Gitiles
Code Review Sign In
gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 26a736f764385826c477d308c08a1a69a5f4289d
  1. 0b7aefe Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  2. f50f118 clang-format audio_processing/aec/* by andrew@webrtc.org · 11 years ago
  3. fd03cb1 Add a parameter to audioproc for overriding the delay. by andrew@webrtc.org · 11 years ago
  4. 472d9a7 Updated WebRTC version to 3.44 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  5. 96ea7ac Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." by stefan@webrtc.org · 11 years ago
  6. 2dd26d8 Fix build error in r4934. by stefan@webrtc.org · 11 years ago
  7. 6dea08f Add a tool for parsing an RTP file and outputting the BWE relevant fields. by stefan@webrtc.org · 11 years ago
  8. e3976cf 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
  9. e9a3119 Accounting for wrap-around of timestamps. by turaj@webrtc.org · 11 years ago
  10. 7a4ff8a VPM: Fixing namespace by mikhal@webrtc.org · 11 years ago
  11. d4b124a Android: enable camera video stabilization when available. by fischman@webrtc.org · 11 years ago
  12. 7de0054 Add owners to [webrtc,talk]/build and *.isolate (take 2) by kjellander@webrtc.org · 11 years ago
  13. b7768d5 Remove unused Android dummy APK by kjellander@webrtc.org · 11 years ago
  14. 9670be6 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  15. 28ea6f8 Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  16. 9466714 Add owners to [webrtc,talk]/build and *.isolate by kjellander@webrtc.org · 11 years ago
  17. d64f84f Only declare kDelayDiffOffset when used. by andrew@webrtc.org · 11 years ago
  18. a299658 Unbreaks Android build after r4915. by henrike@webrtc.org · 11 years ago
  19. b17cc30 Revert r4913 that reverts r4911. Original CL description: by andresp@webrtc.org · 11 years ago
  20. f327500 Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  21. a5c9463 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  22. cca73e7 Revert 4911 "Adding temporal layer strategy that keeps base laye..." by turaj@webrtc.org · 11 years ago
  23. 0a202c7 Reformatting VPM: First step - No functional changes. by mikhal@webrtc.org · 11 years ago
  24. 05927b8 Adding temporal layer strategy that keeps base layer framerate at an acceptable value. by andresp@webrtc.org · 11 years ago
  25. 886983e Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  26. 8217db9 Disable Receiver unittests on Android. by turaj@webrtc.org · 11 years ago
  27. 0d6ee5d ACM test are modified to run with both ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  28. 3de1b22 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  29. c73a60e Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
  30. 15eda06 Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  31. f9159ec Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
  32. 712d30f Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
  33. d6c8fec APK for opensl loopback. by henrike@webrtc.org · 11 years ago
  34. 8ee2137 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  35. 55312fe Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  36. adba142 Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
  37. c6c6c1d Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
  38. 37ae275 Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
  39. be46543 Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  40. 7befa0c Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
  41. d2ca96e PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 11 years ago
  42. cf71152 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
  43. 324dac6 Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
  44. 80f29d8 NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  45. d069ddf Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  46. 9d8bbb7 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
  47. 6ea626e Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
  48. 5c9d8ee Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
  49. 38b1789 Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
  50. 1364cf1 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  51. 10b35b2 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  52. 197a5eb Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  53. 7540ded Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
  54. aaa1ce6 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  55. db10d03 Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  56. 45f59dd Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  57. 8ac5bf4 With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  58. 1a58624 Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
  59. 109108e Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  60. 1bd61f2 Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
  61. cbfa687 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago
  62. 5cb000f Remove include_dirs from remote_bitrate_estimator. by pbos@webrtc.org · 11 years ago
  63. 8895173 Remove include_dirs from bitrate_controller. by pbos@webrtc.org · 11 years ago
  64. a548c20 Remove include_dirs from video_coding. by pbos@webrtc.org · 11 years ago
  65. f8c0ed5 Remove include_dirs from video_processing. by pbos@webrtc.org · 11 years ago
  66. f98e9d8 Remove include_dirs from rtp_rtcp. by pbos@webrtc.org · 11 years ago
  67. 749f3e3 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. by turaj@webrtc.org · 11 years ago
  68. 870404d Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 11 years ago
  69. ad80fde Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  70. e5f36e0 Fix WindowCapturerWin to capture window decorations after window size changes. by sergeyu@chromium.org · 11 years ago
  71. e45a98b 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
  72. d49bc92 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
  73. 6b42cc0 Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org by elham@webrtc.org · 11 years ago
  74. 38cef0a Re-enable verbose logging in NetEq4. by turaj@webrtc.org · 11 years ago
  75. 17583a4 Convert DeviceInfoImpl::_captureCapabilities from a map to a vector. by fischman@webrtc.org · 11 years ago
  76. 85c5e8f Revert 4837 "Add an extended filter mode to AEC." by asapersson@webrtc.org · 11 years ago
  77. 000fecc Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  78. 3d9ce0c Small fixes to run ACM2 tests. by turaj@webrtc.org · 11 years ago
  79. d4b2d0c API add to set background noise mode. by turaj@webrtc.org · 11 years ago
  80. 2b7e421 Fix window capturer not to leak HDC. by sergeyu@chromium.org · 11 years ago
  81. f6795cd Fix window capturer to stop capturing when the target is minimized. by sergeyu@chromium.org · 11 years ago
  82. 0ab1945 Disable some VP8 tests on Android. by andrew@webrtc.org · 11 years ago
  83. a064105 Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
  84. 88bcc98 Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
  85. 957be53 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
  86. 7e07f16 Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
  87. 48b1173 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  88. 653dfe1 Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
  89. d24ce00 Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  90. 5cca7ed - Reset capture deltas at resolution change. by asapersson@webrtc.org · 11 years ago
  91. 72fde7b Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 11 years ago
  92. 584890b Re-enable VideoCaptureTest.CreateDelete by fischman@webrtc.org · 11 years ago
  93. 0715778 Updated WebRTC version to 3.42 by elham@webrtc.org · 11 years ago
  94. 697b7f3 Adding unit tests for default temporal layer strategy. by andresp@webrtc.org · 11 years ago
  95. 21318a9 Revert test change in r4808. by stefan@webrtc.org · 11 years ago
  96. dd3f2e4 Reduce flakiness in network down test. by stefan@webrtc.org · 11 years ago
  97. 0580c2c Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  98. 26caab2 Fix bugs in DesktopRegion::Subtract(). by sergeyu@chromium.org · 11 years ago
  99. 69dfcb4 VAD changes ported to ACM2. by turaj@webrtc.org · 11 years ago
  100. 344a2d7 Address Windows 64-bits warnings. by turaj@webrtc.org · 11 years ago