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gerrit-public.fairphone.software / platform / external / webrtc / b9557a9bb7ed5f9aa1e7b3a64de4238572794ae3
  1. b9557a9 Fix code to handle crashes for non-VP8. by pbos@webrtc.org · 10 years ago
  2. b6817d7 - Add a SetPriority method to ThreadWrapper by tommi@webrtc.org · 10 years ago
  3. 66df3cf Set WebRtcVideoEngine2 as the WebRtcMediaEngine. by pbos@webrtc.org · 10 years ago
  4. 8296ec5 Fix heap-use-after-free in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  5. a3209a2 Release buffer pool in Vp8DecoderImpl::Release(). by pbos@webrtc.org · 10 years ago
  6. 8904290 Make screenshare target bitrate experiment always on by pbos@webrtc.org · 10 years ago
  7. d9c5024 Roll chromium_revision bd49b12..6311617 (320783:321517) by kjellander@webrtc.org · 10 years ago
  8. 9f9ea7e Clean up webrtc external capture. by perkj@webrtc.org · 10 years ago
  9. 443ad40 Remove FullStackTest frame pointer handles. by pbos@webrtc.org · 10 years ago
  10. 6231fb6 Prevent crashes when copying a zero-size frame. by pbos@webrtc.org · 10 years ago
  11. 6069032 Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  12. 4ab23d0 Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  13. bd8c865 Remove build-time beamformer flags. by andrew@webrtc.org · 10 years ago
  14. 04c5098 Add the Ooura FFT to RealFourier. by andrew@webrtc.org · 10 years ago
  15. ba86031 Whitespace change to trigger new Git pollers (2). by kjellander@webrtc.org · 10 years ago
  16. cf3fb9b Whitespace change to trigger new Git pollers. by kjellander@webrtc.org · 10 years ago
  17. 80d9aee Adds full-duplex unit test to AudioDeviceTest on Android by henrika@webrtc.org · 10 years ago
  18. 361981f Use scoped_ptr for ThreadWrapper::CreateThread. by tommi@webrtc.org · 10 years ago
  19. c7d5a73 Disable flaky test on DrMemory bots by tina.legrand@webrtc.org · 10 years ago
  20. 27c0be9 Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper. by tommi@webrtc.org · 10 years ago
  21. 0c26299 Disabling two flaky tests in libjingle_media_unittest. by tina.legrand@webrtc.org · 10 years ago
  22. 17c64d1 Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame" by magjed@webrtc.org · 10 years ago
  23. c7157da Use atomic operations for setting/reading the trace filter. by tommi@webrtc.org · 10 years ago
  24. 9afaee7 Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal() by jmarusic@webrtc.org · 10 years ago
  25. d21406d Remove command-line tool 'video_coding_test'. by pbos@webrtc.org · 10 years ago
  26. c4709a2 Split C++ class from macro overrides to fix Chromium build by tommi@webrtc.org · 10 years ago
  27. 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
  28. 8cc47e9 Objective-C readability review. by tkchin@webrtc.org · 10 years ago
  29. 2a8a46d vp8: Add missing call to SetUsageMessage(). by kjellander@webrtc.org · 10 years ago
  30. 8f76cd2 Renaming neteq_opus_fec_quality_test. by minyue@webrtc.org · 10 years ago
  31. 840da7b Implement Rotation in Android Renderer. by guoweis@webrtc.org · 10 years ago
  32. 143451d Base start bitrate on last observed bitrate. by pbos@webrtc.org · 10 years ago
  33. 5a477a0 DCHECK frame parameters instead of return codes. by pbos@webrtc.org · 10 years ago
  34. 4346d92 Use SendTimeHistory to keep track of send times in simulations. by stefan@webrtc.org · 10 years ago
  35. f189933 Removing henrik.lundin from OWNERS in video_coding/* by henrik.lundin@webrtc.org · 10 years ago
  36. af612d5 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" by perkj@webrtc.org · 10 years ago
  37. 6dba1eb Make AudioDecoder stateless by henrik.lundin@webrtc.org · 10 years ago
  38. 14ee8cc WebRtcVideoFrame: Support odd resolutions by magjed@webrtc.org · 10 years ago
  39. fc562e0 Delete ACMGenericCodec::Encode and use AudioEncoder::Encode directly by henrik.lundin@webrtc.org · 10 years ago
  40. 019955d Revert 8749 "We changed Encode() and EncodeInternal() return typ..." by tommi@webrtc.org · 10 years ago
  41. 3fffd66 Revert "Implement Rotation in Android Renderer." by guoweis@webrtc.org · 10 years ago
  42. 835ec63 Implement Rotation in Android Renderer. by guoweis@webrtc.org · 10 years ago
  43. 52cd828 Allow webrtc external encoder factories to declare encoders have internal camera sources. by pthatcher@webrtc.org · 10 years ago
  44. edd517b Fix FYI build - add a missing include to event_tracer.h in system_wrappers. by tommi@webrtc.org · 10 years ago
  45. 54d072e Add CVO support to video_coding layer. by guoweis@webrtc.org · 10 years ago
  46. 63a1097 Remove troublesome Windows line ending. by pthatcher@webrtc.org · 10 years ago
  47. 462dbcf Fix bug in Transport where channel_.clear() was being called without a lock. by tommi@webrtc.org · 10 years ago
  48. b493cb4 Add storage alignment fix for opengles2.0 for iOS by tkchin@webrtc.org · 10 years ago
  49. da4fcc4 Add minor fixes to video_capture_ios.mm in order to make it more robust. by tkchin@webrtc.org · 10 years ago
  50. 2161234 Add new features to AppRTCDemo from private repo. by glaznev@webrtc.org · 10 years ago
  51. 779c3d1 Use ByteReader/ByteWriter instead of rtputility and manual shift/add. by sprang@webrtc.org · 10 years ago
  52. 09098da Fix screenshare loopback target bitrate which isn't correctly configured by sprang@webrtc.org · 10 years ago
  53. 25819b8 Revert 8753 "Use atomic operations for setting/reading the trace..." by tommi@webrtc.org · 10 years ago
  54. b91d0f5 1. Have IPIsPrivate calling IPIsLinkLocal by guoweis@webrtc.org · 10 years ago
  55. 3093390 Parsing of transport wide sequence number rtp extension header. by sprang@webrtc.org · 10 years ago
  56. 1e69252 Write commit position as a comment in Chromium DEPS. by kjellander@webrtc.org · 10 years ago
  57. 7c64ed2 Move trace_event and associated files to webrtc/base. by tommi@webrtc.org · 10 years ago
  58. 7c112f3 Adding build_opus as a switch in GYP. by minyue@webrtc.org · 10 years ago
  59. c383c24 Use atomic operations for setting/reading the trace filter. by tommi@webrtc.org · 10 years ago
  60. a846371 Modify EventPosix to prevent spurious wakeups. by pbos@webrtc.org · 10 years ago
  61. a78a94e Fix RateTracker to set an initial reference time when first updated. by perkj@webrtc.org · 10 years ago
  62. e155dbe VP8/9EncoderImpl::Encode: Check resolution of input I420VideoFrame by magjed@webrtc.org · 10 years ago
  63. 0cb612b We changed Encode() and EncodeInternal() return type from bool to void in this issue: by jmarusic@webrtc.org · 10 years ago
  64. 73d763e Add I420 buffer pool to avoid unnecessary allocations by magjed@webrtc.org · 10 years ago
  65. ae222b5 Remove dead code in WebRtcVideoEngine2 unittests. by pbos@webrtc.org · 10 years ago
  66. 858024f WebRtcVideoFrame: Initialize members in empty constructor by magjed@webrtc.org · 10 years ago
  67. 646eeac Roll chromium_revision 8d51d96..bd49b12 (320682:320783) by kjellander@webrtc.org · 10 years ago
  68. 06d9390 Adjust a threshold in VP9 test. by marpan@webrtc.org · 10 years ago
  69. 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
  70. 12e7951 Remove libvpx suppression due to fixed bug. by kjellander@webrtc.org · 10 years ago
  71. 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
  72. 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
  73. c04a97f Move from BaseSession::GetStats to WebRtcSession::GetTransportStats by pthatcher@webrtc.org · 10 years ago
  74. 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
  75. 02d166b Fixing a race condition in ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  76. 3f11823 Disables SW AEC when built-in AEC is enabled by bjornv@webrtc.org · 10 years ago
  77. 8bd2f40 Remove code related to REMB suppressor experiment. by sprang@webrtc.org · 10 years ago
  78. 2056ee3 Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." by magjed@webrtc.org · 10 years ago
  79. 93d9d65 I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. by hbos@webrtc.org · 10 years ago
  80. 2dc5fa6 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*. by hbos@webrtc.org · 10 years ago
  81. 7f7d7e3 Prevent crash in NetEQ when decoder overflow. by minyue@webrtc.org · 10 years ago
  82. 4b89aa0 Change StatsCollector to use DCHECK instead of ASSERT. by tommi@webrtc.org · 10 years ago
  83. eed2fca Roll chromium_revision 00e438c..8d51d96 (320241:320682) by kjellander@webrtc.org · 10 years ago
  84. 2d25b44 Check associated payload type when negotiate RTX codecs. by changbin.shao@webrtc.org · 10 years ago
  85. eb44fd6 Add flag to always close previous roll + minor refactor by kjellander@webrtc.org · 10 years ago
  86. c29f7f3 Disable assert for nr of threads in PeerConnectionTest.java. by tommi@webrtc.org · 10 years ago
  87. 6107ba1 Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame by magjed@webrtc.org · 10 years ago
  88. f1f558c Fix AppRTCDemo and AppRTCDemoTest builds. by glaznev@webrtc.org · 10 years ago
  89. d83f4ef Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. by jiayl@webrtc.org · 10 years ago
  90. b01c707 Use a NULL session in unit tests that don't actually use the session. by pthatcher@webrtc.org · 10 years ago
  91. 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 · 10 years ago
  92. 990a00c Remove unused transport code. by pthatcher@webrtc.org · 10 years ago
  93. c449c20 Flag to wait for trybots to complete. by kjellander@webrtc.org · 10 years ago
  94. bc2bb34 Refactor audio_coding/codecs/isac: Removed usage of macro WEBRTC_SPL_MUL_16_16 by bjornv@webrtc.org · 10 years ago
  95. 9b2e114 Supporting Opus DTX in Voice Engine. by minyue@webrtc.org · 10 years ago
  96. dd0292a Send to CQ by default and add --no-commit flag + cleanup. by kjellander@webrtc.org · 10 years ago
  97. 503a9e8 Make AppRTCDemoTest pass without Internet connection. by kjellander@webrtc.org · 10 years ago
  98. 0c5b137 Remove support for iSAC RCU by henrik.lundin@webrtc.org · 10 years ago
  99. 9f41810 Roll chromium_revision 87ce36b..00e438c (319600:320241) by kjellander@webrtc.org · 10 years ago
  100. 8372888 Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns." by jiayl@webrtc.org · 10 years ago