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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b4bc1a60992b0811edcf1aae780dcd1e79b7e948
  1. b4bc1a6 Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 10 years ago
  2. 688af8b common_audio: Removes unused macros by bjornv@webrtc.org · 10 years ago
  3. 15fc7b8 Re-enable almost all NetEqDecodingTests for Android by henrik.lundin@webrtc.org · 10 years ago
  4. 1a22e41 WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process. by braveyao@webrtc.org · 10 years ago
  5. 75364e4 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log. by wu@webrtc.org · 10 years ago
  6. 5ec8fee Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 10 years ago
  7. 33766c6 Added namespace rtc to some base classes and functions. It was causing linker error in the FYI bots: http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Android%20Builder%20%28dbg%29/builds/1808/steps/compile/logs/stdio but also, not doing it pollutes the global namespace. by henrike@webrtc.org · 10 years ago
  8. 2a8fb71 Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. by wu@webrtc.org · 10 years ago
  9. 1bc4fa6 Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
  10. 11fa357 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  11. 7ef6fff Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  12. a2d1993 Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  13. 1a99e7a Re-enable NetEqExternalDecoderTest for Android by henrik.lundin@webrtc.org · 10 years ago
  14. 8334eda Re-enable NetEQ DecoderDatabase test for Android by henrik.lundin@webrtc.org · 10 years ago
  15. 409cf2a Revert "Audio processing: Feed each processing step its choice of int or float data" by mflodman@webrtc.org · 10 years ago
  16. 654bd9e Re-enable the BitrateEstimatorTest cases for the Call API. by solenberg@webrtc.org · 10 years ago
  17. 2248763 Remove all use of AudioFrame::energy_ from AudioCodingModule by henrik.lundin@webrtc.org · 10 years ago
  18. 4ee3791 VoEVolumeTest: Adds error return tests. by bjornv@webrtc.org · 10 years ago
  19. 6047281 Audio processing: Feed each processing step its choice of int or float data by kwiberg@webrtc.org · 10 years ago
  20. 903ce9e Remove WEBRTC_TRACE use in video_capture/ by pbos@webrtc.org · 10 years ago
  21. 04a721c Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 10 years ago
  22. 9754a5d Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 10 years ago
  23. f7795df Adds a modified copy of talk/base to webrtc/base. It is the first step in by henrike@webrtc.org · 10 years ago
  24. 7ea3607 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 by bjornv@webrtc.org · 10 years ago
  25. fc3a6c0 Revert "FieldTrial implementation for webrtc." (rev 6089) by andresp@webrtc.org · 10 years ago
  26. b16a722 Reduced kMaxSampleDiffMs (limit to 22fps). by asapersson@webrtc.org · 10 years ago
  27. 838c9da Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
  28. 2efcf70 Deleting all NetEq3 files by henrik.lundin@webrtc.org · 10 years ago
  29. cf1f0b0 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
  30. 2661819 Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..." by perkj@webrtc.org · 10 years ago
  31. 4a792f0 Deleting all ACM1 files by henrik.lundin@webrtc.org · 10 years ago
  32. a229768 Fix failing test introduced with r6111. by stefan@webrtc.org · 10 years ago
  33. 305fd94 Fixes log spam introduced with r6041. by stefan@webrtc.org · 10 years ago
  34. b9c8d1a 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
  35. bb1e3ff 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
  36. 8773fa6 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
  37. 9c8f347 Echo cancellation functions docs: Follow style guide w.r.t. placement of * by kwiberg@webrtc.org · 10 years ago
  38. b0295bf 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
  39. aa169d2 One of the NetEq methods needs to be virtual. by turaj@webrtc.org · 10 years ago
  40. da9b404 Modifying neteq.gyp by turaj@webrtc.org · 10 years ago
  41. 3a87cff Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 10 years ago
  42. 28e9b66 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 10 years ago
  43. 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
  44. d12d7b6 FieldTrial implementation for webrtc. by andresp@webrtc.org · 10 years ago
  45. 0d47fe1 Raise kViEMaxNumberOfChannels from 32 to 64 by wu@webrtc.org · 10 years ago
  46. 569487d Updated WebRTC version to 3.53 TBR=wu@webrtc.org by elham@webrtc.org · 10 years ago
  47. 8b4f539 AudioBuffer: Eliminate data_was_mixed_, and document what's left of data_ by kwiberg@webrtc.org · 10 years ago
  48. 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
  49. 8539c4a Fix odd codes in video_capture on Mac. by braveyao@webrtc.org · 10 years ago
  50. 4fb1a55 video_render.gypi: clean up some libraries directives to be more specific. by fischman@webrtc.org · 10 years ago
  51. 73c2412 Remove timestamp_extrapolator's dependency to Clock and vcm defines. by wu@webrtc.org · 10 years ago
  52. 8ec46c6 Allow the RTP level indicator computation to work at any sample rate. by andrew@webrtc.org · 10 years ago
  53. 96cccf7 Remove ALLOW_UNUSED. by andrew@webrtc.org · 10 years ago
  54. 5e44f56 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. by wu@webrtc.org · 10 years ago
  55. ebb4b94 Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  56. 7c434be Revert 6048 "Implement the Windows screen capturer using the Mag..." by tina.legrand@webrtc.org · 10 years ago
  57. 6ccd081 WebRTCDemo: correct set trace filter operation. by braveyao@webrtc.org · 10 years ago
  58. 5cc0d0b Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest. by andrew@webrtc.org · 10 years ago
  59. c2b27b5 Implement the Windows screen capturer using the Magnification API. by jiayl@webrtc.org · 10 years ago
  60. 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
  61. 1a9e6ac Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  62. 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
  63. 616cbcd Only clamp to 16 kHz when AECM is enabled. by andrew@webrtc.org · 10 years ago
  64. c2e6438 Fix constness of AudioBuffer accessors. by andrew@webrtc.org · 10 years ago
  65. 0638464 Fix a data race in ACM1 when audio is pulled. by turaj@webrtc.org · 10 years ago
  66. 976ce98 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 10 years ago
  67. 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
  68. 40b200b Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() by henrike@webrtc.org · 10 years ago
  69. 3d5905b Disable failing GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
  70. d592231 Disable GoogleWifiTrace3Mbps. by pbos@webrtc.org · 10 years ago
  71. 3848107 Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios. by stefan@webrtc.org · 10 years ago
  72. c6cfc5c Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
  73. c78232f Fix iOS assembly compile error. by kjellander@webrtc.org · 10 years ago
  74. c523211 Remove neteq_unittests from Android builds by henrik.lundin@webrtc.org · 10 years ago
  75. ad4cce6 Roll chromium_revision 260462:266514 by kjellander@webrtc.org · 10 years ago
  76. 3cbb2df Remove Version method from ACM1 by henrik.lundin@webrtc.org · 10 years ago
  77. dc37088 Remove ACM1 and NetEq3 related targets from modules.gyp by henrik.lundin@webrtc.org · 10 years ago
  78. 68a95e1 Remove AudioCodingModuleFactory by henrik.lundin@webrtc.org · 10 years ago
  79. a48f3c2 Add clock to ACM config struct by henrik.lundin@webrtc.org · 10 years ago
  80. db395e4 AEC: Startup phase only runs if reported_delay_enabled by bjornv@webrtc.org · 10 years ago
  81. be039c2 Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash. by fischman@webrtc.org · 10 years ago
  82. b4945d1 APM: limit native sample rate to 16kHz on mobile. by fischman@webrtc.org · 10 years ago
  83. 93d270f Using realpath instead of android_src in Android webview by michaelbai@google.com · 10 years ago
  84. a2d989b Only download the VS toolchain if DEPOT_TOOLS_WIN_TOOLCHAIN=1. by andrew@webrtc.org · 10 years ago
  85. c54ff69 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  86. 267637b Disable flaky CaptureNtpTimeWithNetworkJitter. by pbos@webrtc.org · 10 years ago
  87. 0e098e0 AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h by bjornv@webrtc.org · 10 years ago
  88. 676638c Disable capture test for FrameRate on Windows. by pbos@webrtc.org · 10 years ago
  89. bb62a93 Introduce a config struct for AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  90. c298835 Disabling flaky CanReceiveFec. by pbos@webrtc.org · 10 years ago
  91. 73e1a8b Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
  92. abf78cc Fix the NetEq build by henrik.lundin@webrtc.org · 10 years ago
  93. 75d1487 Include buffer size limits in NetEq config struct by henrik.lundin@webrtc.org · 10 years ago
  94. a714643 Add henrik.lundin as owner in AudioCoding module by henrik.lundin@webrtc.org · 10 years ago
  95. b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  96. 4820f6b Fix leak in remote bitrate estimator tests introduced in r5980 by stefan@webrtc.org · 10 years ago
  97. 8c4135e Support for simulating multiple independent flows in a network. by stefan@webrtc.org · 10 years ago
  98. 0a5fd54 Casting char to int in logs. by asapersson@webrtc.org · 10 years ago
  99. 85d90de Returns a NULL frame on all platforms if the captured window is closed. by jiayl@webrtc.org · 10 years ago
  100. b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago