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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 4a9843fbbd17cf145b822734db9679130dcc4775
  1. 4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  2. 2622be1 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
  3. 58b912b Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
  4. e7270f5 Faster implementation of BitRateStats. by mikhal@webrtc.org · 11 years ago
  5. 1a5aa03 Updated WebRTC version to 3.47 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  6. 5892ce5 Made video quality toolchain more configurable. by phoglund@webrtc.org · 11 years ago
  7. c86d1c6 Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
  8. 586becf Add test for automatically disabling padding when no video is being captured. by stefan@webrtc.org · 11 years ago
  9. 5ae14be Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error. by fbarchard@google.com · 11 years ago
  10. e8f79c5 Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. by turaj@webrtc.org · 11 years ago
  11. 8bdb87f Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. by sergeyu@chromium.org · 11 years ago
  12. 5fd393f Fix issues with sequence number wrap-around in jitter statistics. by turaj@webrtc.org · 11 years ago
  13. 6508af1 Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out. by turaj@webrtc.org · 11 years ago
  14. 44b21e7 Replace VideoFrameI420 with I420VideoFrame. by pbos@webrtc.org · 11 years ago
  15. 51fa6ac Don't reset the AEC filter in extended mode. by andrew@webrtc.org · 11 years ago
  16. ce4a0b8 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. by dwkang@webrtc.org · 11 years ago
  17. 970c5e5 Renaming ViEEncoderObserver::VideoSuspended by henrik.lundin@webrtc.org · 11 years ago
  18. a706baf Protect reads of ViEEncoder::video_suspended_. by pbos@webrtc.org · 11 years ago
  19. 9c15a62 Increase size of pacer window to 500 ms as that better matches the encoder. by stefan@webrtc.org · 11 years ago
  20. d7d60c8 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  21. 0e6558b Lock access to ModuleRtpRtcpImpl::simulcast_. by pbos@webrtc.org · 11 years ago
  22. f8486d0 Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
  23. b87f528 Fix issues with sequence number wrap-around in jitter statistics by henrik.lundin@webrtc.org · 11 years ago
  24. 3c3a953 Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
  25. e92aec9 Make interface destructor virtual by sprang@webrtc.org · 11 years ago
  26. 3fe2e7f Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  27. 402f34c Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. by asapersson@webrtc.org · 11 years ago
  28. fa7ac56 Typo in vie_autotest_win.cc by braveyao@webrtc.org · 11 years ago
  29. 36fb531 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
  30. 13a4d31 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  31. b06a926 Added ViE API for getting overuse measure. by asapersson@webrtc.org · 11 years ago
  32. 04bcc9d Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
  33. c2162d1 Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
  34. f3b4602 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
  35. 60108c2 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
  36. 48cc9dc Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
  37. 162021c Rename Call::Create{Receive,Send}Stream(). by pbos@webrtc.org · 11 years ago
  38. 4bfa866 Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  39. 2b9794b Fix DesktopAndCursorComposer to restore frames to the original state. by sergeyu@chromium.org · 11 years ago
  40. dbc2a35 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. by turaj@webrtc.org · 11 years ago
  41. 8fdf191 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  42. 8d2354a Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
  43. 26a736f Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
  44. 5eca0c7 Fix breakage after introducing new test. by stefan@webrtc.org · 11 years ago
  45. f8c47a1 Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
  46. 90e2fdd Increment RTP timestamps for padding packets by henrik.lundin@webrtc.org · 11 years ago
  47. 8f2997c Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
  48. 764b28e Disable all vie_auto_tests on Linux for now (take 2) by kjellander@webrtc.org · 11 years ago
  49. d8dc0f5 Disable all automated vie_auto_tests on Linux for now by kjellander@webrtc.org · 11 years ago
  50. 04281a4 Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
  51. 8dda8d2 Inject config when creating channels to override the existing one. by turaj@webrtc.org · 11 years ago
  52. 7cba612 Reimplementing NetEq4's AudioVector by henrik.lundin@webrtc.org · 11 years ago
  53. 0db738b Parse next RTCP XR report block after an unsupported block type. by asapersson@webrtc.org · 11 years ago
  54. c824f2c Reducing opus_test runtime to pass Android test by minyue@webrtc.org · 11 years ago
  55. e5efa32 MIPS optimizations for AECM audio processing module by andrew@webrtc.org · 11 years ago
  56. 7821bd1 Move audio_processing dependencies to a variable. by andrew@webrtc.org · 11 years ago
  57. 590c60f Remove ".." from include_dirs in build/common. by pbos@webrtc.org · 11 years ago
  58. 01966bb Remove unnecessary include_dirs from audio_processing. by andrew@webrtc.org · 11 years ago
  59. 6dc6e03 Remove unneeded includes from trace_posix.cc. by andrew@webrtc.org · 11 years ago
  60. e9274ae Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
  61. 6106bbc Fix log build error for Chromium builds. by henrikg@webrtc.org · 11 years ago
  62. e0df4d7 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
  63. 0aa16d7 Replace disabled logging with a restricted logging mode. by andrew@webrtc.org · 11 years ago
  64. c4a7861 Updated WebRTC version to 3.46 by elham@webrtc.org · 11 years ago
  65. 6196a56 Fix for video_processor_intergration_tests to run in parallel. by marpan@webrtc.org · 11 years ago
  66. 685e91a Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
  67. 1dc0158 Sending status fix for module. by asapersson@webrtc.org · 11 years ago
  68. c359e28 Add missing dependencies to .isolate files by kjellander@webrtc.org · 11 years ago
  69. 4e0ea6a Fix broken build on x86 Android by fischman@webrtc.org · 11 years ago
  70. 65e4415 Removed unused code. by asapersson@webrtc.org · 11 years ago
  71. e3709a8 Make video quality analysis unittests print to log instead of stdout. by kjellander@webrtc.org · 11 years ago
  72. 06977ab Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  73. f5fdd0c Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  74. a4a5bf2 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  75. 987587e Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  76. 565f991 Address Clag Analyzer issues. by turaj@webrtc.org · 11 years ago
  77. f1262f3 Propagate estimated RTT from receivers to rtt observer. by asapersson@webrtc.org · 11 years ago
  78. 3c97268 Video bandwidth not reported correctly by sprang@webrtc.org · 11 years ago
  79. 0f78f7b Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc by sergeyu@chromium.org · 11 years ago
  80. 893c229 Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build. by wu@webrtc.org · 11 years ago
  81. be03ea6 Add delay limit to ChokeFilter. by solenberg@webrtc.org · 11 years ago
  82. 77c834d Logging for BWE test framework. by solenberg@webrtc.org · 11 years ago
  83. 9a1635a Make video/ only depend on video_engine_core. by pbos@webrtc.org · 11 years ago
  84. 6671434 Stop DirectTransports in VideoSendStreamTests. by pbos@webrtc.org · 11 years ago
  85. 267f694 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. by turaj@webrtc.org · 11 years ago
  86. 9ce61d4 Adding tl0idx consideration for continuity by mikhal@webrtc.org · 11 years ago
  87. 56290ed Fix build/isolate.gypi path in webrtc_tests.gypi. by pbos@webrtc.org · 11 years ago
  88. 8e3e298 Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. by fischman@webrtc.org · 11 years ago
  89. b581c90 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
  90. d4ec1f5 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
  91. 4043e7e Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
  92. b397091 Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h by xians@webrtc.org · 11 years ago
  93. d080e35 Added a "interleaved_" flag to webrtc::AudioFrame. by xians@webrtc.org · 11 years ago
  94. e2f3ebc Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_. by andrew@webrtc.org · 11 years ago
  95. ae2b602 Change the low-bitrate handling in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  96. 7af2f81 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  97. d6b231e Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined. by andrew@webrtc.org · 11 years ago
  98. cddf2b1 Add an extended filter option to audioproc. by andrew@webrtc.org · 11 years ago
  99. a881576 Fix for incorrect RTT estimation. A too low RTT value could be estimated. by asapersson@webrtc.org · 11 years ago
  100. ce21c82 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago