1. 09b40ec Add experimental noise suppression dummy API. by aluebs@webrtc.org · 11 years ago
  2. 8b0791c Fix DesktopAndCursorComposer to restore frames to the original state. by sergeyu@chromium.org · 11 years ago
  3. eb45a20 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. by turaj@webrtc.org · 11 years ago
  4. 4590177 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  5. e028410 Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
  6. 69b1aa4 Increment RTP timestamps for padding packets by henrik.lundin@webrtc.org · 11 years ago
  7. b748c9d Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
  8. 591be3b Reimplementing NetEq4's AudioVector by henrik.lundin@webrtc.org · 11 years ago
  9. f4fbef3 Parse next RTCP XR report block after an unsupported block type. by asapersson@webrtc.org · 11 years ago
  10. 26f5492 Reducing opus_test runtime to pass Android test by minyue@webrtc.org · 11 years ago
  11. ff4fc2b MIPS optimizations for AECM audio processing module by andrew@webrtc.org · 11 years ago
  12. 731a87b Move audio_processing dependencies to a variable. by andrew@webrtc.org · 11 years ago
  13. 9965e3a Remove ".." from include_dirs in build/common. by pbos@webrtc.org · 11 years ago
  14. 3051ff7 Remove unnecessary include_dirs from audio_processing. by andrew@webrtc.org · 11 years ago
  15. 7e97e4c Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
  16. b4d7835 Fix for video_processor_intergration_tests to run in parallel. by marpan@webrtc.org · 11 years ago
  17. 93bf70f Add missing dependencies to .isolate files by kjellander@webrtc.org · 11 years ago
  18. 690a03c Fix broken build on x86 Android by fischman@webrtc.org · 11 years ago
  19. 1bd9a7b Removed unused code. by asapersson@webrtc.org · 11 years ago
  20. af92d3e Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  21. a191cb0 Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  22. 6baaf30 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  23. 7773eec Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  24. a9a788e Address Clag Analyzer issues. by turaj@webrtc.org · 11 years ago
  25. 72cc32a Propagate estimated RTT from receivers to rtt observer. by asapersson@webrtc.org · 11 years ago
  26. 1d76e9b Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc by sergeyu@chromium.org · 11 years ago
  27. bd5d9fa Add delay limit to ChokeFilter. by solenberg@webrtc.org · 11 years ago
  28. c2b6166 Logging for BWE test framework. by solenberg@webrtc.org · 11 years ago
  29. 367af84 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. by turaj@webrtc.org · 11 years ago
  30. 991d66a Adding tl0idx consideration for continuity by mikhal@webrtc.org · 11 years ago
  31. 4ce7590 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
  32. 2714c79 Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. by sprang@webrtc.org · 11 years ago
  33. 77035fd Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h by xians@webrtc.org · 11 years ago
  34. 2ea3645 Added a "interleaved_" flag to webrtc::AudioFrame. by xians@webrtc.org · 11 years ago
  35. 191f4fe Change the low-bitrate handling in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  36. ecfef19 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  37. 84c1485 Add an extended filter option to audioproc. by andrew@webrtc.org · 11 years ago
  38. a2d942a Fix for incorrect RTT estimation. A too low RTT value could be estimated. by asapersson@webrtc.org · 11 years ago
  39. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  40. d7e9041 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. by andrew@webrtc.org · 11 years ago
  41. 45b5167 Add CurrentLayerId() to temporal layers. by marpan@webrtc.org · 11 years ago
  42. 4b3ff2d Framework for testing bandwidth estimation. by solenberg@webrtc.org · 11 years ago
  43. 4633e15 Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  44. c5b5ad1 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
  45. 127d8ad Minor comment fix after clang reformat. by andrew@webrtc.org · 11 years ago
  46. 2873c4c MouseCursorMonitor implementation for OSX and Windows. by sergeyu@chromium.org · 11 years ago
  47. fbd6969 Remove unused kPowTableFrac which causes anroid clang build failure. by wu@webrtc.org · 11 years ago
  48. ba6d56c Add MouseCursorRenderer. by sergeyu@chromium.org · 11 years ago
  49. af54d4b Add MouseCursorCapturer interface with implementation for X11. by sergeyu@chromium.org · 11 years ago
  50. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  51. aa445e7 Make RtpData and RtpFeedback destructors public. by stefan@webrtc.org · 11 years ago
  52. bec453d Compile out unused kMinTrustedDelayMs. by andrew@webrtc.org · 11 years ago
  53. 9b1b525 NetEq4: Removing templatization for AudioVector by henrik.lundin@webrtc.org · 11 years ago
  54. 438ae6f Remove empty line in SharedXDisplay::RemoveEventHandler. by sergeyu@chromium.org · 11 years ago
  55. cbde20c Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. by henrike@webrtc.org · 11 years ago
  56. 07e0f6c Add event handling in SharedXDisplay. by sergeyu@chromium.org · 11 years ago
  57. 91685dc Add DesktopCaptureOptions class. by sergeyu@chromium.org · 11 years ago
  58. cb90617 WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties. by henrike@webrtc.org · 11 years ago
  59. 2f40af3 clang-format audio_processing/aec/* by andrew@webrtc.org · 11 years ago
  60. 17fdf2a Add a parameter to audioproc for overriding the delay. by andrew@webrtc.org · 11 years ago
  61. 757a950 Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." by stefan@webrtc.org · 11 years ago
  62. 244d629 Fix build error in r4934. by stefan@webrtc.org · 11 years ago
  63. 73063f3 Add a tool for parsing an RTP file and outputting the BWE relevant fields. by stefan@webrtc.org · 11 years ago
  64. 0a1c75a Wrap ACM2 code inside acm2 namespace. This gurantees that one ACM would not use components of others by accident. by turaj@webrtc.org · 11 years ago
  65. bda9cbe Accounting for wrap-around of timestamps. by turaj@webrtc.org · 11 years ago
  66. 0640850 VPM: Fixing namespace by mikhal@webrtc.org · 11 years ago
  67. 3213616 Android: enable camera video stabilization when available. by fischman@webrtc.org · 11 years ago
  68. bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  69. 73dacd4 Only declare kDelayDiffOffset when used. by andrew@webrtc.org · 11 years ago
  70. fae046e Unbreaks Android build after r4915. by henrike@webrtc.org · 11 years ago
  71. 3f02f98 Revert r4913 that reverts r4911. Original CL description: by andresp@webrtc.org · 11 years ago
  72. 4b14e5a Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  73. 81cd5ca VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  74. e98a3de Revert 4911 "Adding temporal layer strategy that keeps base laye..." by turaj@webrtc.org · 11 years ago
  75. b576a69 Reformatting VPM: First step - No functional changes. by mikhal@webrtc.org · 11 years ago
  76. 03ced52 Adding temporal layer strategy that keeps base layer framerate at an acceptable value. by andresp@webrtc.org · 11 years ago
  77. a8532a8 Disable Receiver unittests on Android. by turaj@webrtc.org · 11 years ago
  78. 85cdc39 ACM test are modified to run with both ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  79. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  80. 1b59234 Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
  81. 59e1db1 Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
  82. 6583dff APK for opensl loopback. by henrike@webrtc.org · 11 years ago
  83. d6da239 Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
  84. 053d45a Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
  85. b9421ac Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
  86. 2934af5 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
  87. 37da9ab Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
  88. 0e9c399 NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  89. 24f0702 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  90. d4e1329 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
  91. 76a6ffb Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
  92. 0d4d51b Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
  93. 76238f6 Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
  94. cd5c882 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  95. 2b35b95 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  96. 9e035d2 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  97. b503d1e Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  98. 424e0e4 With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  99. 44f030c Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
  100. dc1f7e9 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago