1. b45ab8b Roll chromium_revision 228675:229708 by kjellander@webrtc.org · 11 years ago
  2. d7e9041 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. by andrew@webrtc.org · 11 years ago
  3. 45b5167 Add CurrentLayerId() to temporal layers. by marpan@webrtc.org · 11 years ago
  4. 6796d68 Updated WebRTC version to 3.45 by elham@webrtc.org · 11 years ago
  5. 4b3ff2d Framework for testing bandwidth estimation. by solenberg@webrtc.org · 11 years ago
  6. 4633e15 Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  7. 7c46e95 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
  8. 63301bd Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  9. c5b5ad1 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
  10. 50edafc Move audio_e2e_harness into include_tests==1 condition. by kjellander@webrtc.org · 11 years ago
  11. b9586f0 Add audio_e2e_test target to tools.gyp by kjellander@webrtc.org · 11 years ago
  12. b27e670 Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug. by wu@webrtc.org · 11 years ago
  13. 5e74d96 Have padding decay to zero if no frames are being captured. by stefan@webrtc.org · 11 years ago
  14. ba368fc Disable the -Wno-unused-const-variable Clang warning on Mac by kjellander@webrtc.org · 11 years ago
  15. 127d8ad Minor comment fix after clang reformat. by andrew@webrtc.org · 11 years ago
  16. 2873c4c MouseCursorMonitor implementation for OSX and Windows. by sergeyu@chromium.org · 11 years ago
  17. f7651ef Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
  18. 3d553d4 Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
  19. 51e0101 Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  20. fbd6969 Remove unused kPowTableFrac which causes anroid clang build failure. by wu@webrtc.org · 11 years ago
  21. 44bb62a Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
  22. 93cd397 Don't pad if only one stream is sent, except if auto muted. by stefan@webrtc.org · 11 years ago
  23. 9398252 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
  24. 6c9c551 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
  25. a24c356 Run FullStack tests without render windows. by pbos@webrtc.org · 11 years ago
  26. cdcedb2 Remove TSan v2 disabled test in condition_variable_unittest.cc by kjellander@webrtc.org · 11 years ago
  27. f2e99be Open file in binary in CreateFromYuvFile(). by pbos@webrtc.org · 11 years ago
  28. ba6d56c Add MouseCursorRenderer. by sergeyu@chromium.org · 11 years ago
  29. af54d4b Add MouseCursorCapturer interface with implementation for X11. by sergeyu@chromium.org · 11 years ago
  30. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  31. aa445e7 Make RtpData and RtpFeedback destructors public. by stefan@webrtc.org · 11 years ago
  32. e2c52d7 Move ChromaGenerator to common_video/. by pbos@webrtc.org · 11 years ago
  33. bec453d Compile out unused kMinTrustedDelayMs. by andrew@webrtc.org · 11 years ago
  34. 9caedd0 Android: Fixes WebRTCDemo build (missing Java code). by henrike@webrtc.org · 11 years ago
  35. 9b1b525 NetEq4: Removing templatization for AudioVector by henrik.lundin@webrtc.org · 11 years ago
  36. 438ae6f Remove empty line in SharedXDisplay::RemoveEventHandler. by sergeyu@chromium.org · 11 years ago
  37. cbde20c Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. by henrike@webrtc.org · 11 years ago
  38. 07e0f6c Add event handling in SharedXDisplay. by sergeyu@chromium.org · 11 years ago
  39. 91685dc Add DesktopCaptureOptions class. by sergeyu@chromium.org · 11 years ago
  40. 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
  41. 8e70108 Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  42. 2f40af3 clang-format audio_processing/aec/* by andrew@webrtc.org · 11 years ago
  43. 17fdf2a Add a parameter to audioproc for overriding the delay. by andrew@webrtc.org · 11 years ago
  44. eeaea08 Updated WebRTC version to 3.44 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  45. 757a950 Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." by stefan@webrtc.org · 11 years ago
  46. 244d629 Fix build error in r4934. by stefan@webrtc.org · 11 years ago
  47. 73063f3 Add a tool for parsing an RTP file and outputting the BWE relevant fields. by stefan@webrtc.org · 11 years ago
  48. 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
  49. bda9cbe Accounting for wrap-around of timestamps. by turaj@webrtc.org · 11 years ago
  50. 0640850 VPM: Fixing namespace by mikhal@webrtc.org · 11 years ago
  51. 3213616 Android: enable camera video stabilization when available. by fischman@webrtc.org · 11 years ago
  52. 7c789f4 Add owners to [webrtc,talk]/build and *.isolate (take 2) by kjellander@webrtc.org · 11 years ago
  53. 8b8ae0f Remove unused Android dummy APK by kjellander@webrtc.org · 11 years ago
  54. bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  55. e06943f Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  56. 3124b2e Add owners to [webrtc,talk]/build and *.isolate by kjellander@webrtc.org · 11 years ago
  57. 73dacd4 Only declare kDelayDiffOffset when used. by andrew@webrtc.org · 11 years ago
  58. fae046e Unbreaks Android build after r4915. by henrike@webrtc.org · 11 years ago
  59. 3f02f98 Revert r4913 that reverts r4911. Original CL description: by andresp@webrtc.org · 11 years ago
  60. 4b14e5a Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  61. 81cd5ca VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  62. e98a3de Revert 4911 "Adding temporal layer strategy that keeps base laye..." by turaj@webrtc.org · 11 years ago
  63. b576a69 Reformatting VPM: First step - No functional changes. by mikhal@webrtc.org · 11 years ago
  64. 03ced52 Adding temporal layer strategy that keeps base layer framerate at an acceptable value. by andresp@webrtc.org · 11 years ago
  65. 499392c Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  66. a8532a8 Disable Receiver unittests on Android. by turaj@webrtc.org · 11 years ago
  67. 85cdc39 ACM test are modified to run with both ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  68. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  69. 1b59234 Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
  70. a6063fd Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  71. 59e1db1 Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
  72. 369da50 Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
  73. 6583dff APK for opensl loopback. by henrike@webrtc.org · 11 years ago
  74. b5d2d16 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  75. 39079d1 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  76. d6da239 Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
  77. 69598c5 Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
  78. 053d45a Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
  79. c5080a9 Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  80. b9421ac Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
  81. b82f683 PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 11 years ago
  82. 2934af5 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
  83. 37da9ab Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
  84. 0e9c399 NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  85. 24f0702 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  86. d4e1329 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
  87. 76a6ffb Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
  88. 0d4d51b Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
  89. 76238f6 Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
  90. 0de0049 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  91. cd5c882 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  92. 2b35b95 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  93. a3a3a0f Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
  94. 9e035d2 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  95. b503d1e Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  96. 362e3e5 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  97. 424e0e4 With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  98. cdc5e6a Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
  99. b655adf Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  100. 44f030c Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago