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webrtc / src / 3f55dea259ba2a4f7c541ea4718fd879f6ae49e4 / webrtc / modules
  1. 3f55dea Replace scoped_ptr with unique_ptr in webrtc/modules/video_coding/ by kwiberg · 9 years ago
  2. a4f31bd TMMBRSet become vector<rtcp::TmmbItem> by danilchap · 9 years ago
  3. 739fcb9 Cleanup of webrtc::VideoFrame. by Niels Möller · 9 years ago
  4. 944744b Calculating ERLE in AEC more properly. by minyuel · 9 years ago
  5. f3ed9d7 Remove thread checker from CongestionController. by Stefan Holmer · 9 years ago
  6. 7e937e9 Remove workaround for Opus DTX noise pumping issue. by minyuel · 9 years ago
  7. 0a00759 Fix the stereo support in IntelligibilityEnhancer by aluebs · 9 years ago
  8. 54ebfca Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ ) by kjellander · 9 years ago
  9. 8b79b07 Move RTP module activation into PayloadRouter. by Peter Boström · 9 years ago
  10. 9c01725 Simplify registration of RTP-header extensions. by Peter Boström · 9 years ago
  11. 2080196 Cleanup of webrtc::VideoFrame. by nisse · 9 years ago
  12. c63f79a Fix ubsan warning in byteio_unittest by sprang · 9 years ago
  13. 8067068 iSAC entropy coder: Avoid signed integer overflow by kwiberg · 9 years ago
  14. fc59c44 Fix lowPowerModeEnabled crash on iOS8 by tkchin · 9 years ago
  15. 0e40f7c Remove incorrect reinterpret_cast from const. by Peter Boström · 9 years ago
  16. 3dd5d1d Remove PacketRouter sender distinction. by Peter Boström · 9 years ago
  17. a2644c0 Disable tests failing under UBSan to enable deployment to main waterfall. by kjellander@webrtc.org · 9 years ago
  18. a26ac92 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ ) by pbos · 9 years ago
  19. da33a8a Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ ) by torbjorng · 9 years ago
  20. 91c5b56 Remove DCHECK on duplicate packets in RemoteEstimatorProxy. by Stefan Holmer · 9 years ago
  21. bf66ae6 Remove thread check from PacketRouter::SendFeedback. by Stefan Holmer · 9 years ago
  22. f99af6b Fix the gain calculation in IntelligibilityEnhancer by aluebs · 9 years ago
  23. f6ff971 Fix division by zero in FindTMMBRBoundingSet by danilchap · 9 years ago
  24. f14c47a Remove ignored return code from modules. by Peter Boström · 9 years ago
  25. c379fcb Break out pacer thread from CongestionController to increase testability. by Stefan Holmer · 9 years ago
  26. 3e60bf0 Adds low complexity audio mode for single core CPUs by henrika · 9 years ago
  27. c2b785d Replace scoped_ptr with unique_ptr in webrtc/common_audio/ by kwiberg · 9 years ago
  28. 837b39e Fix ubsan warnings in BWE tests. by Stefan Holmer · 9 years ago
  29. f01633e Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/ by kwiberg · 9 years ago
  30. 58e08cb Reset indexer upon initialization in AudioLoop. by minyue · 9 years ago
  31. 0665f05 Fix OOB read in pacing test. by stefan · 9 years ago
  32. 12f4cda Histograms for H264EncoderImpl/H264DecoderImpl initialization and errors. by hbos · 9 years ago
  33. b7261fd iSAC float: Check for end of input buffer while decoding by kwiberg · 9 years ago
  34. f75d008 Bitrate controller for VideoToolbox encoder. by tkchin · 9 years ago
  35. 8df5d4f Moved the AEC C code to be built using C++. by peah · 9 years ago
  36. e80f9d0 Revert of Replace scoped_ptr with unique_ptr in webrtc/common_audio/ (patchset #4 id:60001 of https://codereview.webrtc.org/1712513002/ ) by kjellander · 9 years ago
  37. 77f3e0d Screen was flickering when the picker for desktop medias showed up in Windows platform. Keeping track of window size for each window so that BitBlt() instead of PrintWindow() will be called for windows with unchanged sizes. by gyzhou · 9 years ago
  38. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  39. 79d7a49 Replace scoped_ptr with unique_ptr in webrtc/common_audio/ by kwiberg · 9 years ago
  40. 18fcbcf Use VAD to get a better speech power estimation in the IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
  41. a094fd1 RTT intermediate calculation use ntp time instead of milliseconds. by Danil Chapovalov · 9 years ago
  42. 253d8fa Simplified the function for detecting whether capture data is modified. by peah · 9 years ago
  43. 91fe304 vp9: Adjust parameter for a test in videoprocessor_integrationtest.cc by Marco · 9 years ago
  44. a9d08929 Add initial bitrate and frame resolution parameters to quality scaler. by Alex Glaznev · 9 years ago
  45. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
  46. fc968a2 Fix sequence-number replay race for padding. by Peter Boström · 9 years ago
  47. 88788ad Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/ by kwiberg · 9 years ago
  48. 1e80ce4 webrtc::RtpPacket name freed for better RtpPacket by Danil Chapovalov · 9 years ago
  49. 4458d09 Drop support for playing output through aplay in intelligibility_proc by Alejandro Luebs · 9 years ago
  50. b3fb71c Add RTCAudioSession proxy class. by Zeke Chin · 9 years ago
  51. 9ac4df1 iOS: Enable modules_unittests and common_audio_unittests by kjellander · 9 years ago
  52. 66a9928 Roll chromium_revision 1d144ca..fa5d546 (375480:376142) by kjellander@webrtc.org · 9 years ago
  53. 0e2e50c Always append the BYE packet type at the end by aleungbroadsoft · 9 years ago
  54. 3ee73a5 Make RemoteBitrateEstimator::GetStats() virtual. by Stefan Holmer · 9 years ago
  55. 59c634b Re-add RemoteBitrateEstimator::GetStats. by Stefan Holmer · 9 years ago
  56. 3234819 Fix and simplify the power estimation in the IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
  57. dabf07f Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/vad/ by kwiberg · 9 years ago
  58. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  59. 62eaacf Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/ by kwiberg · 9 years ago
  60. 28c99bc iOS: Include legacy objc API in all.gyp + fix H264 libyuv dependency by kjellander · 9 years ago
  61. 69e59e6 [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer by danilchap · 9 years ago
  62. a332e2d Added boilerplate code for being able to test the upcoming AEC functionality. by peah · 9 years ago
  63. 0206000 iOS: Add resource files for tests and implement OutputPath by kjellander · 9 years ago
  64. 85d8bb0 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/transient/ by kwiberg · 9 years ago
  65. 2c38c20 Fix out-of-buffer write in iLBC by henrik.lundin · 9 years ago
  66. 44c65e9 Enable adaptive threshold experiment by default. by Stefan Holmer · 9 years ago
  67. 9d0c432 Remove video-codec max bitrate from TMMBN. by Peter Boström · 9 years ago
  68. ee31f0a Fix out-of-buffer read in iLBC by henrik.lundin · 9 years ago
  69. 62a5ccd Update bitrate only when we have incoming packet. by Stefan Holmer · 9 years ago
  70. 58cf5f1 Changed order of events when synthesizing a call. by peah · 9 years ago
  71. 0453ef8 Prevent busy-looping PacedSender on small packets. by Peter Boström · 9 years ago
  72. 16c5a96 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/ by kwiberg · 9 years ago
  73. 3747838 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/test/ by kwiberg · 9 years ago
  74. 2d0c332 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/ by kwiberg · 9 years ago
  75. 91d9756 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/ by kwiberg · 9 years ago
  76. be61562 Moved the GainControlForNewAGC class to be a separate file. by peah · 9 years ago
  77. e78765b Removes Nexus 5 from AEC and NS blacklists by henrika · 9 years ago
  78. b72dada Remove Reset from conditionally-compiled decoders. by Peter Boström · 9 years ago
  79. 73e2373 Fix negative shift exponent in WPDTree by aluebs · 9 years ago
  80. 66d2481 Fix division by zero errors in IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
  81. 9f35d55 Added accessor and Parse function. Create function merged into one. by danilchap · 9 years ago
  82. 07c51e3 Fix two UBSan warnings in NetEq by henrik.lundin · 9 years ago
  83. 42d8aa7 Fix IntelligibilityEnhancerTest.TestRenderUpdate test by Alejandro Luebs · 9 years ago
  84. c466bad Using the NS noise estimate for the IE by aluebs · 9 years ago
  85. 541f186 Cleanup temporary files created by tests. by jbauch · 9 years ago
  86. 1147b75 Moved buffering of farend into the EchoSubtraction method. by peah · 9 years ago
  87. 6608d9a NetEq: Fix a negative shift value by henrik.lundin · 9 years ago
  88. cc9669c Cleanup shared memory handling in DesktopCapturer interface. by sergeyu · 9 years ago
  89. fa639f0 Surface the noise estimate of the NS to be used by other components by Alejandro Luebs · 9 years ago
  90. 78ddd73 Update path for audioproc_debug proto output. by kjellander · 9 years ago
  91. 4bba35f Switch third_party/gflags to use updated GitHub repo. by kjellander · 9 years ago
  92. 09fef9e [rtp_rtcp] Added Sender Report Request rtcp packet. by danilchap · 9 years ago
  93. 0715a83 Avoid OpenH264 encoder bug for #threads > 1 on Mac and Chromium+Sandbox. by hbos · 9 years ago
  94. e594213 Fix div-by-0 in NetEq's StatisticsCalculator by henrik.lundin · 9 years ago
  95. d2a2296 Enable cpplint for webrtc/modules/pacing and fix all uncovered cpplint errors. by jbauch · 9 years ago
  96. c0ae305 Fix null-pointer dereference in RTPSenderVideo. by Peter Boström · 9 years ago
  97. 58c664c Clean up of CongestionController. by Stefan Holmer · 9 years ago
  98. 7336eeb [rtp_rtcp] rtcp::Tmmbn cleaned and got Parse function by danilchap · 9 years ago
  99. f6b5509 Fix GYP and GN references that are invalid in Chromium builds. by kjellander · 9 years ago
  100. e796f96 check v4l frame rate capability with bitwise method. by Weiyong Yao · 9 years ago