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webrtc / src / webrtc / 933267f0b6fddf3fe62b5f990188da5980bca984
  1. 933267f Small fixes to run ACM2 tests. by turaj@webrtc.org · 12 years ago
  2. 6ca9e7d API add to set background noise mode. by turaj@webrtc.org · 12 years ago
  3. 08099e0 Fix window capturer not to leak HDC. by sergeyu@chromium.org · 12 years ago
  4. 82707bf Fix window capturer to stop capturing when the target is minimized. by sergeyu@chromium.org · 12 years ago
  5. 4b067da Disable some VP8 tests on Android. by andrew@webrtc.org · 12 years ago
  6. 8da2f65 Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 12 years ago
  7. 3bd659f Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 12 years ago
  8. a89f7e8 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 12 years ago
  9. 890706b Reenable test and remove flaky expects. by stefan@webrtc.org · 12 years ago
  10. da6d2a2 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 12 years ago
  11. b0382ea Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 12 years ago
  12. 510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 12 years ago
  13. ae14504 - Reset capture deltas at resolution change. by asapersson@webrtc.org · 12 years ago
  14. a6665e7 Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 12 years ago
  15. 36441e3 Re-enable VideoCaptureTest.CreateDelete by fischman@webrtc.org · 12 years ago
  16. 3b6d2d4 Updated WebRTC version to 3.42 by elham@webrtc.org · 12 years ago
  17. 84afa19 Adding unit tests for default temporal layer strategy. by andresp@webrtc.org · 12 years ago
  18. 199555c Revert test change in r4808. by stefan@webrtc.org · 12 years ago
  19. d704640 Reduce flakiness in network down test. by stefan@webrtc.org · 12 years ago
  20. 2529558 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 12 years ago
  21. d1fe828 Fix bugs in DesktopRegion::Subtract(). by sergeyu@chromium.org · 12 years ago
  22. 717267a VAD changes ported to ACM2. by turaj@webrtc.org · 12 years ago
  23. 045e45e Address Windows 64-bits warnings. by turaj@webrtc.org · 12 years ago
  24. 0011252 Enable FEC for VideoSendStream. by pbos@webrtc.org · 12 years ago
  25. 54f0246 Disable flaky video capture test. by stefan@webrtc.org · 12 years ago
  26. 51d53aa Avoid recursively taking critical section. by stefan@webrtc.org · 12 years ago
  27. 7ab577d Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 12 years ago
  28. 6876512 Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 12 years ago
  29. 28a1166 Rename EngineTest to CallTest. by pbos@webrtc.org · 12 years ago
  30. f5013c0 Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct by tina.legrand@webrtc.org · 12 years ago
  31. 28631e7 Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 12 years ago
  32. a89566f Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 12 years ago
  33. 93b9912 Fixes a flake in network down tests. by stefan@webrtc.org · 12 years ago
  34. 032f731 Disable tests for TSan v2 by kjellander@webrtc.org · 12 years ago
  35. 4d08199 Compile ACM2 and ACM1. by turaj@webrtc.org · 12 years ago
  36. 80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 12 years ago
  37. ab34f11 NetEq4: Making a few more members scoped_ptrs by henrik.lundin@webrtc.org · 12 years ago
  38. 05dd6c0 Dedicated speed test for NetEq3 by henrik.lundin@webrtc.org · 12 years ago
  39. 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 · 12 years ago
  40. ec09fcb Revert r4772 "Compile ACM1 and ACM2." by stefan@webrtc.org · 12 years ago
  41. 671d90b NetEq4: Make some DSP operation classes member variables by henrik.lundin@webrtc.org · 12 years ago
  42. c2c8e6a Fix races in vcm::Process(). by stefan@webrtc.org · 12 years ago
  43. 1ddd57f Break out glue for old->new Transport. by pbos@webrtc.org · 12 years ago
  44. 5b7878f Changing 'frame' method to 'bounds' method. by sjlee@webrtc.org · 12 years ago
  45. 7556d2d Compile ACM1 and ACM2. by turaj@webrtc.org · 12 years ago
  46. 0c57671 Use the native sample rate for OpenSL recording. by henrike@webrtc.org · 12 years ago
  47. 39e22a1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 12 years ago
  48. 0277aa4 Fix typo in r4765. by pbos@webrtc.org · 12 years ago
  49. 54bc776 Fix dangling pointer _encoder in video_sender.cc. by pbos@webrtc.org · 12 years ago
  50. 64b5c61 Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 12 years ago
  51. 79d3355 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 12 years ago
  52. 7485573 Add support for multiple report blocks. by stefan@webrtc.org · 12 years ago
  53. e9d2898 This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 by sjlee@webrtc.org · 12 years ago
  54. e3a12da This is related to https://code.google.com/p/webrtc/issues/detail?id=846 by sjlee@webrtc.org · 12 years ago
  55. d8a5b00 To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 12 years ago
  56. b0fb1d6 Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 12 years ago
  57. e8fdc9d Split video coding module unit tests into sender and receiver unit tests. by andresp@webrtc.org · 12 years ago
  58. 041d54b Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 12 years ago
  59. 36c3652 Remove use of vcm->ResetDecoder from modules/utility. by andresp@webrtc.org · 12 years ago
  60. a4bbaa6 Allocate float_buffer_ in the initializer list. by andrew@webrtc.org · 12 years ago
  61. 42a65a2 Split VideoCodingModuleImpl into VideoSender and VideoReceiver. by andresp@webrtc.org · 12 years ago
  62. ed0b4fb Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 12 years ago
  63. 26251da Implement DesktopRegion subtraction. by sergeyu@chromium.org · 12 years ago
  64. a26a7f6 Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 12 years ago
  65. 388d16c Fix win trybot errors due to r4729. by andrew@webrtc.org · 12 years ago
  66. d0737d9 Fix crash in the window capturer on windows by sergeyu@chromium.org · 12 years ago
  67. 3f39c00 ACM2 integration with NetEq 4. by turaj@webrtc.org · 12 years ago
  68. a3351c4 Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 12 years ago
  69. bc375b5 The video render module for iOS. by fischman@webrtc.org · 12 years ago
  70. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 12 years ago
  71. 66bfae2 Make PCM16 available in Chromium builds. by andrew@webrtc.org · 12 years ago
  72. 5e3379e Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 12 years ago
  73. 0fd885e Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 12 years ago
  74. f5556f2 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 12 years ago
  75. 9fea95a Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. by henrike@webrtc.org · 12 years ago
  76. bfad17e Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 12 years ago
  77. 8fdce8e OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 12 years ago
  78. 66dbbd9 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 12 years ago
  79. f2982c9 Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 12 years ago
  80. 990c5e3 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 12 years ago
  81. f0adedc Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 12 years ago
  82. 054bc03 Remove repeated conditions key. by andrew@webrtc.org · 12 years ago
  83. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 12 years ago
  84. dadb2a1 Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 12 years ago
  85. 7b30ce3 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 12 years ago
  86. eb2d9dd Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 12 years ago
  87. 3524ade Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 12 years ago
  88. b676ac7 Lock RTPSender statistics. by pbos@webrtc.org · 12 years ago
  89. fa996f2 Split up EngineTests and RampupTests. by pbos@webrtc.org · 12 years ago
  90. 5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 12 years ago
  91. 0920142 Updated WebRTC version to 3.41 by elham@webrtc.org · 12 years ago
  92. 6b4698e Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 12 years ago
  93. 0245bee Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 12 years ago
  94. 4e7777b Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 12 years ago
  95. bf6d572 Rename VideoCall to Call. by pbos@webrtc.org · 12 years ago
  96. 6a79c9f Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 12 years ago
  97. 618a0ec ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 12 years ago
  98. e97b69f Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 12 years ago
  99. 11a8868 Reduce cost of PushSincResampler::Resample(). by andrew@webrtc.org · 12 years ago
  100. ca20f3d Clamp camera id to legal values. by fischman@webrtc.org · 12 years ago