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webrtc / src / webrtc / 9653397068c20ea3d77f4acfd07254e091434baa
  1. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  2. aa445e7 Make RtpData and RtpFeedback destructors public. by stefan@webrtc.org · 11 years ago
  3. e2c52d7 Move ChromaGenerator to common_video/. by pbos@webrtc.org · 11 years ago
  4. bec453d Compile out unused kMinTrustedDelayMs. by andrew@webrtc.org · 11 years ago
  5. 9caedd0 Android: Fixes WebRTCDemo build (missing Java code). by henrike@webrtc.org · 11 years ago
  6. 9b1b525 NetEq4: Removing templatization for AudioVector by henrik.lundin@webrtc.org · 11 years ago
  7. 438ae6f Remove empty line in SharedXDisplay::RemoveEventHandler. by sergeyu@chromium.org · 11 years ago
  8. cbde20c Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. by henrike@webrtc.org · 11 years ago
  9. 07e0f6c Add event handling in SharedXDisplay. by sergeyu@chromium.org · 11 years ago
  10. 91685dc Add DesktopCaptureOptions class. by sergeyu@chromium.org · 11 years ago
  11. 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
  12. 8e70108 Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  13. 2f40af3 clang-format audio_processing/aec/* by andrew@webrtc.org · 11 years ago
  14. 17fdf2a Add a parameter to audioproc for overriding the delay. by andrew@webrtc.org · 11 years ago
  15. eeaea08 Updated WebRTC version to 3.44 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  16. 757a950 Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." by stefan@webrtc.org · 11 years ago
  17. 244d629 Fix build error in r4934. by stefan@webrtc.org · 11 years ago
  18. 73063f3 Add a tool for parsing an RTP file and outputting the BWE relevant fields. by stefan@webrtc.org · 11 years ago
  19. 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
  20. bda9cbe Accounting for wrap-around of timestamps. by turaj@webrtc.org · 11 years ago
  21. 0640850 VPM: Fixing namespace by mikhal@webrtc.org · 11 years ago
  22. 3213616 Android: enable camera video stabilization when available. by fischman@webrtc.org · 11 years ago
  23. 7c789f4 Add owners to [webrtc,talk]/build and *.isolate (take 2) by kjellander@webrtc.org · 11 years ago
  24. 8b8ae0f Remove unused Android dummy APK by kjellander@webrtc.org · 11 years ago
  25. bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  26. e06943f Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  27. 3124b2e Add owners to [webrtc,talk]/build and *.isolate by kjellander@webrtc.org · 11 years ago
  28. 73dacd4 Only declare kDelayDiffOffset when used. by andrew@webrtc.org · 11 years ago
  29. fae046e Unbreaks Android build after r4915. by henrike@webrtc.org · 11 years ago
  30. 3f02f98 Revert r4913 that reverts r4911. Original CL description: by andresp@webrtc.org · 11 years ago
  31. 4b14e5a Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  32. 81cd5ca VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  33. e98a3de Revert 4911 "Adding temporal layer strategy that keeps base laye..." by turaj@webrtc.org · 11 years ago
  34. b576a69 Reformatting VPM: First step - No functional changes. by mikhal@webrtc.org · 11 years ago
  35. 03ced52 Adding temporal layer strategy that keeps base layer framerate at an acceptable value. by andresp@webrtc.org · 11 years ago
  36. 499392c Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  37. a8532a8 Disable Receiver unittests on Android. by turaj@webrtc.org · 11 years ago
  38. 85cdc39 ACM test are modified to run with both ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  39. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  40. 1b59234 Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
  41. a6063fd Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  42. 59e1db1 Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
  43. 369da50 Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
  44. 6583dff APK for opensl loopback. by henrike@webrtc.org · 11 years ago
  45. b5d2d16 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  46. 39079d1 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  47. d6da239 Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
  48. 69598c5 Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
  49. 053d45a Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
  50. c5080a9 Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  51. b9421ac Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
  52. b82f683 PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 11 years ago
  53. 2934af5 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
  54. 37da9ab Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
  55. 0e9c399 NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  56. 24f0702 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  57. d4e1329 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
  58. 76a6ffb Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
  59. 0d4d51b Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
  60. 76238f6 Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
  61. 0de0049 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  62. cd5c882 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  63. 2b35b95 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  64. a3a3a0f Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
  65. 9e035d2 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  66. b503d1e Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  67. 362e3e5 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  68. 424e0e4 With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  69. cdc5e6a Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
  70. b655adf Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  71. 44f030c Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
  72. dc1f7e9 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago
  73. ee817d3 Remove include_dirs from remote_bitrate_estimator. by pbos@webrtc.org · 11 years ago
  74. fc75214 Remove include_dirs from bitrate_controller. by pbos@webrtc.org · 11 years ago
  75. 1fc4659 Remove include_dirs from video_coding. by pbos@webrtc.org · 11 years ago
  76. 85592ad Remove include_dirs from video_processing. by pbos@webrtc.org · 11 years ago
  77. 1800406 Remove include_dirs from rtp_rtcp. by pbos@webrtc.org · 11 years ago
  78. 2f0a942 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. by turaj@webrtc.org · 11 years ago
  79. d4f6789 Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 11 years ago
  80. 8ddec2c Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  81. 53fdd3b Fix WindowCapturerWin to capture window decorations after window size changes. by sergeyu@chromium.org · 11 years ago
  82. 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
  83. 72790c7 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
  84. f7d5a08 Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org by elham@webrtc.org · 11 years ago
  85. 7f35836 Re-enable verbose logging in NetEq4. by turaj@webrtc.org · 11 years ago
  86. 79c884c Convert DeviceInfoImpl::_captureCapabilities from a map to a vector. by fischman@webrtc.org · 11 years ago
  87. 99b6d9e Revert 4837 "Add an extended filter mode to AEC." by asapersson@webrtc.org · 11 years ago
  88. 83c5f62 Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  89. 933267f Small fixes to run ACM2 tests. by turaj@webrtc.org · 11 years ago
  90. 6ca9e7d API add to set background noise mode. by turaj@webrtc.org · 11 years ago
  91. 08099e0 Fix window capturer not to leak HDC. by sergeyu@chromium.org · 11 years ago
  92. 82707bf Fix window capturer to stop capturing when the target is minimized. by sergeyu@chromium.org · 11 years ago
  93. 4b067da Disable some VP8 tests on Android. by andrew@webrtc.org · 11 years ago
  94. 8da2f65 Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
  95. 3bd659f Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
  96. a89f7e8 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
  97. 890706b Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
  98. da6d2a2 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  99. b0382ea Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
  100. 510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago