vp8: Add missing call to SetUsageMessage(). by kjellander@webrtc.org · 10 years ago
  44. 8f76cd2 Renaming neteq_opus_fec_quality_test. by minyue@webrtc.org · 10 years ago
  45. 840da7b Implement Rotation in Android Renderer. by guoweis@webrtc.org · 10 years ago
  46. 143451d Base start bitrate on last observed bitrate. by pbos@webrtc.org · 10 years ago
  47. 5a477a0 DCHECK frame parameters instead of return codes. by pbos@webrtc.org · 10 years ago
  48. 4346d92 Use SendTimeHistory to keep track of send times in simulations. by stefan@webrtc.org · 10 years ago
  49. f189933 Removing henrik.lundin from OWNERS in video_coding/* by henrik.lundin@webrtc.org · 10 years ago
  50. af612d5 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" by perkj@webrtc.org · 10 years ago
  51. 6dba1eb Make AudioDecoder stateless by henrik.lundin@webrtc.org · 10 years ago
  52. 14ee8cc WebRtcVideoFrame: Support odd resolutions by magjed@webrtc.org · 10 years ago
  53. fc562e0 Delete ACMGenericCodec::Encode and use AudioEncoder::Encode directly by henrik.lundin@webrtc.org · 10 years ago
  54. 019955d Revert 8749 "We changed Encode() and EncodeInternal() return typ..." by tommi@webrtc.org · 10 years ago
  55. 3fffd66 Revert "Implement Rotation in Android Renderer." by guoweis@webrtc.org · 10 years ago
  56. 835ec63 Implement Rotation in Android Renderer. by guoweis@webrtc.org · 10 years ago
  57. 52cd828 Allow webrtc external encoder factories to declare encoders have internal camera sources. by pthatcher@webrtc.org · 10 years ago
  58. edd517b Fix FYI build - add a missing include to event_tracer.h in system_wrappers. by tommi@webrtc.org · 10 years ago
  59. 54d072e Add CVO support to video_coding layer. by guoweis@webrtc.org · 10 years ago
  60. 63a1097 Remove troublesome Windows line ending. by pthatcher@webrtc.org · 10 years ago
  61. 462dbcf Fix bug in Transport where channel_.clear() was being called without a lock. by tommi@webrtc.org · 10 years ago
  62. b493cb4 Add storage alignment fix for opengles2.0 for iOS by tkchin@webrtc.org · 10 years ago
  63. da4fcc4 Add minor fixes to video_capture_ios.mm in order to make it more robust. by tkchin@webrtc.org · 10 years ago
  64. 2161234 Add new features to AppRTCDemo from private repo. by glaznev@webrtc.org · 10 years ago
  65. 779c3d1 Use ByteReader/ByteWriter instead of rtputility and manual shift/add. by sprang@webrtc.org · 10 years ago
  66. 09098da Fix screenshare loopback target bitrate which isn't correctly configured by sprang@webrtc.org · 10 years ago
  67. 25819b8 Revert 8753 "Use atomic operations for setting/reading the trace..." by tommi@webrtc.org · 10 years ago
  68. b91d0f5 1. Have IPIsPrivate calling IPIsLinkLocal by guoweis@webrtc.org · 10 years ago
  69. 3093390 Parsing of transport wide sequence number rtp extension header. by sprang@webrtc.org · 10 years ago
  70. 1e69252 Write commit position as a comment in Chromium DEPS. by kjellander@webrtc.org · 10 years ago
  71. 7c64ed2 Move trace_event and associated files to webrtc/base. by tommi@webrtc.org · 10 years ago
  72. 7c112f3 Adding build_opus as a switch in GYP. by minyue@webrtc.org · 10 years ago
  73. c383c24 Use atomic operations for setting/reading the trace filter. by tommi@webrtc.org · 10 years ago
  74. a846371 Modify EventPosix to prevent spurious wakeups. by pbos@webrtc.org · 10 years ago
  75. a78a94e Fix RateTracker to set an initial reference time when first updated. by perkj@webrtc.org · 10 years ago
  76. e155dbe VP8/9EncoderImpl::Encode: Check resolution of input I420VideoFrame by magjed@webrtc.org · 10 years ago
  77. 0cb612b We changed Encode() and EncodeInternal() return type from bool to void in this issue: by jmarusic@webrtc.org · 10 years ago
  78. 73d763e Add I420 buffer pool to avoid unnecessary allocations by magjed@webrtc.org · 10 years ago
  79. ae222b5 Remove dead code in WebRtcVideoEngine2 unittests. by pbos@webrtc.org · 10 years ago
  80. 858024f WebRtcVideoFrame: Initialize members in empty constructor by magjed@webrtc.org · 10 years ago
  81. 646eeac Roll chromium_revision 8d51d96..bd49b12 (320682:320783) by kjellander@webrtc.org · 10 years ago
  82. 06d9390 Adjust a threshold in VP9 test. by marpan@webrtc.org · 10 years ago
  83. 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
  84. 12e7951 Remove libvpx suppression due to fixed bug. by kjellander@webrtc.org · 10 years ago
  85. 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
  86. 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
  87. c04a97f Move from BaseSession::GetStats to WebRtcSession::GetTransportStats by pthatcher@webrtc.org · 10 years ago
  88. aba9219 Change ThreadPosix to use an auto-reset event instead of manual reset now that we know the problem we had with EventWrapper::Wait was simply a bug in the EventWrapper. Also removing |started_| since we can just check the thread_ instead. by tommi@webrtc.org · 10 years ago
  89. 02d166b Fixing a race condition in ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  90. 3f11823 Disables SW AEC when built-in AEC is enabled by bjornv@webrtc.org · 10 years ago
  91. 8bd2f40 Remove code related to REMB suppressor experiment. by sprang@webrtc.org · 10 years ago
  92. 2056ee3 Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." by magjed@webrtc.org · 10 years ago
  93. 93d9d65 I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. by hbos@webrtc.org · 10 years ago
  94. 2dc5fa6 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*. by hbos@webrtc.org · 10 years ago
  95. 7f7d7e3 Prevent crash in NetEQ when decoder overflow. by minyue@webrtc.org · 10 years ago
  96. 4b89aa0 Change StatsCollector to use DCHECK instead of ASSERT. by tommi@webrtc.org · 10 years ago
  97. eed2fca Roll chromium_revision 00e438c..8d51d96 (320241:320682) by kjellander@webrtc.org · 10 years ago
  98. 2d25b44 Check associated payload type when negotiate RTX codecs. by changbin.shao@webrtc.org · 10 years ago
  99. eb44fd6 Add flag to always close previous roll + minor refactor by kjellander@webrtc.org · 10 years ago
  100. c29f7f3 Disable assert for nr of threads in PeerConnectionTest.java. by tommi@webrtc.org · 10 years ago