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webrtc / src / b56d0e383e62f74651aa78b9c2857e1b57a8729b / webrtc
  1. b56d0e3 Change the low-bitrate handling in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  2. 37bb497 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  3. d1bcf11 Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined. by andrew@webrtc.org · 11 years ago
  4. 22858d4 Add an extended filter option to audioproc. by andrew@webrtc.org · 11 years ago
  5. 042e91c Fix for incorrect RTT estimation. A too low RTT value could be estimated. by asapersson@webrtc.org · 11 years ago
  6. ba975e2 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
  7. 886aef0 Fixing broken tests in voe_auto_test extended by tina.legrand@webrtc.org · 11 years ago
  8. 31628aa Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  9. 06b60c0 Roll chromium_revision 228675:229708 by kjellander@webrtc.org · 11 years ago
  10. 621df67 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. by andrew@webrtc.org · 11 years ago
  11. 943e3b9 Add CurrentLayerId() to temporal layers. by marpan@webrtc.org · 11 years ago
  12. 9c735c4 Updated WebRTC version to 3.45 by elham@webrtc.org · 11 years ago
  13. 8215106 Framework for testing bandwidth estimation. by solenberg@webrtc.org · 11 years ago
  14. 29dd0de Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  15. 0d19ed9 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
  16. fe1ef93 Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  17. e053629 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
  18. eb61a85 Move audio_e2e_harness into include_tests==1 condition. by kjellander@webrtc.org · 11 years ago
  19. 88a3108 Add audio_e2e_test target to tools.gyp by kjellander@webrtc.org · 11 years ago
  20. fb648da 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
  21. 3e00505 Have padding decay to zero if no frames are being captured. by stefan@webrtc.org · 11 years ago
  22. 893c07f Disable the -Wno-unused-const-variable Clang warning on Mac by kjellander@webrtc.org · 11 years ago
  23. 89b1e68 Minor comment fix after clang reformat. by andrew@webrtc.org · 11 years ago
  24. 2df89c0 MouseCursorMonitor implementation for OSX and Windows. by sergeyu@chromium.org · 11 years ago
  25. 6342066 Fix tsan failures in channel.cc regarding to the volume settings. by wu@webrtc.org · 11 years ago
  26. 675e260 Check the number of playout channels instead of the send channels in StopPlayout() by xians@webrtc.org · 11 years ago
  27. c11148b Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  28. d030972 Remove unused kPowTableFrac which causes anroid clang build failure. by wu@webrtc.org · 11 years ago
  29. 25fce9a Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
  30. b400aa7 Don't pad if only one stream is sent, except if auto muted. by stefan@webrtc.org · 11 years ago
  31. e7009f3 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
  32. 5d957e2 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
  33. 9401524 Run FullStack tests without render windows. by pbos@webrtc.org · 11 years ago
  34. 5ed4f46 Remove TSan v2 disabled test in condition_variable_unittest.cc by kjellander@webrtc.org · 11 years ago
  35. b44c2a3 Open file in binary in CreateFromYuvFile(). by pbos@webrtc.org · 11 years ago
  36. e6e749d Add MouseCursorRenderer. by sergeyu@chromium.org · 11 years ago
  37. 2767b53 Add MouseCursorCapturer interface with implementation for X11. by sergeyu@chromium.org · 11 years ago
  38. 3555303 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  39. e5021fe Make RtpData and RtpFeedback destructors public. by stefan@webrtc.org · 11 years ago
  40. 266c7b3 Move ChromaGenerator to common_video/. by pbos@webrtc.org · 11 years ago
  41. c2e471d Compile out unused kMinTrustedDelayMs. by andrew@webrtc.org · 11 years ago
  42. 901ae77 Android: Fixes WebRTCDemo build (missing Java code). by henrike@webrtc.org · 11 years ago
  43. 1871dd2 NetEq4: Removing templatization for AudioVector by henrik.lundin@webrtc.org · 11 years ago
  44. 3079298 Remove empty line in SharedXDisplay::RemoveEventHandler. by sergeyu@chromium.org · 11 years ago
  45. 05773e5 Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots. by henrike@webrtc.org · 11 years ago
  46. 7419a72 Add event handling in SharedXDisplay. by sergeyu@chromium.org · 11 years ago
  47. 894e6fe9 Add DesktopCaptureOptions class. by sergeyu@chromium.org · 11 years ago
  48. f53622d 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
  49. 5b3b6b1 Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  50. 13b2d46 clang-format audio_processing/aec/* by andrew@webrtc.org · 11 years ago
  51. ca764ab Add a parameter to audioproc for overriding the delay. by andrew@webrtc.org · 11 years ago
  52. 11e9cbc Updated WebRTC version to 3.44 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  53. f5d7c58 Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." by stefan@webrtc.org · 11 years ago
  54. 611e514 Fix build error in r4934. by stefan@webrtc.org · 11 years ago
  55. bc99bcf Add a tool for parsing an RTP file and outputting the BWE relevant fields. by stefan@webrtc.org · 11 years ago
  56. 6d5d248 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
  57. f316396 Accounting for wrap-around of timestamps. by turaj@webrtc.org · 11 years ago
  58. 35e4dd3 VPM: Fixing namespace by mikhal@webrtc.org · 11 years ago
  59. 4598380 Android: enable camera video stabilization when available. by fischman@webrtc.org · 11 years ago
  60. 7fca2ce Add owners to [webrtc,talk]/build and *.isolate (take 2) by kjellander@webrtc.org · 11 years ago
  61. 495f29e Remove unused Android dummy APK by kjellander@webrtc.org · 11 years ago
  62. 3f9288f Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  63. 6c264cc Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  64. 83b9e5b Add owners to [webrtc,talk]/build and *.isolate by kjellander@webrtc.org · 11 years ago
  65. acb0050 Only declare kDelayDiffOffset when used. by andrew@webrtc.org · 11 years ago
  66. ad2eb6f Unbreaks Android build after r4915. by henrike@webrtc.org · 11 years ago
  67. be9c560 Revert r4913 that reverts r4911. Original CL description: by andresp@webrtc.org · 11 years ago
  68. 6c82e04 Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  69. 4e65e07 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  70. 44db9d1 Revert 4911 "Adding temporal layer strategy that keeps base laye..." by turaj@webrtc.org · 11 years ago
  71. b43d807 Reformatting VPM: First step - No functional changes. by mikhal@webrtc.org · 11 years ago
  72. 26f78f7 Adding temporal layer strategy that keeps base layer framerate at an acceptable value. by andresp@webrtc.org · 11 years ago
  73. 70df305 Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  74. 7ee3efb Disable Receiver unittests on Android. by turaj@webrtc.org · 11 years ago
  75. 6ea3d1c ACM test are modified to run with both ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  76. 2a97317 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  77. f8f78b1 Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
  78. 9b5c807 Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  79. 4887114 Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
  80. c0b4c4a Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
  81. 1fdc51a APK for opensl loopback. by henrike@webrtc.org · 11 years ago
  82. de74b64 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  83. 7ea4f24 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  84. 8469f7b Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
  85. c016770 Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
  86. a6101d7 Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 12 years ago
  87. b74b96f Test multiple send/receive streams in Call. by pbos@webrtc.org · 12 years ago
  88. e546f02 Remove include_dirs from utility. by pbos@webrtc.org · 12 years ago
  89. 7e4d0df PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 12 years ago
  90. 5222270 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 12 years ago
  91. 8e2f9bc Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 12 years ago
  92. fd11bbf NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 12 years ago
  93. 6ad6a07 Support for CELT in NetEq4. by turaj@webrtc.org · 12 years ago
  94. 9532fa5 Remove include_dirs from video_render. by pbos@webrtc.org · 12 years ago
  95. 1c974ef Remove include_dirs from video_capture. by pbos@webrtc.org · 12 years ago
  96. 4cd7622 Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 12 years ago
  97. 572699d Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 12 years ago
  98. cc92e00 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 12 years ago
  99. a20a22a Support for CELT in NetEq4. by turaj@webrtc.org · 12 years ago
  100. 30377c7 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 12 years ago