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webrtc / src / webrtc / d63e47be28be0c5d18e1985a7ff91cfd385679db
  1. d63e47b Cleanup: Remove MD5_CTX typedef. by Thiago Farina · 10 years ago
  2. 94fec52 Revert "Port frame_analyzer and rgba_to_i420_converter targets to GN build." by Henrik Kjellander · 10 years ago
  3. 5d397d8 Port frame_analyzer and rgba_to_i420_converter targets to GN build. by Henrik Kjellander · 10 years ago
  4. 5173c60 Remove henrike@ from OWNERS by Henrik Kjellander · 10 years ago
  5. a5b531b Revert "Split EventWrapper in twain." by Minyue · 10 years ago
  6. b5741b6 Revert "Enable CVO by default through webrtc pipeline." by Minyue · 10 years ago
  7. 5195090 Ensures that AudioManager.isVolumeFixed() is only used for Android L and above by henrika · 10 years ago
  8. 7ab7360 Remove duplicated source listing of gtest_prod_util.h by Henrik Kjellander · 10 years ago
  9. f89343c Fix bug in WebRtcIsacfix_FilterMaLoopNeon. by Zhongwei Yao · 10 years ago
  10. 4ca8fcb Remove er_tables_xor.h. by Peter Boström · 10 years ago
  11. 4e6bfb0 Enable CVO by default through webrtc pipeline. by Guo-wei Shieh · 10 years ago
  12. 7c14951 VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const. by Magnus Jedvert · 10 years ago
  13. c72ec5d Revert "Avoid critsect for protection- and qm setting callbacks in VideoSender." by mflodman · 10 years ago
  14. a611ad0 Reject RTP one-byte extension ID 0. by Peter Boström · 10 years ago
  15. a513406 Avoid critsect for protection- and qm setting callbacks in VideoSender. by mflodman · 10 years ago
  16. 0c73b7e Remove old suppression for ProcessThreadImpl. by Tommi · 10 years ago
  17. 6dd58de common_audio: Explicit cast in WebRtcSpl_NormW16 on ARM by Bjorn Volcker · 10 years ago
  18. 04ba279 Revert "Suppress data races in libjingle_peerconnection_unittest" by Tommi · 10 years ago
  19. ddde4b2 Remove non-functional asynchronous resampling mode. by Andrew MacDonald · 10 years ago
  20. 046319e Introduce CodecManager and move code from AudioCodingModuleImpl by Henrik Lundin · 10 years ago
  21. d1bec1e Suppress "EndToEndTest::ReceivedFecPacketsNotNacked" on Asan, Tsan by Minyue Li · 10 years ago
  22. fc397a2 Add locks to Start(), Stop() methods in ProcessThread. by Tommi · 10 years ago
  23. f26dff7 Introduce AudioCodingModuleImpl::current_encoder_ by Henrik Lundin · 10 years ago
  24. f64db88 Clamp decoder sample rate to 32000 in iSAC by Henrik Lundin · 10 years ago
  25. 54dc0bb Fix gyp path for bwe simulator include. by Stefan Holmer · 10 years ago
  26. 4696418 Suppress data races in libjingle_peerconnection_unittest by Henrik Kjellander · 10 years ago
  27. be04880 GN: Cleanup no longer needed libvpx config. by Henrik Kjellander · 10 years ago
  28. 1f04531 Additional suppression for TSan deadlock detection by Henrik Kjellander · 10 years ago
  29. 28542a0 Add tests for r8811. by Peter Boström · 10 years ago
  30. 9f97dce Suppress TSan errors triggered when deadlock detection is enabled. by Henrik Kjellander · 10 years ago
  31. 55d2426 Final minor fix in WebRtcAudioManager by henrika · 10 years ago
  32. 2452911 audio_processing/agc: Put entire method set_output_will_be_muted() under lock by Bjorn Volcker · 10 years ago
  33. 6f4e0c4 Adding playout volume control to WebRtcAudioTrack.java. by henrika · 10 years ago
  34. 0bd0468 Add a lock to NSSContext to fix data race by Jiayang Liu · 10 years ago
  35. fbd5555 Update speed setting in VP9. by Marco · 10 years ago
  36. 547c64a AcmReceiver: index decoders by payload type instead of ACM codec ID by Jelena Marusic · 10 years ago
  37. d5fbc22 Add some sanity CHECKs to webrtc::Call. by Peter Boström · 10 years ago
  38. cb76ae7 Fix build error introduced by r8864. by Stefan Holmer · 10 years ago
  39. ca55fa1 Moving the pacer and the pacer thread to ChannelGroup. by Stefan Holmer · 10 years ago
  40. 70e3d3d Reparent Nonlinear beamformer under beamforming interface. by Michael Graczyk · 10 years ago
  41. 67171f3 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android by Bjorn Volcker · 10 years ago
  42. a37e3fb Register sample rate of Audio RED in RTPPayloadRegistry. by Minyue Li · 10 years ago
  43. 94053f6 Fix crash on decode found by fuzz tester. by Stefan Holmer · 10 years ago
  44. edd91ff Refactor common_audio/vad: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
  45. d067f0d Remove video from WebRTC Android example. by Per · 10 years ago
  46. 36be253 Refactor common_audio/signal_processing: Removed usage of WEBRTC_SPL_MUL_16_16_RSFT by Bjorn Volcker · 10 years ago
  47. 3918de6 Address comments from cr 43769004. by Tommi · 10 years ago
  48. 3ab0411 Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/. by Donald Curtis · 10 years ago
  49. dd6794b rtc::Buffer: Restore length method for backwards compatibility by kwiberg@webrtc.org · 10 years ago
  50. 8c2ce2a Remove I420VideoFrame::SwapFrame by magjed@webrtc.org · 10 years ago
  51. 3213b36 Remove I420VideoFrame::CloneFrame by magjed@webrtc.org · 10 years ago
  52. da161da Improve logging and add DCHECKs in codec database. by pbos@webrtc.org · 10 years ago
  53. a1c44bb rtc::Buffer: Rename length to size, for conformance with the STL by kwiberg@webrtc.org · 10 years ago
  54. 2112dbb Allow setting thread priorities in Chromium on all but linux platforms. by tommi@webrtc.org · 10 years ago
  55. 1f00000 Split EventWrapper in twain. by tommi@webrtc.org · 10 years ago
  56. 6eaf09a Convert webrtc/video/ abort/assert to CHECK/DCHECK. by pbos@webrtc.org · 10 years ago
  57. d2759e7 Limit RED audio payload to narrow band. by minyue@webrtc.org · 10 years ago
  58. 56dc67d Temporarily disable SetPriority when building with Chromium. by tommi@webrtc.org · 10 years ago
  59. 607f5b3 Scaler: Recycle allocations using buffer pool. by magjed@webrtc.org · 10 years ago
  60. 2c6ff66 Disable PLC for iSAC by henrik.lundin@webrtc.org · 10 years ago
  61. 65e5aa1 Remove unused version.py script. by kjellander@webrtc.org · 10 years ago
  62. 9f5fdd4 Fix build failure by jmarusic@webrtc.org · 10 years ago
  63. b5efa46 AcmReceiver: use std::map instead of an array to keep the list of decoders by jmarusic@webrtc.org · 10 years ago
  64. cecb6ae Prevent asserting on unset start bitrate. by pbos@webrtc.org · 10 years ago
  65. 01c1481 Re-land 8810 "- Add a SetPriority method to ThreadWr..." by tommi@webrtc.org · 10 years ago
  66. eb2730e Revert 8810 "- Add a SetPriority method to ThreadWrapper" by tommi@webrtc.org · 10 years ago
  67. 462472f 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
  68. b787c8f Document the 'int' return value of Resampler methods. by wtc@chromium.org · 10 years ago
  69. ce07d7c Minor fix for MIPS Android build. by andrew@webrtc.org · 10 years ago
  70. cf7fc77 Fix code to handle crashes for non-VP8. by pbos@webrtc.org · 10 years ago
  71. be8dbba - Add a SetPriority method to ThreadWrapper by tommi@webrtc.org · 10 years ago
  72. 213c6ca Release buffer pool in Vp8DecoderImpl::Release(). by pbos@webrtc.org · 10 years ago
  73. 792c873 Make screenshare target bitrate experiment always on by pbos@webrtc.org · 10 years ago
  74. 0db69ce Clean up webrtc external capture. by perkj@webrtc.org · 10 years ago
  75. e9a37fa Remove FullStackTest frame pointer handles. by pbos@webrtc.org · 10 years ago
  76. 528348a Prevent crashes when copying a zero-size frame. by pbos@webrtc.org · 10 years ago
  77. 5e08fc4 Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  78. f42f229 Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32 by bjornv@webrtc.org · 10 years ago
  79. 82d9c74 Remove build-time beamformer flags. by andrew@webrtc.org · 10 years ago
  80. 2db66fc Add the Ooura FFT to RealFourier. by andrew@webrtc.org · 10 years ago
  81. 403ca2b Adds full-duplex unit test to AudioDeviceTest on Android by henrika@webrtc.org · 10 years ago
  82. 0e57be8 Use scoped_ptr for ThreadWrapper::CreateThread. by tommi@webrtc.org · 10 years ago
  83. 42cc212 Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper. by tommi@webrtc.org · 10 years ago
  84. 78aaaa6 Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame" by magjed@webrtc.org · 10 years ago
  85. 50fb934 Use atomic operations for setting/reading the trace filter. by tommi@webrtc.org · 10 years ago
  86. cb9fe9c Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal() by jmarusic@webrtc.org · 10 years ago
  87. eac491c Remove command-line tool 'video_coding_test'. by pbos@webrtc.org · 10 years ago
  88. 3b9d0a5 Split C++ class from macro overrides to fix Chromium build by tommi@webrtc.org · 10 years ago
  89. 09e4ed8 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
  90. 3bb0ab8 vp8: Add missing call to SetUsageMessage(). by kjellander@webrtc.org · 10 years ago
  91. 76919af Renaming neteq_opus_fec_quality_test. by minyue@webrtc.org · 10 years ago
  92. 7131030 Base start bitrate on last observed bitrate. by pbos@webrtc.org · 10 years ago
  93. a7347d2 DCHECK frame parameters instead of return codes. by pbos@webrtc.org · 10 years ago
  94. 328ab85 Use SendTimeHistory to keep track of send times in simulations. by stefan@webrtc.org · 10 years ago
  95. a94d042 Removing henrik.lundin from OWNERS in video_coding/* by henrik.lundin@webrtc.org · 10 years ago
  96. bddb588 Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" by perkj@webrtc.org · 10 years ago
  97. 19e0384 Make AudioDecoder stateless by henrik.lundin@webrtc.org · 10 years ago
  98. 75fb068 Delete ACMGenericCodec::Encode and use AudioEncoder::Encode directly by henrik.lundin@webrtc.org · 10 years ago
  99. 64f52f0 Revert 8749 "We changed Encode() and EncodeInternal() return typ..." by tommi@webrtc.org · 10 years ago
  100. a80280f Fix FYI build - add a missing include to event_tracer.h in system_wrappers. by tommi@webrtc.org · 10 years ago