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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 0de0049a4e555f347cab027c2fddd32e7f916ed3 / modules
  1. cd5c882 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  2. 2b35b95 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  3. 9e035d2 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  4. b503d1e Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  5. 424e0e4 With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  6. 44f030c Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
  7. dc1f7e9 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago
  8. ee817d3 Remove include_dirs from remote_bitrate_estimator. by pbos@webrtc.org · 11 years ago
  9. fc75214 Remove include_dirs from bitrate_controller. by pbos@webrtc.org · 11 years ago
  10. 1fc4659 Remove include_dirs from video_coding. by pbos@webrtc.org · 11 years ago
  11. 85592ad Remove include_dirs from video_processing. by pbos@webrtc.org · 11 years ago
  12. 1800406 Remove include_dirs from rtp_rtcp. by pbos@webrtc.org · 11 years ago
  13. 2f0a942 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. by turaj@webrtc.org · 11 years ago
  14. d4f6789 Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 11 years ago
  15. 8ddec2c Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  16. 53fdd3b Fix WindowCapturerWin to capture window decorations after window size changes. by sergeyu@chromium.org · 11 years ago
  17. 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
  18. 72790c7 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
  19. 7f35836 Re-enable verbose logging in NetEq4. by turaj@webrtc.org · 11 years ago
  20. 79c884c Convert DeviceInfoImpl::_captureCapabilities from a map to a vector. by fischman@webrtc.org · 11 years ago
  21. 99b6d9e Revert 4837 "Add an extended filter mode to AEC." by asapersson@webrtc.org · 11 years ago
  22. 83c5f62 Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  23. 933267f Small fixes to run ACM2 tests. by turaj@webrtc.org · 11 years ago
  24. 6ca9e7d API add to set background noise mode. by turaj@webrtc.org · 11 years ago
  25. 08099e0 Fix window capturer not to leak HDC. by sergeyu@chromium.org · 11 years ago
  26. 82707bf Fix window capturer to stop capturing when the target is minimized. by sergeyu@chromium.org · 11 years ago
  27. 4b067da Disable some VP8 tests on Android. by andrew@webrtc.org · 11 years ago
  28. da6d2a2 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  29. 510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  30. a6665e7 Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 11 years ago
  31. 36441e3 Re-enable VideoCaptureTest.CreateDelete by fischman@webrtc.org · 11 years ago
  32. 84afa19 Adding unit tests for default temporal layer strategy. by andresp@webrtc.org · 11 years ago
  33. 2529558 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  34. d1fe828 Fix bugs in DesktopRegion::Subtract(). by sergeyu@chromium.org · 11 years ago
  35. 717267a VAD changes ported to ACM2. by turaj@webrtc.org · 11 years ago
  36. 045e45e Address Windows 64-bits warnings. by turaj@webrtc.org · 11 years ago
  37. 54f0246 Disable flaky video capture test. by stefan@webrtc.org · 11 years ago
  38. 51d53aa Avoid recursively taking critical section. by stefan@webrtc.org · 11 years ago
  39. 7ab577d Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 11 years ago
  40. 6876512 Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 11 years ago
  41. f5013c0 Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct by tina.legrand@webrtc.org · 11 years ago
  42. 4d08199 Compile ACM2 and ACM1. by turaj@webrtc.org · 11 years ago
  43. 80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
  44. ab34f11 NetEq4: Making a few more members scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  45. 05dd6c0 Dedicated speed test for NetEq3 by henrik.lundin@webrtc.org · 11 years ago
  46. ec09fcb Revert r4772 "Compile ACM1 and ACM2." by stefan@webrtc.org · 11 years ago
  47. 671d90b NetEq4: Make some DSP operation classes member variables by henrik.lundin@webrtc.org · 11 years ago
  48. c2c8e6a Fix races in vcm::Process(). by stefan@webrtc.org · 11 years ago
  49. 5b7878f Changing 'frame' method to 'bounds' method. by sjlee@webrtc.org · 11 years ago
  50. 7556d2d Compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  51. 0c57671 Use the native sample rate for OpenSL recording. by henrike@webrtc.org · 11 years ago
  52. 0277aa4 Fix typo in r4765. by pbos@webrtc.org · 11 years ago
  53. 54bc776 Fix dangling pointer _encoder in video_sender.cc. by pbos@webrtc.org · 11 years ago
  54. 64b5c61 Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 11 years ago
  55. 79d3355 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 11 years ago
  56. 7485573 Add support for multiple report blocks. by stefan@webrtc.org · 11 years ago
  57. e9d2898 This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 by sjlee@webrtc.org · 11 years ago
  58. e3a12da This is related to https://code.google.com/p/webrtc/issues/detail?id=846 by sjlee@webrtc.org · 11 years ago
  59. e8fdc9d Split video coding module unit tests into sender and receiver unit tests. by andresp@webrtc.org · 11 years ago
  60. 36c3652 Remove use of vcm->ResetDecoder from modules/utility. by andresp@webrtc.org · 11 years ago
  61. 42a65a2 Split VideoCodingModuleImpl into VideoSender and VideoReceiver. by andresp@webrtc.org · 11 years ago
  62. ed0b4fb Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  63. 26251da Implement DesktopRegion subtraction. by sergeyu@chromium.org · 11 years ago
  64. a26a7f6 Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 11 years ago
  65. 388d16c Fix win trybot errors due to r4729. by andrew@webrtc.org · 11 years ago
  66. d0737d9 Fix crash in the window capturer on windows by sergeyu@chromium.org · 11 years ago
  67. 3f39c00 ACM2 integration with NetEq 4. by turaj@webrtc.org · 11 years ago
  68. a3351c4 Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 11 years ago
  69. bc375b5 The video render module for iOS. by fischman@webrtc.org · 11 years ago
  70. 4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
  71. 5e3379e Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 11 years ago
  72. 0fd885e Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 11 years ago
  73. f5556f2 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 11 years ago
  74. 9fea95a 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
  75. 8fdce8e OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 11 years ago
  76. 66dbbd9 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 11 years ago
  77. f2982c9 Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 11 years ago
  78. f0adedc Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 11 years ago
  79. 054bc03 Remove repeated conditions key. by andrew@webrtc.org · 11 years ago
  80. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  81. b676ac7 Lock RTPSender statistics. by pbos@webrtc.org · 11 years ago
  82. 5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
  83. 6b4698e Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 11 years ago
  84. 4e7777b Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 11 years ago
  85. 6a79c9f Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 11 years ago
  86. 618a0ec ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
  87. e97b69f Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 11 years ago
  88. 7dc1790 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
  89. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  90. 4014302 Add temporal layer factory. by andresp@webrtc.org · 11 years ago
  91. 31a8ce7 Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
  92. f2ef20c Pre-multiply images for MouseCursorShape. by alexeypa@chromium.org · 11 years ago
  93. 6f458ed Recognize armv7 target_arch for ios support in webrtc common.gyp by fischman@webrtc.org · 11 years ago
  94. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  95. 0e2cb29 Add clockdrift to RtpGenerator by henrik.lundin@webrtc.org · 11 years ago
  96. 787364c NetEq4: Small change to reduce allocs in AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  97. 7a21a64 Clean capture timestamp code. by andresp@webrtc.org · 11 years ago
  98. 00c95bf Protecting Bitrate to avoid data race found by tsan. by mflodman@webrtc.org · 11 years ago
  99. 0f62690 Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
  100. 0fe8944 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago