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webrtc / src / webrtc / 6c12dfacedf965e85995bafdcc75a35e54b208d9
  1. 6c12dfa Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ ) by peah · 9 years ago
  2. c6c9cdc Revert of Moved the ringbuffer to be built using C++ (patchset #2 id:20001 of https://codereview.webrtc.org/1851873003/ ) by peah · 9 years ago
  3. 4c1e43e Moved the ringbuffer to be built using C++ by peah · 9 years ago
  4. b16ff13 Revert of Set defines for Chromium build. (patchset #3 id:40001 of https://codereview.webrtc.org/1847013002/ ) by kjellander · 9 years ago
  5. 4b374eb Keep reads within buffer in AnalysisUpdateNeon(). by simon.hosie · 9 years ago
  6. cc10505 Moved ring-buffer related files from common_audio to audio_processing by peah · 9 years ago
  7. a5024df Make naming of APM perf test consistent by aluebs · 9 years ago
  8. 5aca9d7 Add CoreVideoFrameBuffer. by tkchin · 9 years ago
  9. d2e84e5 Don't reconfigure the encoder if the video options aren't changing. by deadbeef · 9 years ago
  10. 5d91518 Allowing a Java object field to be null in a new JNI helper method. by deadbeef · 9 years ago
  11. c459bcd Add mock AudioDeviceModule. by solenberg · 9 years ago
  12. db3e510 Make QualityScaler more responsive to downgrades. by Peter Boström · 9 years ago
  13. 81e36b5 Replace NULL with nullptr in webrtc/video. by Peter Boström · 9 years ago
  14. 82f26cb Deprecate GetWidth() and GetHeight() methods. Replaced by width() and height(). by nisse · 9 years ago
  15. caa1632 Set defines for Chromium build. by kjellander@webrtc.org · 9 years ago
  16. c57e5bc Avoid rescheduling the next RTCP packet if the RTCP sender status doesn't change. by skvlad · 9 years ago
  17. 288ef7c Tweak kDecayRate in the IntelligibilityEnhancer by Alejandro Luebs · 9 years ago
  18. 3bbeaf8 Fix normalization of noise estimate in NoiseSuppressor by Alejandro Luebs · 9 years ago
  19. 69562b7 Re-enabling tests that were disabled for Windows debug builds. by Taylor Brandstetter · 9 years ago
  20. a9aae60 Fix C4434 warning about 32-bit shift assigned to 64-bits by brucedawson · 9 years ago
  21. 4f5eb5c External denoiser based on noise estimation and moving object detection. by jackychen · 9 years ago
  22. f97e5d7 Disable VideoCaptureExternalTest.FrameRate on Mac by henrik.lundin · 9 years ago
  23. e477a02 GYP: Add webrtc/pc/pc.gyp:* to 'All' target. by Henrik Kjellander · 9 years ago
  24. c70e4ec Packet buffer for the new jitter buffer. by philipel · 9 years ago
  25. 66a510a Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector, by nisse · 9 years ago
  26. a19e26e Remove deprecated RtpReceiver::CreateAudioReceiver() function. by solenberg · 9 years ago
  27. 2d13bc6 Give a more specific URL for creating WebRTC checkout by brucedawson · 9 years ago
  28. 65fad3e Delete unused cricket::VideoFrame methods MakeExclusive and CopyToFrame. by nisse · 9 years ago
  29. b600d48 Only split into bands when the reverse stream is analyzed in the APM by Alejandro Luebs · 9 years ago
  30. 91d31fe Improve iOS frame capture threading. by tkchin · 9 years ago
  31. d8df5e5 Use mobile platform settings for VP8 and VP9 decoders on all Android builds. by Alex Glaznev · 9 years ago
  32. 56d9dfc Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  33. e47efce Minor ObjC header updates. by tkchin · 9 years ago
  34. 6bf669c Remove webrtc::ScopedVector by kwiberg · 9 years ago
  35. 3c4cb59 Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool> by Per · 9 years ago
  36. e1e16f0 Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ ) by kjellander · 9 years ago
  37. 83147bf This CL addresses late feedback on https://codereview.webrtc.org/1683193003/ by Torbjorn Granlund · 9 years ago
  38. 6e044aa Android EGL: Synchronize calls to eglSwapBuffers and eglMakeCurrent by Magnus Jedvert · 9 years ago
  39. 7c6a74f Add RTCConfiguration getter and setter methods. The immediate plan is to move some flags into an embedded MediaConfig (https://codereview.webrtc.org/1818033002/), which will be possible after Chrome is updated to use these new setter methods. by Niels Möller · 9 years ago
  40. 892e6df Android SurfaceTextureHelper: Distinguish thread names for decoder and camera by magjed · 9 years ago
  41. a7f1aa5 Limit max spatial layers to be configured through field trial (3->2) to match current limit in VP9EncoderImpl::InitEncode. by asapersson · 9 years ago
  42. 018a259 Add histogram stats for average QP per frame for VP8 (for sent video streams): by asapersson · 9 years ago
  43. d25cdbb - Add temporary VoEBase::audio_device_module() method. by solenberg · 9 years ago
  44. 9b791e4 MB: Remove unnecessary configs. by kjellander · 9 years ago
  45. 78a25a2 Changed the names of some of the bitexactness unittests to by peah · 9 years ago
  46. 5fb249f Changed tests to be DISABLED on non-supported platforms rather than not to build at all. by peah · 9 years ago
  47. 10e2675 [rtcp] Sdes::Parse updated not to use RTCPUtility by danilchap · 9 years ago
  48. c6ff47e Signal ready-to-send when switching to a writable connection. by Honghai Zhang · 9 years ago
  49. 1f0f771 Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ ) by kjellander · 9 years ago
  50. 463cc09 Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. by kjellander · 9 years ago
  51. da65a68 Make rtcp sender use max transfer unit. by danilchap · 9 years ago
  52. 88ef094 [rtcp] ReceiverReport::Parse updated not to use RTCPUtility by danilchap · 9 years ago
  53. acf1581 Fix division by zero in NonlinearBeamformer by aluebs · 9 years ago
  54. 5ea80b1 Renamed the test::BitExactFrame method to test::VectorDifferenceBounded. by peah · 9 years ago
  55. d46864d Shorten single-stream VP8 HW implementation names. by Peter Boström · 9 years ago
  56. ca09a4b Split ByteBuffer into writer/reader objects. by jbauch · 9 years ago
  57. d0e6010 Rent-A-Codec: Reference count the shared iSAC bandwidth estimation state by kwiberg · 9 years ago
  58. 80eacc6 AudioCodingModule: Add methods for injecting external encoder stacks by kwiberg · 9 years ago
  59. 84bb523 Reland https://codereview.webrtc.org/1802993002/ by solenberg · 9 years ago
  60. 5ed7972 New method I420Buffer::Copy. by nisse · 9 years ago
  61. 82688e9 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  62. 603a219 Adding JNI binding for 'active' field in RTP encodings. by Taylor Brandstetter · 9 years ago
  63. 67d4abe Remove unused stuff from AudioFrame: by solenberg · 9 years ago
  64. fda92c0 Added missing TODOs in the beamformer unit test code. by peah · 9 years ago
  65. 14d1e8c Reland: Add IntelligibilityEnhancer support to audioproc_float by Alejandro Luebs · 9 years ago
  66. 88a9699 Making new unit test assertions use the standard timeout. by Taylor Brandstetter · 9 years ago
  67. 8263d42 Revert "Add IntelligibilityEnhancer support to audioproc_float" by Alejandro Luebs · 9 years ago
  68. 374ac25 Add IntelligibilityEnhancer support to audioproc_float by Alejandro Luebs · 9 years ago
  69. 8e9df80 Update QuicTransportChannel to latest version of libquic by mikescarlett · 9 years ago
  70. 14d9fb1 Don't call operator== with scoped_ptr<T> and T* by kwiberg · 9 years ago
  71. 920b86a Remove calls to rtc::scoped_ptr::accept by kwiberg · 9 years ago
  72. ce55976 Reland of Added a bitexactness test for the gain controller in the audio processing module. by peah · 9 years ago
  73. b9c001f Remove accidentally readded webrtc/base/sslstreamadapterhelper.cc by Henrik Kjellander · 9 years ago
  74. 12cfa58 Remove all uses of the HAVE_CONFIG_H define. by Henrik Kjellander · 9 years ago
  75. 9d541b1 [rtcp] SenderReport::Parse updated not to use RTCPUtility by danilchap · 9 years ago
  76. 893fb1f Move to x509 v3 as required by the WebRTC draft. by torbjorng · 9 years ago
  77. 8d360b6 Remove orphaned files. by torbjorng · 9 years ago
  78. 91b510a Added a bitexactness test for the beamformer in the audio processing module by peah · 9 years ago
  79. e884401 Android HW decoder: Add support for textures when using EGL 1.0 by magjed · 9 years ago
  80. 908eeb9 Change include in metrics.h (change to use systems_wrappers/include/logging.h, base logging breaks chromium.fyi). by asapersson · 9 years ago
  81. 7274c0a Fix typo in FakeAdmTest.TestProcess name. by Peter Boström · 9 years ago
  82. 63b8d3f Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*). by asapersson · 9 years ago
  83. d27854b Allow passing in strings of length zero to FileWrapper::Write without closing the file. by terelius · 9 years ago
  84. 7a29e63 Re-reland of Added a bitexactness test for the echo canceller in the audio processing module. by peah · 9 years ago
  85. 019caf7 Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates by Per · 9 years ago
  86. 7766e51 Added a bitexactness test for the intelligibility enhancer in the audio processing module by peah · 9 years ago
  87. 770073f Build dynamic framework with podspec for Objective-C API. by Jon Hjelle · 9 years ago
  88. 08ef89b Revert of Added a bitexactness test for the gain controller in the audio processing module. (patchset #3 id:60001 of https://codereview.webrtc.org/1812433002/ ) by peah · 9 years ago
  89. d79962e Added the JNI interface to get and set RtpParameters and the maximum bitrate limits. by skvlad · 9 years ago
  90. 5c481f7 Added a bitexactness test for the gain controller in the audio processing module. by peah · 9 years ago
  91. d4e1bb9 Adding BlockMeanCalculator for AEC. by minyue · 9 years ago
  92. c92ec0a Fixed a potential deadlock problem in the AGC by peah · 9 years ago
  93. b120b5b Avoid clicks when muting/unmuting a voe::Channel. by solenberg · 9 years ago
  94. 57abdbf Don't override curve preferences in BoringSSL. by David Benjamin · 9 years ago
  95. df760cd Revert of Added a bitexactness test for the echo canceller in the audio processing module. (patchset #2 id:40001 of https://codereview.webrtc.org/1827833006/ ) by guidou · 9 years ago
  96. f2defc4 ReAdd dummy MediaStreamTrack::set_state to make Chrome build happy. by perkj · 9 years ago
  97. 2bef95e Revert of Remove code interfacing legacy openssl. (patchset #3 id:40001 of https://codereview.webrtc.org/1808763002/ ) by Torbjorn Granlund · 9 years ago
  98. 6338d74 More cleanup of cricket::VideoCapturer by perkj · 9 years ago
  99. 3c013b3 Removed MediaStreamTrackInterface::set_state by perkj · 9 years ago
  100. 3716131 Reland of Added a bitexactness test for the echo canceller in the audio processing module. by peah · 9 years ago