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gerrit-public.fairphone.software / platform / external / webrtc / 19f27e6a24f877fc2b0409a94b02d5f40ba3dc8c
  1. 19f27e6 Update talk to 54527154. by mallinath@webrtc.org · 11 years ago
  2. 7419a72 Add event handling in SharedXDisplay. by sergeyu@chromium.org · 11 years ago
  3. 894e6fe Add DesktopCaptureOptions class. by sergeyu@chromium.org · 11 years ago
  4. 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
  5. 4c61792 Add SyzyASan to DEPS by kjellander@webrtc.org · 11 years ago
  6. 5b3b6b1 Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  7. 40dfbc4 Update talk to 53984350. by wu@webrtc.org · 11 years ago
  8. 4551b79 Update libjingle to 53920541. by wu@webrtc.org · 11 years ago
  9. 13b2d46 clang-format audio_processing/aec/* by andrew@webrtc.org · 11 years ago
  10. d241718 Increase base Chromium revision to get an update to libnss. by wu@webrtc.org · 11 years ago
  11. ff7b360 * Remove suppressions that are fixed. by wu@webrtc.org · 11 years ago
  12. 7818752 Update libjingle to 53856368. by wu@webrtc.org · 11 years ago
  13. e0d55a0 Removing suppressions that has been fixed, i.e. r4661. by wu@webrtc.org · 11 years ago
  14. ca764ab Add a parameter to audioproc for overriding the delay. by andrew@webrtc.org · 11 years ago
  15. 11e9cbc Updated WebRTC version to 3.44 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  16. f5d7c58 Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields." by stefan@webrtc.org · 11 years ago
  17. 611e514 Fix build error in r4934. by stefan@webrtc.org · 11 years ago
  18. bc99bcf Add a tool for parsing an RTP file and outputting the BWE relevant fields. by stefan@webrtc.org · 11 years ago
  19. 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
  20. f316396 Accounting for wrap-around of timestamps. by turaj@webrtc.org · 11 years ago
  21. 20078e2 Support video constraints and use key/value pairs. by andrew@webrtc.org · 11 years ago
  22. 35e4dd3 VPM: Fixing namespace by mikhal@webrtc.org · 11 years ago
  23. 4598380 Android: enable camera video stabilization when available. by fischman@webrtc.org · 11 years ago
  24. 7fca2ce Add owners to [webrtc,talk]/build and *.isolate (take 2) by kjellander@webrtc.org · 11 years ago
  25. 495f29e Remove unused Android dummy APK by kjellander@webrtc.org · 11 years ago
  26. e693818 Add isolate targets for libjingle by kjellander@webrtc.org · 11 years ago
  27. 3f9288f Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  28. 6c264cc Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
  29. 83b9e5b Add owners to [webrtc,talk]/build and *.isolate by kjellander@webrtc.org · 11 years ago
  30. acb0050 Only declare kDelayDiffOffset when used. by andrew@webrtc.org · 11 years ago
  31. ad2eb6f Unbreaks Android build after r4915. by henrike@webrtc.org · 11 years ago
  32. be9c560 Revert r4913 that reverts r4911. Original CL description: by andresp@webrtc.org · 11 years ago
  33. bab2aa5 Add audio and video parameters for setting media constraints. by andrew@webrtc.org · 11 years ago
  34. 4446134 AppRTCDemo(android): support boolean value for MediaStreamConstraints.{audio,video}. by fischman@webrtc.org · 11 years ago
  35. a7266ca Fix clang build break by fischman@webrtc.org · 11 years ago
  36. 6c82e04 Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  37. 4e65e07 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  38. ddc5a19 AppRTCDemo(android): uncaught exceptions now display a modal dialog box before killing the app. by fischman@webrtc.org · 11 years ago
  39. 44db9d1 Revert 4911 "Adding temporal layer strategy that keeps base laye..." by turaj@webrtc.org · 11 years ago
  40. b43d807 Reformatting VPM: First step - No functional changes. by mikhal@webrtc.org · 11 years ago
  41. 26f78f7 Adding temporal layer strategy that keeps base layer framerate at an acceptable value. by andresp@webrtc.org · 11 years ago
  42. 70df305 Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  43. 7ee3efb Disable Receiver unittests on Android. by turaj@webrtc.org · 11 years ago
  44. 6ea3d1c ACM test are modified to run with both ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
  45. 2a97317 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  46. f8f78b1 Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
  47. 9b5c807 Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  48. 4887114 Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
  49. c0b4c4a Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
  50. 1fdc51a APK for opensl loopback. by henrike@webrtc.org · 11 years ago
  51. de74b64 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  52. 7ea4f24 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  53. 8469f7b Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
  54. c016770 Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
  55. a6101d7 Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
  56. ee6d0dd Upload Demo page to allow edit offer & Answer sdp in pc1 demo. by vikasmarwaha@webrtc.org · 11 years ago
  57. 1d731e4 Roll chromium_revision 224141:226099 to pick up jsoncpp fix for ARM by fischman@webrtc.org · 11 years ago
  58. 19134ba Updated device-switch demo page to work with Chrome M30. by vikasmarwaha@webrtc.org · 11 years ago
  59. b74b96f Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  60. e546f02 Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
  61. 7e4d0df PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 11 years ago
  62. 5222270 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
  63. 8e2f9bc Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
  64. fd11bbf NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
  65. 6ad6a07 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  66. 7e809c3 Update libjingle to CL 53496343. by mallinath@webrtc.org · 11 years ago
  67. 9532fa5 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
  68. 1c974ef Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
  69. 4cd7622 Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
  70. 572699d Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
  71. cc92e00 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
  72. a20a22a Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
  73. ad81ab8 Suppress SSL error strings on mac_asan to unbreak that build by mallinath@webrtc.org · 11 years ago
  74. 30377c7 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
  75. a27be8e Update libjingle to CL 53398036. by mallinath@webrtc.org · 11 years ago
  76. 34c50c1 Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
  77. a39b323 Add tools/sharding_supervisor to .gitignore by kjellander@webrtc.org · 11 years ago
  78. 8b7ec82 Exclude P2PTransportChannelSameNatTest.TestConesBehindSameCone for TSan Linux by kjellander@webrtc.org · 11 years ago
  79. 4475905 Disable flaky RapidSpeakerChange test. by andrew@webrtc.org · 11 years ago
  80. 6049787 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
  81. 137b379 Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
  82. 287f07b Add sharding_supervisor to DEPS to prepare for swarm/isolated testing. by kjellander@webrtc.org · 11 years ago
  83. 2e246b4 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  84. 3223a3d Roll libvpx 212975:225010 to pick up iOS Release fixes by fischman@webrtc.org · 11 years ago
  85. 663da0a With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
  86. f26e8f6 Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
  87. f8b2966 Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  88. 544b17c Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
  89. 04b6179 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago
  90. 97eefb7 Remove include_dirs from remote_bitrate_estimator. by pbos@webrtc.org · 11 years ago
  91. 339fe12 Remove include_dirs from bitrate_controller. by pbos@webrtc.org · 11 years ago
  92. 054ccd2 Remove include_dirs from video_coding. by pbos@webrtc.org · 11 years ago
  93. 73f2076 Remove include_dirs from video_processing. by pbos@webrtc.org · 11 years ago
  94. dc3fa08 Remove include_dirs from rtp_rtcp. by pbos@webrtc.org · 11 years ago
  95. 7b75ac6 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. by turaj@webrtc.org · 11 years ago
  96. 6b1e219 Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 11 years ago
  97. 1760a17 Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
  98. becbefa Fix WindowCapturerWin to capture window decorations after window size changes. by sergeyu@chromium.org · 11 years ago
  99. 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
  100. 3e77036 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago