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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 0e9c399746f45ceaf46f12b11ba93c09cca0c2bb
  1. 0e9c399 NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  2. 24f0702 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  3. d4e1329 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
  4. 76a6ffb Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
  5. 0d4d51b Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
  6. 76238f6 Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
  7. 0de0049 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  8. cd5c882 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  9. 2b35b95 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  10. a3a3a0f Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
  11. 9e035d2 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  12. b503d1e Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  13. 362e3e5 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  14. 424e0e4 With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  15. cdc5e6a Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
  16. b655adf Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  17. 44f030c Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
  18. dc1f7e9 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago
  19. ee817d3 Remove include_dirs from remote_bitrate_estimator. by pbos@webrtc.org · 11 years ago
  20. fc75214 Remove include_dirs from bitrate_controller. by pbos@webrtc.org · 11 years ago
  21. 1fc4659 Remove include_dirs from video_coding. by pbos@webrtc.org · 11 years ago
  22. 85592ad Remove include_dirs from video_processing. by pbos@webrtc.org · 11 years ago
  23. 1800406 Remove include_dirs from rtp_rtcp. by pbos@webrtc.org · 11 years ago
  24. 2f0a942 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. by turaj@webrtc.org · 11 years ago
  25. d4f6789 Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 11 years ago
  26. 8ddec2c Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  27. 53fdd3b Fix WindowCapturerWin to capture window decorations after window size changes. by sergeyu@chromium.org · 11 years ago
  28. 605daf0 Disable a NetEq unittest on Android. The test tries to register iSAC-swb as send codec and fails. by turaj@webrtc.org · 11 years ago
  29. 72790c7 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
  30. f7d5a08 Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org by elham@webrtc.org · 11 years ago
  31. 7f35836 Re-enable verbose logging in NetEq4. by turaj@webrtc.org · 11 years ago
  32. 79c884c Convert DeviceInfoImpl::_captureCapabilities from a map to a vector. by fischman@webrtc.org · 11 years ago
  33. 99b6d9e Revert 4837 "Add an extended filter mode to AEC." by asapersson@webrtc.org · 11 years ago
  34. 83c5f62 Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  35. 933267f Small fixes to run ACM2 tests. by turaj@webrtc.org · 11 years ago
  36. 6ca9e7d API add to set background noise mode. by turaj@webrtc.org · 11 years ago
  37. 08099e0 Fix window capturer not to leak HDC. by sergeyu@chromium.org · 11 years ago
  38. 82707bf Fix window capturer to stop capturing when the target is minimized. by sergeyu@chromium.org · 11 years ago
  39. 4b067da Disable some VP8 tests on Android. by andrew@webrtc.org · 11 years ago
  40. 8da2f65 Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
  41. 3bd659f Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
  42. a89f7e8 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
  43. 890706b Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
  44. da6d2a2 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  45. b0382ea Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
  46. 510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  47. ae14504 - Reset capture deltas at resolution change. by asapersson@webrtc.org · 11 years ago
  48. a6665e7 Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 11 years ago
  49. 36441e3 Re-enable VideoCaptureTest.CreateDelete by fischman@webrtc.org · 11 years ago
  50. 3b6d2d4 Updated WebRTC version to 3.42 by elham@webrtc.org · 11 years ago
  51. 84afa19 Adding unit tests for default temporal layer strategy. by andresp@webrtc.org · 11 years ago
  52. 199555c Revert test change in r4808. by stefan@webrtc.org · 11 years ago
  53. d704640 Reduce flakiness in network down test. by stefan@webrtc.org · 11 years ago
  54. 2529558 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  55. d1fe828 Fix bugs in DesktopRegion::Subtract(). by sergeyu@chromium.org · 11 years ago
  56. 717267a VAD changes ported to ACM2. by turaj@webrtc.org · 11 years ago
  57. 045e45e Address Windows 64-bits warnings. by turaj@webrtc.org · 11 years ago
  58. 0011252 Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
  59. 54f0246 Disable flaky video capture test. by stefan@webrtc.org · 11 years ago
  60. 51d53aa Avoid recursively taking critical section. by stefan@webrtc.org · 11 years ago
  61. 7ab577d Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 11 years ago
  62. 6876512 Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 11 years ago
  63. 28a1166 Rename EngineTest to CallTest. by pbos@webrtc.org · 11 years ago
  64. f5013c0 Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct by tina.legrand@webrtc.org · 11 years ago
  65. 28631e7 Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 11 years ago
  66. a89566f Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 11 years ago
  67. 93b9912 Fixes a flake in network down tests. by stefan@webrtc.org · 11 years ago
  68. 032f731 Disable tests for TSan v2 by kjellander@webrtc.org · 11 years ago
  69. 4d08199 Compile ACM2 and ACM1. by turaj@webrtc.org · 11 years ago
  70. 80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  71. ab34f11 NetEq4: Making a few more members scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  72. 05dd6c0 Dedicated speed test for NetEq3 by henrik.lundin@webrtc.org · 11 years ago
  73. c61a170 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
  74. ec09fcb Revert r4772 "Compile ACM1 and ACM2." by stefan@webrtc.org · 11 years ago
  75. 671d90b NetEq4: Make some DSP operation classes member variables by henrik.lundin@webrtc.org · 11 years ago
  76. c2c8e6a Fix races in vcm::Process(). by stefan@webrtc.org · 11 years ago
  77. 1ddd57f Break out glue for old->new Transport. by pbos@webrtc.org · 11 years ago
  78. 5b7878f Changing 'frame' method to 'bounds' method. by sjlee@webrtc.org · 11 years ago
  79. 7556d2d Compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  80. 0c57671 Use the native sample rate for OpenSL recording. by henrike@webrtc.org · 11 years ago
  81. 39e22a1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
  82. 0277aa4 Fix typo in r4765. by pbos@webrtc.org · 11 years ago
  83. 54bc776 Fix dangling pointer _encoder in video_sender.cc. by pbos@webrtc.org · 11 years ago
  84. 64b5c61 Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 11 years ago
  85. 79d3355 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  86. 7485573 Add support for multiple report blocks. by stefan@webrtc.org · 11 years ago
  87. e9d2898 This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 by sjlee@webrtc.org · 11 years ago
  88. e3a12da This is related to https://code.google.com/p/webrtc/issues/detail?id=846 by sjlee@webrtc.org · 11 years ago
  89. d8a5b00 To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 11 years ago
  90. b0fb1d6 Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 11 years ago
  91. e8fdc9d Split video coding module unit tests into sender and receiver unit tests. by andresp@webrtc.org · 11 years ago
  92. 041d54b Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
  93. 36c3652 Remove use of vcm->ResetDecoder from modules/utility. by andresp@webrtc.org · 11 years ago
  94. a4bbaa6 Allocate float_buffer_ in the initializer list. by andrew@webrtc.org · 11 years ago
  95. 42a65a2 Split VideoCodingModuleImpl into VideoSender and VideoReceiver. by andresp@webrtc.org · 11 years ago
  96. ed0b4fb Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  97. 26251da Implement DesktopRegion subtraction. by sergeyu@chromium.org · 11 years ago
  98. a26a7f6 Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 11 years ago
  99. 388d16c Fix win trybot errors due to r4729. by andrew@webrtc.org · 11 years ago
  100. d0737d9 Fix crash in the window capturer on windows by sergeyu@chromium.org · 11 years ago