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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 88bcc9819bd6dade51ae7d8f7f32ff2cb16f66bf
  1. 88bcc98 Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
  2. 957be53 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
  3. 7e07f16 Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
  4. 48b1173 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  5. 653dfe1 Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
  6. d24ce00 Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  7. 5cca7ed - Reset capture deltas at resolution change. by asapersson@webrtc.org · 11 years ago
  8. 72fde7b Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 11 years ago
  9. 584890b Re-enable VideoCaptureTest.CreateDelete by fischman@webrtc.org · 11 years ago
  10. 0715778 Updated WebRTC version to 3.42 by elham@webrtc.org · 11 years ago
  11. 697b7f3 Adding unit tests for default temporal layer strategy. by andresp@webrtc.org · 11 years ago
  12. 21318a9 Revert test change in r4808. by stefan@webrtc.org · 11 years ago
  13. dd3f2e4 Reduce flakiness in network down test. by stefan@webrtc.org · 11 years ago
  14. 0580c2c Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  15. 26caab2 Fix bugs in DesktopRegion::Subtract(). by sergeyu@chromium.org · 11 years ago
  16. 69dfcb4 VAD changes ported to ACM2. by turaj@webrtc.org · 11 years ago
  17. 344a2d7 Address Windows 64-bits warnings. by turaj@webrtc.org · 11 years ago
  18. aa693dd Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
  19. 8797e63 Disable flaky video capture test. by stefan@webrtc.org · 11 years ago
  20. 0d416cf Avoid recursively taking critical section. by stefan@webrtc.org · 11 years ago
  21. 0c6a78c Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 11 years ago
  22. 927bbc2 Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 11 years ago
  23. 60fa827 Rename EngineTest to CallTest. by pbos@webrtc.org · 11 years ago
  24. 14f44ba Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct by tina.legrand@webrtc.org · 11 years ago
  25. 9612f5a Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 11 years ago
  26. 97e2f4e Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 11 years ago
  27. 71c8df6 Fixes a flake in network down tests. by stefan@webrtc.org · 11 years ago
  28. 4bb3362 Disable tests for TSan v2 by kjellander@webrtc.org · 11 years ago
  29. edf08ee Compile ACM2 and ACM1. by turaj@webrtc.org · 11 years ago
  30. bee99b1 Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  31. c12119c NetEq4: Making a few more members scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  32. b00b61d Dedicated speed test for NetEq3 by henrik.lundin@webrtc.org · 11 years ago
  33. fccf64c MIPS optimizations for the functions WebRtcSpl_SqrtFloor, WebRtcSpl_CrossCorrelation, WebRtcSpl_ScaleAndAddVectorsWithRound and the inline functions from signal_processing spl_inl.h file. by andrew@webrtc.org · 11 years ago
  34. 65a237a Revert r4772 "Compile ACM1 and ACM2." by stefan@webrtc.org · 11 years ago
  35. 7c41c3b NetEq4: Make some DSP operation classes member variables by henrik.lundin@webrtc.org · 11 years ago
  36. 3f9ebdb Fix races in vcm::Process(). by stefan@webrtc.org · 11 years ago
  37. 26d75f3 Break out glue for old->new Transport. by pbos@webrtc.org · 11 years ago
  38. cfdf698 Changing 'frame' method to 'bounds' method. by sjlee@webrtc.org · 11 years ago
  39. 19c663b Compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  40. 0ae4638 Use the native sample rate for OpenSL recording. by henrike@webrtc.org · 11 years ago
  41. e30fde1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  42. 7901868 Fix typo in r4765. by pbos@webrtc.org · 11 years ago
  43. 5777a0a Fix dangling pointer _encoder in video_sender.cc. by pbos@webrtc.org · 11 years ago
  44. c9b400c Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 11 years ago
  45. ed8ce36 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  46. 0c0f882 Add support for multiple report blocks. by stefan@webrtc.org · 11 years ago
  47. bc90ee3 This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 by sjlee@webrtc.org · 11 years ago
  48. 1a8c9b3 This is related to https://code.google.com/p/webrtc/issues/detail?id=846 by sjlee@webrtc.org · 11 years ago
  49. 29fce82 To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 11 years ago
  50. e8eaed8 Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 11 years ago
  51. 0180fc4 Split video coding module unit tests into sender and receiver unit tests. by andresp@webrtc.org · 11 years ago
  52. f952fce Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
  53. 3b6ab4a Remove use of vcm->ResetDecoder from modules/utility. by andresp@webrtc.org · 11 years ago
  54. 91b0d23 Allocate float_buffer_ in the initializer list. by andrew@webrtc.org · 11 years ago
  55. f458c43 Split VideoCodingModuleImpl into VideoSender and VideoReceiver. by andresp@webrtc.org · 11 years ago
  56. e125ca7 Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  57. cda8e61 Implement DesktopRegion subtraction. by sergeyu@chromium.org · 11 years ago
  58. 564ba1e Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 11 years ago
  59. 1d8ceab Fix win trybot errors due to r4729. by andrew@webrtc.org · 11 years ago
  60. 985848d Fix crash in the window capturer on windows by sergeyu@chromium.org · 11 years ago
  61. 242b8a5 ACM2 integration with NetEq 4. by turaj@webrtc.org · 11 years ago
  62. 54164d5 Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 11 years ago
  63. 4d57e48 The video render module for iOS. by fischman@webrtc.org · 11 years ago
  64. 1963a68 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  65. a5b7b8c Make PCM16 available in Chromium builds. by andrew@webrtc.org · 11 years ago
  66. 40bd492 Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 11 years ago
  67. 93da8cb Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 11 years ago
  68. e45a8a8 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 11 years ago
  69. 9d775a6 Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. by henrike@webrtc.org · 11 years ago
  70. e22b761 Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  71. 75e7cff OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 11 years ago
  72. 811e4c9 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 11 years ago
  73. 7efd262 Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 11 years ago
  74. 905cebd Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  75. 910520a Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 11 years ago
  76. 96da891 Remove repeated conditions key. by andrew@webrtc.org · 11 years ago
  77. 3965d1f OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  78. 4c94668 Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 11 years ago
  79. d1deeb6 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
  80. af73083 Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 11 years ago
  81. b1b278e Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 11 years ago
  82. 0a477d1 Lock RTPSender statistics. by pbos@webrtc.org · 11 years ago
  83. 4d1cb14 Split up EngineTests and RampupTests. by pbos@webrtc.org · 11 years ago
  84. 81c4d24 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
  85. 98691c2 Updated WebRTC version to 3.41 by elham@webrtc.org · 11 years ago
  86. 0313e5b Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 11 years ago
  87. 462460f Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 11 years ago
  88. 38ba534 Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 11 years ago
  89. fdc4352 Rename VideoCall to Call. by pbos@webrtc.org · 11 years ago
  90. 9c843fd Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 11 years ago
  91. 0ee03f9 ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
  92. b8aa042 Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 11 years ago
  93. d44ec1c Reduce cost of PushSincResampler::Resample(). by andrew@webrtc.org · 11 years ago
  94. 252b16f Clamp camera id to legal values. by fischman@webrtc.org · 11 years ago
  95. 5ee7139 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
  96. 5632a64 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  97. b49897c Add temporal layer factory. by andresp@webrtc.org · 11 years ago
  98. 4a4d15b Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
  99. 1e88712 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 11 years ago
  100. 5a196e6 Pre-multiply images for MouseCursorShape. by alexeypa@chromium.org · 11 years ago