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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 47be73b8629244d6bb63a28198f97f040ce53d21
  1. 47be73b Adds a modified copy of talk/base to webrtc/base. It is the first step in by henrike@webrtc.org · 10 years ago
  2. efe9461 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 10 years ago
  3. 8378f1e Revert "FieldTrial implementation for webrtc." (rev 6089) by andresp@webrtc.org · 10 years ago
  4. c773ded Reduced kMaxSampleDiffMs (limit to 22fps). by asapersson@webrtc.org · 10 years ago
  5. 11de507 Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
  6. ae9a21f Deleting all NetEq3 files by henrik.lundin@webrtc.org · 10 years ago
  7. ad230ee The webrtc::AudioFrame struct contains a variable energy_. Since the energy isn't always calculated when the frame is created, this change makes the CalculateEnergy method in Audio Conference Mixer always calculate the energy. by henrik.lundin@webrtc.org · 10 years ago
  8. 50daa53 Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." by perkj@webrtc.org · 10 years ago
  9. c9ccea3 Deleting all ACM1 files by henrik.lundin@webrtc.org · 10 years ago
  10. 068cd6f Fix failing test introduced with r6111. by stefan@webrtc.org · 10 years ago
  11. b50d671 Fixes log spam introduced with r6041. by stefan@webrtc.org · 10 years ago
  12. 04e6703 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base. by henrike@webrtc.org · 10 years ago
  13. 7f5e297 Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  14. d2632a0 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  15. 3f0b9bf Echo cancellation functions docs: Follow style guide w.r.t. placement of * by kwiberg@webrtc.org · 10 years ago
  16. 12884ba Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  17. 6ca6896 One of the NetEq methods needs to be virtual. by turaj@webrtc.org · 10 years ago
  18. 53c1d3c Modifying neteq.gyp by turaj@webrtc.org · 10 years ago
  19. e639a03 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 10 years ago
  20. b8db407 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 10 years ago
  21. a4943ea Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  22. d88b46f FieldTrial implementation for webrtc. by andresp@webrtc.org · 10 years ago
  23. 6ecc773 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
  24. e1f0419 Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  25. ecbc55f AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ by kwiberg@webrtc.org · 10 years ago
  26. 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
  27. 519f74e Fix odd codes in video_capture on Mac. by braveyao@webrtc.org · 10 years ago
  28. a4bb5f2 video_render.gypi: clean up some libraries directives to be more specific. by fischman@webrtc.org · 10 years ago
  29. 39d9fa5 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  30. 3cd0f7c Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
  31. 60c62f8 Remove ALLOW_UNUSED. by andrew@webrtc.org · 10 years ago
  32. 1cbc360 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. by wu@webrtc.org · 10 years ago
  33. 6b6e3ea Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  34. 1f415cb Revert 6048 "Implement the Windows screen capturer using the Mag..." by tina.legrand@webrtc.org · 10 years ago
  35. c9d0634 WebRTCDemo: correct set trace filter operation. by braveyao@webrtc.org · 10 years ago
  36. 66a2eae Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest. by andrew@webrtc.org · 10 years ago
  37. 547a7cd Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  38. 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
  39. d885109 Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  40. 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
  41. c1878ac Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
  42. 785c2fd Fix constness of AudioBuffer accessors. by andrew@webrtc.org · 10 years ago
  43. cfed80f Fix a data race in ACM1 when audio is pulled. by turaj@webrtc.org · 10 years ago
  44. 151f6f2 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 10 years ago
  45. 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
  46. 2a52e53 Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() by henrike@webrtc.org · 10 years ago
  47. 223aa0f Disable failing GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
  48. 70deb1f Disable GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
  49. caa56eb Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios. by stefan@webrtc.org · 10 years ago
  50. 5435208 Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
  51. f9afa0d Fix iOS assembly compile error. by kjellander@webrtc.org · 10 years ago
  52. b2b3a14 Remove neteq_unittests from Android builds by henrik.lundin@webrtc.org · 10 years ago
  53. 2574899 Roll chromium_revision 260462:266514 by kjellander@webrtc.org · 10 years ago
  54. acfe5d3 Remove Version method from ACM1 by henrik.lundin@webrtc.org · 10 years ago
  55. 0f88af1 Remove ACM1 and NetEq3 related targets from modules.gyp by henrik.lundin@webrtc.org · 10 years ago
  56. 3f4fb3e Remove AudioCodingModuleFactory by henrik.lundin@webrtc.org · 10 years ago
  57. f9cbb15 Add clock to ACM config struct by henrik.lundin@webrtc.org · 10 years ago
  58. cca1d09 AEC: Startup phase only runs if reported_delay_enabled by bjornv@webrtc.org · 10 years ago
  59. f487c68 Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash. by fischman@webrtc.org · 10 years ago
  60. 7e6a355 APM: limit native sample rate to 16kHz on mobile. by fischman@webrtc.org · 10 years ago
  61. bd801bc Using realpath instead of android_src in Android webview by michaelbai@google.com · 10 years ago
  62. 8f445c4 Only download the VS toolchain if DEPOT_TOOLS_WIN_TOOLCHAIN=1. by andrew@webrtc.org · 10 years ago
  63. c476e64 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  64. 33d613a Disable flaky CaptureNtpTimeWithNetworkJitter. by pbos@webrtc.org · 10 years ago
  65. 90c64d6 AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h by bjornv@webrtc.org · 10 years ago
  66. 004a7b1 Disable capture test for FrameRate on Windows. by pbos@webrtc.org · 10 years ago
  67. 6c7fe5b Introduce a config struct for AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  68. fb3f14e Disabling flaky CanReceiveFec. by pbos@webrtc.org · 10 years ago
  69. c5fccd6 Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
  70. fc656dd Fix the NetEq build by henrik.lundin@webrtc.org · 10 years ago
  71. cbfcdd7 Include buffer size limits in NetEq config struct by henrik.lundin@webrtc.org · 10 years ago
  72. 55d952d Add henrik.lundin as owner in AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  73. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  74. b10fcf5 Fix leak in remote bitrate estimator tests introduced in r5980 by stefan@webrtc.org · 10 years ago
  75. 51ef6b0 Support for simulating multiple independent flows in a network. by stefan@webrtc.org · 10 years ago
  76. 1d95c5a Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
  77. dcbf62b Returns a NULL frame on all platforms if the captured window is closed. by jiayl@webrtc.org · 10 years ago
  78. 093fc0b Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  79. 47e54ba * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 10 years ago
  80. 1ca1e17 Add an output capacity parameter to ACMResampler::Resample10Msec() by henrik.lundin@webrtc.org · 10 years ago
  81. 39ae310 Add keyboard channel support to AudioBuffer. by andrew@webrtc.org · 10 years ago
  82. ab81d85 Fix the Android compilation (better structure for NetEq test libs) by henrik.lundin@webrtc.org · 10 years ago
  83. 99c0139 Remove TraceCallback use from Call. by pbos@webrtc.org · 10 years ago
  84. 16a058a Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 10 years ago
  85. aa07e3c Fixing a bug in ACM2 where the output frame energy was incorrectly set by henrik.lundin@webrtc.org · 10 years ago
  86. 6f9d083 Use unique filenames in AudioProcessingTests for parallelization. by andrew@webrtc.org · 10 years ago
  87. 376df2c AEC: Adds a reported_delay_enabled_ flag by bjornv@webrtc.org · 10 years ago
  88. 945d969 Remove ACM1/ACM2 switching from VoiceEngine tests by henrik.lundin@webrtc.org · 10 years ago
  89. b031016 Restore sample_rate_hz() until Chromium is updated to not use it. by andrew@webrtc.org · 10 years ago
  90. 467f756 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 10 years ago
  91. 6ce3720 Reland "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  92. 8e29911 Remove 44.1 kHz workaround from the iOS AudioDevice. by andrew@webrtc.org · 10 years ago
  93. fa29ce6 Fix a bug in AcmReceiver::NetworkStatistics by henrik.lundin@webrtc.org · 10 years ago
  94. e064ae4 Revert "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  95. eae2214 Stop using ACM factory in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  96. 6cee2ba Let A/V sync test use default AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  97. 139c7c1 Create ACM2 instance when calling AudioCodingModule::Create by henrik.lundin@webrtc.org · 10 years ago
  98. ff164b8 Reland "Make VoiceEngine choose ACM2 by default"" by henrik.lundin@webrtc.org · 10 years ago
  99. ffc2de0 audio_processing: DestroyHandle() now returns void by bjornv@webrtc.org · 10 years ago
  100. 642e80e common_audio: VADFree() now returns void by bjornv@webrtc.org · 10 years ago