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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / d658c1128db26424e73674abd5a5051cc8ef15ab
  1. d658c11 Update makefiles after merge of Chromium at 269467 by Torne (Richard Coles) · 10 years ago
  2. 56c3ac0 Update makefiles after merge of Chromium at 269336 by Android Chromium Automerger · 10 years ago
  3. b43e80d Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at e1f0419756d58f744f902fe62fa31da126c55abe by Android Chromium Automerger · 10 years ago
  4. e1f0419 Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  5. ecbc55f AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ by kwiberg@webrtc.org · 10 years ago
  6. 5b371a8 Update makefiles after merge of Chromium at 269041 by Android Chromium Automerger · 10 years ago
  7. 0a2aa9a Update makefiles after merge of Chromium at 269030 by Android Chromium Automerger · 10 years ago
  8. 13ea3ca Update makefiles after merge of Chromium at 269003 by Android Chromium Automerger · 10 years ago
  9. 78c3fc4 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at d2fb259b3bc61c68f368929510215a7ee7d00fda by Android Chromium Automerger · 10 years ago
  10. d2fb259 Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine. by wu@webrtc.org · 10 years ago
  11. ab0ecd1 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 519f74e6b0c8a59db50e58158331f8922c4509b9 by Primiano Tucci · 10 years ago
  12. c52369a Update makefiles after merge of Chromium at 268710 by Android Chromium Automerger · 10 years ago
  13. 625ace9 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 39d9fa5486157fb4b3ab28ae403aeaa6d651e92b by Android Chromium Automerger · 10 years ago
  14. 519f74e Fix odd codes in video_capture on Mac. by braveyao@webrtc.org · 10 years ago
  15. a4bb5f2 video_render.gypi: clean up some libraries directives to be more specific. by fischman@webrtc.org · 10 years ago
  16. 39d9fa5 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  17. 0de5f22 Update makefiles after merge of Chromium at 268379 by Android Chromium Automerger · 10 years ago
  18. 3cd0f7c Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
  19. 60c62f8 Remove ALLOW_UNUSED. by andrew@webrtc.org · 10 years ago
  20. 1cbc360 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. by wu@webrtc.org · 10 years ago
  21. 6b6e3ea Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  22. 1f415cb Revert 6048 "Implement the Windows screen capturer using the Mag..." by tina.legrand@webrtc.org · 10 years ago
  23. c9d0634 WebRTCDemo: correct set trace filter operation. by braveyao@webrtc.org · 10 years ago
  24. 66a2eae Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest. by andrew@webrtc.org · 10 years ago
  25. 547a7cd Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  26. 99ec896 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. by henrika@webrtc.org · 10 years ago
  27. d885109 Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  28. 4d61a36 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 10 years ago
  29. a15f0aa Update makefiles after merge of Chromium at 267293 by Android Chromium Automerger · 10 years ago
  30. 3a802b9 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at cfed80f78803395dd066261afb5c5d99e5048d5d by Android Chromium Automerger · 10 years ago
  31. c1878ac Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
  32. 785c2fd Fix constness of AudioBuffer accessors. by andrew@webrtc.org · 10 years ago
  33. dfa35b4 Update makefiles after merge of Chromium at 266947 by Android Chromium Automerger · 10 years ago
  34. cfed80f Fix a data race in ACM1 when audio is pulled. by turaj@webrtc.org · 10 years ago
  35. 151f6f2 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 10 years ago
  36. 93ae821 Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65 by henrike@webrtc.org · 10 years ago
  37. 2a52e53 Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() by henrike@webrtc.org · 10 years ago
  38. 223aa0f Disable failing GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
  39. 70deb1f Disable GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
  40. caa56eb Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios. by stefan@webrtc.org · 10 years ago
  41. 5435208 Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
  42. f9afa0d Fix iOS assembly compile error. by kjellander@webrtc.org · 10 years ago
  43. b2b3a14 Remove neteq_unittests from Android builds by henrik.lundin@webrtc.org · 10 years ago
  44. 2574899 Roll chromium_revision 260462:266514 by kjellander@webrtc.org · 10 years ago
  45. acfe5d3 Remove Version method from ACM1 by henrik.lundin@webrtc.org · 10 years ago
  46. 0f88af1 Remove ACM1 and NetEq3 related targets from modules.gyp by henrik.lundin@webrtc.org · 10 years ago
  47. 3f4fb3e Remove AudioCodingModuleFactory by henrik.lundin@webrtc.org · 10 years ago
  48. f9cbb15 Add clock to ACM config struct by henrik.lundin@webrtc.org · 10 years ago
  49. cca1d09 AEC: Startup phase only runs if reported_delay_enabled by bjornv@webrtc.org · 10 years ago
  50. f487c68 Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash. by fischman@webrtc.org · 10 years ago
  51. 7e6a355 APM: limit native sample rate to 16kHz on mobile. by fischman@webrtc.org · 10 years ago
  52. bd801bc Using realpath instead of android_src in Android webview by michaelbai@google.com · 10 years ago
  53. 8f445c4 Only download the VS toolchain if DEPOT_TOOLS_WIN_TOOLCHAIN=1. by andrew@webrtc.org · 10 years ago
  54. 109a4e7 Update makefiles after merge of Chromium at 266543 by Android Chromium Automerger · 10 years ago
  55. c476e64 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  56. 33d613a Disable flaky CaptureNtpTimeWithNetworkJitter. by pbos@webrtc.org · 10 years ago
  57. 90c64d6 AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h by bjornv@webrtc.org · 10 years ago
  58. 004a7b1 Disable capture test for FrameRate on Windows. by pbos@webrtc.org · 10 years ago
  59. 6c7fe5b Introduce a config struct for AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  60. fb3f14e Disabling flaky CanReceiveFec. by pbos@webrtc.org · 10 years ago
  61. c5fccd6 Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
  62. fc656dd Fix the NetEq build by henrik.lundin@webrtc.org · 10 years ago
  63. cbfcdd7 Include buffer size limits in NetEq config struct by henrik.lundin@webrtc.org · 10 years ago
  64. 55d952d Add henrik.lundin as owner in AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  65. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  66. 0c9f7d3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 1d95c5aecd0a4924f39f24834fb06d06e61f181e by Android Chromium Automerger · 10 years ago
  67. b10fcf5 Fix leak in remote bitrate estimator tests introduced in r5980 by stefan@webrtc.org · 10 years ago
  68. 51ef6b0 Support for simulating multiple independent flows in a network. by stefan@webrtc.org · 10 years ago
  69. 1d95c5a Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
  70. dcbf62b Returns a NULL frame on all platforms if the captured window is closed. by jiayl@webrtc.org · 10 years ago
  71. 093fc0b Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  72. 47e54ba * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 10 years ago
  73. 1ca1e17 Add an output capacity parameter to ACMResampler::Resample10Msec() by henrik.lundin@webrtc.org · 10 years ago
  74. 39ae310 Add keyboard channel support to AudioBuffer. by andrew@webrtc.org · 10 years ago
  75. ab81d85 Fix the Android compilation (better structure for NetEq test libs) by henrik.lundin@webrtc.org · 10 years ago
  76. 99c0139 Remove TraceCallback use from Call. by pbos@webrtc.org · 10 years ago
  77. 16a058a Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 10 years ago
  78. aa07e3c Fixing a bug in ACM2 where the output frame energy was incorrectly set by henrik.lundin@webrtc.org · 10 years ago
  79. 6f9d083 Use unique filenames in AudioProcessingTests for parallelization. by andrew@webrtc.org · 10 years ago
  80. 0197db2 Update makefiles after merge of Chromium at 265680 by Android Chromium Automerger · 10 years ago
  81. 376df2c AEC: Adds a reported_delay_enabled_ flag by bjornv@webrtc.org · 10 years ago
  82. 91734bd Update makefiles after merge of Chromium at 265607 by Android Chromium Automerger · 10 years ago
  83. 658bfc4 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at b031016337f09d758a97ed51b67788e574431103 by Android Chromium Automerger · 10 years ago
  84. 945d969 Remove ACM1/ACM2 switching from VoiceEngine tests by henrik.lundin@webrtc.org · 10 years ago
  85. b031016 Restore sample_rate_hz() until Chromium is updated to not use it. by andrew@webrtc.org · 10 years ago
  86. 467f756 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 10 years ago
  87. 6ce3720 Reland "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  88. 8e29911 Remove 44.1 kHz workaround from the iOS AudioDevice. by andrew@webrtc.org · 10 years ago
  89. fa29ce6 Fix a bug in AcmReceiver::NetworkStatistics by henrik.lundin@webrtc.org · 10 years ago
  90. e064ae4 Revert "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  91. eae2214 Stop using ACM factory in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  92. 6cee2ba Let A/V sync test use default AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  93. 139c7c1 Create ACM2 instance when calling AudioCodingModule::Create by henrik.lundin@webrtc.org · 10 years ago
  94. ff164b8 Reland "Make VoiceEngine choose ACM2 by default"" by henrik.lundin@webrtc.org · 10 years ago
  95. ffc2de0 audio_processing: DestroyHandle() now returns void by bjornv@webrtc.org · 10 years ago
  96. 642e80e common_audio: VADFree() now returns void by bjornv@webrtc.org · 10 years ago
  97. 708ff4d Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 10 years ago
  98. 74658f6 Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 10 years ago
  99. 998d063 Revert r5937 "Fix multi-monitor support in the screen capturer for Mac." by sergeyu@chromium.org · 10 years ago
  100. 6958c0f Add Chromium's ScopedVector. by andrew@webrtc.org · 10 years ago