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gerrit-public.fairphone.software / platform / external / webrtc / f8f78b1316443335209cf5b255863279c017e285
  1. f8f78b1 Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
  2. 9b5c807 Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  3. 4887114 Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
  4. c0b4c4a Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
  5. 1fdc51a APK for opensl loopback. by henrike@webrtc.org · 11 years ago
  6. de74b64 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  7. 7ea4f24 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  8. 8469f7b Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
  9. c016770 Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
  10. a6101d7 Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
  11. ee6d0dd Upload Demo page to allow edit offer & Answer sdp in pc1 demo. by vikasmarwaha@webrtc.org · 11 years ago
  12. 1d731e4 Roll chromium_revision 224141:226099 to pick up jsoncpp fix for ARM by fischman@webrtc.org · 11 years ago
  13. 19134ba Updated device-switch demo page to work with Chrome M30. by vikasmarwaha@webrtc.org · 11 years ago
  14. b74b96f Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  15. e546f02 Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
  16. 7e4d0df PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 11 years ago
  17. 5222270 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
  18. 8e2f9bc Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
  19. fd11bbf NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  20. 6ad6a07 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  21. 7e809c3 Update libjingle to CL 53496343. by mallinath@webrtc.org · 11 years ago
  22. 9532fa5 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
  23. 1c974ef Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
  24. 4cd7622 Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
  25. 572699d Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
  26. cc92e00 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  27. a20a22a Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  28. ad81ab8 Suppress SSL error strings on mac_asan to unbreak that build by mallinath@webrtc.org · 11 years ago
  29. 30377c7 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  30. a27be8e Update libjingle to CL 53398036. by mallinath@webrtc.org · 11 years ago
  31. 34c50c1 Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
  32. a39b323 Add tools/sharding_supervisor to .gitignore by kjellander@webrtc.org · 11 years ago
  33. 8b7ec82 Exclude P2PTransportChannelSameNatTest.TestConesBehindSameCone for TSan Linux by kjellander@webrtc.org · 11 years ago
  34. 4475905 Disable flaky RapidSpeakerChange test. by andrew@webrtc.org · 11 years ago
  35. 6049787 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  36. 137b379 Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  37. 287f07b Add sharding_supervisor to DEPS to prepare for swarm/isolated testing. by kjellander@webrtc.org · 11 years ago
  38. 2e246b4 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  39. 3223a3d Roll libvpx 212975:225010 to pick up iOS Release fixes by fischman@webrtc.org · 11 years ago
  40. 663da0a With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  41. f26e8f6 Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
  42. f8b2966 Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  43. 544b17c Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
  44. 04b6179 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago
  45. 97eefb7 Remove include_dirs from remote_bitrate_estimator. by pbos@webrtc.org · 11 years ago
  46. 339fe12 Remove include_dirs from bitrate_controller. by pbos@webrtc.org · 11 years ago
  47. 054ccd2 Remove include_dirs from video_coding. by pbos@webrtc.org · 11 years ago
  48. 73f2076 Remove include_dirs from video_processing. by pbos@webrtc.org · 11 years ago
  49. dc3fa08 Remove include_dirs from rtp_rtcp. by pbos@webrtc.org · 11 years ago
  50. 7b75ac6 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. by turaj@webrtc.org · 11 years ago
  51. 6b1e219 Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 11 years ago
  52. 1760a17 Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  53. becbefa Fix WindowCapturerWin to capture window decorations after window size changes. by sergeyu@chromium.org · 11 years ago
  54. 3fdeddb 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
  55. 3e77036 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
  56. cecaae2 Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org by elham@webrtc.org · 11 years ago
  57. 0c0fae8 Re-enable verbose logging in NetEq4. by turaj@webrtc.org · 11 years ago
  58. 69fc315 Convert DeviceInfoImpl::_captureCapabilities from a map to a vector. by fischman@webrtc.org · 11 years ago
  59. ce014d9 Revert 4837 "Add an extended filter mode to AEC." by asapersson@webrtc.org · 11 years ago
  60. 2f240b4 Disable some flaky libjingle base tests. by andrew@webrtc.org · 11 years ago
  61. 26e02f0 Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  62. d6a7a5f Small fixes to run ACM2 tests. by turaj@webrtc.org · 11 years ago
  63. ff43c85 API add to set background noise mode. by turaj@webrtc.org · 11 years ago
  64. 8d757ac Fix window capturer not to leak HDC. by sergeyu@chromium.org · 11 years ago
  65. 958cdf6 Fix window capturer to stop capturing when the target is minimized. by sergeyu@chromium.org · 11 years ago
  66. f832a55 Disable flaky TestPartialFrameHeader. by andrew@webrtc.org · 11 years ago
  67. c2ac5ed Add more TSAN suppressions for libjingle_media_unittest. by andrew@webrtc.org · 11 years ago
  68. 641587f Disable some VP8 tests on Android. by andrew@webrtc.org · 11 years ago
  69. f0f92fa Disable flaky SendDataMultipleClocks. by andrew@webrtc.org · 11 years ago
  70. 9b6eefc Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
  71. ebd3ab0 Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
  72. 6e86349 Disable tests that crash the OS X kernel when run under memcheck. by kjellander@webrtc.org · 11 years ago
  73. b0e6eb5 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
  74. 01aad09 Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
  75. b426c46 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
  76. 6ffc74e Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
  77. eb524d9 Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
  78. 1112c30 Update libjingle to 53057474. by mallinath@webrtc.org · 11 years ago
  79. e2af622 - Reset capture deltas at resolution change. by asapersson@webrtc.org · 11 years ago
  80. bec11ef Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 11 years ago
  81. b533a82 Disabled flaky tests. BUG=2409 R=andrew@webrtc.org, mallinath@webrtc.org by asapersson@webrtc.org · 11 years ago
  82. 7a7b929 Updated dc1.html to support SCTP transport. by vikasmarwaha@webrtc.org · 11 years ago
  83. 334865e Re-enable VideoCaptureTest.CreateDelete by fischman@webrtc.org · 11 years ago
  84. 038e8e6 Updated WebRTC version to 3.42 by elham@webrtc.org · 11 years ago
  85. 98fcd2d Adding unit tests for default temporal layer strategy. by andresp@webrtc.org · 11 years ago
  86. cdd3d4d Revert test change in r4808. by stefan@webrtc.org · 11 years ago
  87. 269dd42 Reduce flakiness in network down test. by stefan@webrtc.org · 11 years ago
  88. 63fe8e1 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
  89. 2edb642 Fix bugs in DesktopRegion::Subtract(). by sergeyu@chromium.org · 11 years ago
  90. cee0dfb Made sure that DTLS/SRTP is set to false in apprtc demo when testing loopback. See crbug/294881 for details. by vikasmarwaha@webrtc.org · 11 years ago
  91. 10e6cc7 VAD changes ported to ACM2. by turaj@webrtc.org · 11 years ago
  92. 362a55e Address Windows 64-bits warnings. by turaj@webrtc.org · 11 years ago
  93. 0e63e76 Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
  94. 9c74be7 Disable flaky video capture test. by stefan@webrtc.org · 11 years ago
  95. 4f3624d Avoid recursively taking critical section. by stefan@webrtc.org · 11 years ago
  96. dd57cd6 Removing the tsan text exclusion since the tests should be passing now. by jiayl@webrtc.org · 11 years ago
  97. d29ab4e Suppress SSL error strings on mac_asan to unbreak that build by fischman@webrtc.org · 11 years ago
  98. 76fe930 Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 11 years ago
  99. ccddd0a Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 11 years ago
  100. 6917e19 Rename EngineTest to CallTest. by pbos@webrtc.org · 11 years ago