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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 5c6f3fd622b58453109bfefc48e0f2ccc6a5e8ee
  1. 5c6f3fd 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
  2. d12d7b6 FieldTrial implementation for webrtc. by andresp@webrtc.org · 10 years ago
  3. 0d47fe1 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
  4. 569487d Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  5. 8b4f539 AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ by kwiberg@webrtc.org · 10 years ago
  6. 60f1422 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
  7. 8539c4a Fix odd codes in video_capture on Mac. by braveyao@webrtc.org · 10 years ago
  8. 4fb1a55 video_render.gypi: clean up some libraries directives to be more specific. by fischman@webrtc.org · 10 years ago
  9. 73c2412 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  10. 8ec46c6 Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
  11. 96cccf7 Remove ALLOW_UNUSED. by andrew@webrtc.org · 10 years ago
  12. 5e44f56 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. by wu@webrtc.org · 10 years ago
  13. ebb4b94 Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  14. 7c434be Revert 6048 "Implement the Windows screen capturer using the Mag..." by tina.legrand@webrtc.org · 10 years ago
  15. 6ccd081 WebRTCDemo: correct set trace filter operation. by braveyao@webrtc.org · 10 years ago
  16. 5cc0d0b Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest. by andrew@webrtc.org · 10 years ago
  17. c2b27b5 Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  18. ba9daa7 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. by henrika@webrtc.org · 10 years ago
  19. 1a9e6ac Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  20. 42fe6b3 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
  21. 616cbcd Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
  22. c2e6438 Fix constness of AudioBuffer accessors. by andrew@webrtc.org · 10 years ago
  23. 0638464 Fix a data race in ACM1 when audio is pulled. by turaj@webrtc.org · 10 years ago
  24. 976ce98 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 10 years ago
  25. f13f4a7 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
  26. 40b200b Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() by henrike@webrtc.org · 10 years ago
  27. 3d5905b Disable failing GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
  28. d592231 Disable GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
  29. 3848107 Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios. by stefan@webrtc.org · 10 years ago
  30. c6cfc5c Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
  31. c78232f Fix iOS assembly compile error. by kjellander@webrtc.org · 10 years ago
  32. c523211 Remove neteq_unittests from Android builds by henrik.lundin@webrtc.org · 10 years ago
  33. ad4cce6 Roll chromium_revision 260462:266514 by kjellander@webrtc.org · 10 years ago
  34. 3cbb2df Remove Version method from ACM1 by henrik.lundin@webrtc.org · 10 years ago
  35. dc37088 Remove ACM1 and NetEq3 related targets from modules.gyp by henrik.lundin@webrtc.org · 10 years ago
  36. 68a95e1 Remove AudioCodingModuleFactory by henrik.lundin@webrtc.org · 10 years ago
  37. a48f3c2 Add clock to ACM config struct by henrik.lundin@webrtc.org · 10 years ago
  38. db395e4 AEC: Startup phase only runs if reported_delay_enabled by bjornv@webrtc.org · 10 years ago
  39. be039c2 Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash. by fischman@webrtc.org · 10 years ago
  40. b4945d1 APM: limit native sample rate to 16kHz on mobile. by fischman@webrtc.org · 10 years ago
  41. 93d270f Using realpath instead of android_src in Android webview by michaelbai@google.com · 10 years ago
  42. a2d989b Only download the VS toolchain if DEPOT_TOOLS_WIN_TOOLCHAIN=1. by andrew@webrtc.org · 10 years ago
  43. c54ff69 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  44. 267637b Disable flaky CaptureNtpTimeWithNetworkJitter. by pbos@webrtc.org · 10 years ago
  45. 0e098e0 AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h by bjornv@webrtc.org · 10 years ago
  46. 676638c Disable capture test for FrameRate on Windows. by pbos@webrtc.org · 10 years ago
  47. bb62a93 Introduce a config struct for AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  48. c298835 Disabling flaky CanReceiveFec. by pbos@webrtc.org · 10 years ago
  49. 73e1a8b Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
  50. abf78cc Fix the NetEq build by henrik.lundin@webrtc.org · 10 years ago
  51. 75d1487 Include buffer size limits in NetEq config struct by henrik.lundin@webrtc.org · 10 years ago
  52. a714643 Add henrik.lundin as owner in AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  53. b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  54. 4820f6b Fix leak in remote bitrate estimator tests introduced in r5980 by stefan@webrtc.org · 10 years ago
  55. 8c4135e Support for simulating multiple independent flows in a network. by stefan@webrtc.org · 10 years ago
  56. 0a5fd54 Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
  57. 85d90de Returns a NULL frame on all platforms if the captured window is closed. by jiayl@webrtc.org · 10 years ago
  58. b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  59. 0061d86 * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 10 years ago
  60. ee6695b Add an output capacity parameter to ACMResampler::Resample10Msec() by henrik.lundin@webrtc.org · 10 years ago
  61. fbf2568 Add keyboard channel support to AudioBuffer. by andrew@webrtc.org · 10 years ago
  62. 86e3fa8 Fix the Android compilation (better structure for NetEq test libs) by henrik.lundin@webrtc.org · 10 years ago
  63. dbebc39 Remove TraceCallback use from Call. by pbos@webrtc.org · 10 years ago
  64. 9d0f79f Rename Start/Stop in Video{Send,Receive}Streams. by pbos@webrtc.org · 10 years ago
  65. e846663 Fixing a bug in ACM2 where the output frame energy was incorrectly set by henrik.lundin@webrtc.org · 10 years ago
  66. 757a92f Use unique filenames in AudioProcessingTests for parallelization. by andrew@webrtc.org · 10 years ago
  67. e1b0595 AEC: Adds a reported_delay_enabled_ flag by bjornv@webrtc.org · 10 years ago
  68. 110a2d2 Remove ACM1/ACM2 switching from VoiceEngine tests by henrik.lundin@webrtc.org · 10 years ago
  69. 3ab5093 Restore sample_rate_hz() until Chromium is updated to not use it. by andrew@webrtc.org · 10 years ago
  70. 2e24460 Support arbitrary input/output rates and downmixing in AudioProcessing. by andrew@webrtc.org · 10 years ago
  71. 8b4811b Reland "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  72. 79a6030 Remove 44.1 kHz workaround from the iOS AudioDevice. by andrew@webrtc.org · 10 years ago
  73. 0f437b0 Fix a bug in AcmReceiver::NetworkStatistics by henrik.lundin@webrtc.org · 10 years ago
  74. a19bee3 Revert "Stop using ACM factory in VoiceEngine" by henrik.lundin@webrtc.org · 10 years ago
  75. a61127d Stop using ACM factory in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
  76. 69b14d5 Let A/V sync test use default AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  77. 68bd1f3 Create ACM2 instance when calling AudioCodingModule::Create by henrik.lundin@webrtc.org · 10 years ago
  78. 13f9d37 Reland "Make VoiceEngine choose ACM2 by default"" by henrik.lundin@webrtc.org · 10 years ago
  79. 17d096a audio_processing: DestroyHandle() now returns void by bjornv@webrtc.org · 10 years ago
  80. fb54df6 common_audio: VADFree() now returns void by bjornv@webrtc.org · 10 years ago
  81. cf526f7 Resampler modifications in preparation for arbitrary audioproc rates. by andrew@webrtc.org · 10 years ago
  82. 11720c2 Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 10 years ago
  83. 5fd5020 Revert r5937 "Fix multi-monitor support in the screen capturer for Mac." by sergeyu@chromium.org · 10 years ago
  84. a4fbfd9 Add Chromium's ScopedVector. by andrew@webrtc.org · 10 years ago
  85. a73081a Fix multi-monitor support in the screen capturer for Mac. by sergeyu@chromium.org · 10 years ago
  86. bc6b15d Fix iSAC/48000 issue with ACM2. by turaj@webrtc.org · 10 years ago
  87. 499ee5e WebRtcAecm_Process: Reduce code duplication by kwiberg@webrtc.org · 10 years ago
  88. 2991a30 StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16 by kwiberg@webrtc.org · 10 years ago
  89. 514abde Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  90. 3ea24b2 Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  91. 14c5e8a Revert "Make VoiceEngine choose ACM2 by default" by henrik.lundin@webrtc.org · 10 years ago
  92. 722cd19 Removing AudioCoding duplicate tests by henrik.lundin@webrtc.org · 10 years ago
  93. 4f9c08f Make VoiceEngine choose ACM2 by default by henrik.lundin@webrtc.org · 10 years ago
  94. db4b867 Fix crashes due to dangling external decoder pointer. by fischman@webrtc.org · 10 years ago
  95. 988e753 Set include_internal_video_capture=1 for video_capture_tests by kjellander@webrtc.org · 10 years ago
  96. 1857d7e Re-enable AGC tests: by aluebs@webrtc.org · 10 years ago
  97. 32e7755 Remove use of tmpnam. by kjellander@webrtc.org · 10 years ago
  98. 566af28 Replace flooding logs in rtp_sender.cc with a comment. by andrew@webrtc.org · 10 years ago
  99. a738ae3 iOS: baby steps to being able to include_tests=1 by fischman@webrtc.org · 10 years ago
  100. 0b559b6 Moved voe_neteq_stats_unittest to audio_coding_module_unittest by henrik.lundin@webrtc.org · 10 years ago