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gerrit-public.fairphone.software / platform / external / webrtc / e51b41ae4431abeac2eab4e9004172273722dde5
  1. e51b41a Added GN target for isac_test. by ivoc · 8 years ago
  2. 5d167d6 Removals and renamings in the new audio mixer. by aleloi · 8 years ago
  3. 76f91cd Add ThreadChecker to the TimestampAligner class. by nisse · 8 years ago
  4. 665d181 Increased column width for python tool rtp_analyzer.py. by aleloi · 8 years ago
  5. 30be5d7 Updated mixer unittests and fixed a related bug in the new mixer. by aleloi · 8 years ago
  6. 615d301 RTCStats and RTCStatsReport added (webrtc/stats). by hbos · 8 years ago
  7. 616df1e Added a level indicator to new mixer. by aleloi · 8 years ago
  8. 1f77905 Remove outdated symlink by kthelgason · 8 years ago
  9. a53fa0a Fix AppRTC Android Demo GN build when is_component_build=true. by sakal · 8 years ago
  10. 4c8adb1 MB: Flip Android bots to GN by default. by kjellander · 8 years ago
  11. 24ee050 CQ: Remove android_arm64_rel trybot by kjellander · 8 years ago
  12. b246a29 Define a protobuf format for representing plots. Add code to convert the C-representation generated by the RtcEventLog analysis tool, to the new protobuf format. by terelius · 8 years ago
  13. 6addf49 Adds function for computing moving average to visualization tool. by terelius · 8 years ago
  14. 5048f57 Add logs and small change in BasicPortAllocator. by Honghai Zhang · 8 years ago
  15. f99a9de ProbingEstimator: Erase history based on time threshold by Irfan Sheriff · 8 years ago
  16. 185ba29 Extract library from the RTCEventLog visualizer by skvlad · 8 years ago
  17. 5bed20f Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video. by Per · 8 years ago
  18. b37c45c GN: Add libjingle_peerconnection_java to peerconnection_unittests. by kjellander · 8 years ago
  19. a246cfb Don't include RTP headers in send-side BWE. by Stefan Holmer · 8 years ago
  20. 9a11784 Migrated GN target :g722_test by aleloi · 8 years ago
  21. 16f55a1 Migrated GN target :g711_test by aleloi · 8 years ago
  22. 649a21a Disable RampUpTest.UpDownUpThreeStreams. by philipel · 8 years ago
  23. 2e48646 RTC_CHECK and RTC_DCHECK macros for C by kwiberg · 8 years ago
  24. 7924697 Refactor WebRtcVideoCapturer. by nisse · 8 years ago
  25. d8dd190 GN: Fix test_support_unittests and MIPS compile issue. by kjellander · 8 years ago
  26. 84c03ba Add rtc_media as a direct dependency of rtc_media_unittests. by nisse · 8 years ago
  27. 0d1ad32 Add histogram for percentage of incoming frames that are limited in resolution due to cpu: by asapersson · 8 years ago
  28. 14cf12b Fixing TSan data race warning in video end-to-end tests. by Taylor Brandstetter · 8 years ago
  29. 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 8 years ago
  30. b3f1c5d Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine by henrik.lundin · 8 years ago
  31. e131ea5 Adding deadbeef and honghaiz as owners of webrtc/pc. by deadbeef · 8 years ago
  32. 72a5645 Removed the deactivation of the level controller when there is a built-in AGC available by peah · 8 years ago
  33. 8c16520 Method to parse event log directly from a string. by terelius · 8 years ago
  34. 6c46eaa Add gtest as a dependency for neteq_quality_test_support. by ehmaldonado · 8 years ago
  35. d48717b Fix issue where the number of packets reported in tests/simulations sometimes are negative. by stefan · 8 years ago
  36. 4ec01d9 Fix trivial lint errors in FileRecorder and FilePlayer by kwiberg · 8 years ago
  37. 853ecb2 Style cleanup in UpdateTmmbr: by danilchap · 8 years ago
  38. 7f82fc9 WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left shift of 1 by 31 overflows) by kwiberg · 8 years ago
  39. 642e3bc [rtcp] TransportFeedback adjusted to match other rtcp packets: by danilchap · 8 years ago
  40. 4981051 [Reland] Cleanup of the AudioDeviceBuffer class. by henrika · 8 years ago
  41. 83d79cd Revert of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #5 id:80001 of https://codereview.webrtc.org/2238253002/ ) by kjellander · 8 years ago
  42. 4381700 WebRtcVideoFrame constructor without transport_frame_id. by nisse · 8 years ago
  43. e5b4141 Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData by danilchap · 8 years ago
  44. ff101d6 iOS: add PlistBuddy location to path to avoid build errors by vopatop.skam · 8 years ago
  45. 94b9199 Add a copy of gyp_flag_compare from Chromium to WebRTC's webrtc/tools. by ehmaldonado · 8 years ago
  46. 4905f06 Disable the software noise suppressor on iOS devices as that by peah · 8 years ago
  47. abcc3de Add pps id and sps id parsing to the h.264 depacketizer. by stefan · 8 years ago
  48. 86ccd7b Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ ) by sakal · 8 years ago
  49. a7a01df Add field_trial_default dependency to libjingle_peerconnection by arlolra · 8 years ago
  50. 8177452 iOS H264VideoToolBoxEncoder: Stop scaling native CVPixelBuffers by magjed · 8 years ago
  51. d7a89db Revert of Cleanup of the AudioDeviceBuffer class (patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ ) by henrika · 8 years ago
  52. cf327b4 Cleanup of the AudioDeviceBuffer class. by henrika · 8 years ago
  53. da161d7 Reformat rtcp_receiver git cl format --full by danilchap · 8 years ago
  54. 861da3c Refactor neteq_test_support. by ehmaldonado · 8 years ago
  55. 294fb05 Add a timeout for starting the camera on CameraCapturer. by sakal · 8 years ago
  56. bcba64a GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets. by ehmaldonado · 8 years ago
  57. 4a85abb Add support for more resolutions on iOS/macOS by kthelgason · 8 years ago
  58. ec5c906 GN: Fix errors when some variables are set to non-default values. by kjellander · 8 years ago
  59. 72333d2 Add kjellander@webrtc.org to more BUILD.gn OWNERS files. by kjellander · 8 years ago
  60. 96b6b83 iOS: add type to peer connection local streams by vopatop.skam · 8 years ago
  61. c21560b Remove pbos@webrtc.org from autoroll TBRs. by Peter Boström · 8 years ago
  62. 9b5306c Adding test for unordered, fragmented SCTP message delivery. by Taylor Brandstetter · 8 years ago
  63. b5b3090 Corrected the testvectors for the level controller by peah · 8 years ago
  64. 8df4d0e Add playout_delay_oracle_unittest as gn target by isheriff · 8 years ago
  65. 3a11933 Remove audio_device_test_func. by maxmorin · 8 years ago
  66. 644fa96 Added recording of the configuration for the AudioFrame API call by peah · 8 years ago
  67. 7320866 Reland of Adding audio to video_quality_test. by minyue · 8 years ago
  68. 2b61639 Remove TMMBRSet class by danilchap · 8 years ago
  69. e1f5b4a voice_engine: Removed old variants of Channel constructor and CreateChannel by ossu · 8 years ago
  70. 38d840c NetEq: Changing checked_cast to saturated_cast by henrik.lundin · 8 years ago
  71. 96bbdd5 WebRtcSpl_SynthesisQMF: Fix UBSan fuzzer bug (left shift of negative value) by kwiberg · 8 years ago
  72. e9a6acf Added missing unittest to the modules/BUILD.gn build file by peah · 8 years ago
  73. cb2d701 Add kjellander as BUILD.gn OWNER in webrtc/modules by kjellander · 8 years ago
  74. 71fead2 Reland of StartTimestamp generated randomly in RtpSender constructor (patchset #1 id:1 of https://codereview.webrtc.org/2248413002/ ) by danilchap · 8 years ago
  75. d4e9f62 Updated AudioDecoderFactory to list AudioCodecSpecs instead of SdpAudioFormats. by ossu · 8 years ago
  76. 235020d Roll chromium_revision 915e47250f..e3860bd297 (412201:412289) by magjed · 8 years ago
  77. 010f092 GN: Add Android support to video_engine_tests. by sakal · 8 years ago
  78. fd16da2 Do not switch to a high-cost connection that is not receiving. by Honghai Zhang · 8 years ago
  79. 41a3287 Nil out EAGLContext explicitly on RTCEAGLVideoView dealloc. by tkchin · 8 years ago
  80. 869dab7 Disable Intel VP8 HW encoder. by Alex Glaznev · 8 years ago
  81. 6a35590 Add code for dummy file audio to fallback to dummy audio. by noahric · 8 years ago
  82. 7c0f8ee Avoid null pointer exception if Android getCameraInfo fails. by Alex Glaznev · 8 years ago
  83. d8a72f0 Close input file in FileAudioDevice::StopRecording. by noahric · 8 years ago
  84. 78810b6 Expose media constraint string constants as ObjC NSStrings by magjed · 8 years ago
  85. d22854b FilePlayer: Remove unused default values for arguments by kwiberg · 8 years ago
  86. 4a42900 Removes redundant log warning in WebRtcAudioManager. by henrika · 8 years ago
  87. 86c9694 Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ ) by danilchap · 8 years ago
  88. 5a25d95 FileRecorder + FilePlayer: Let Create functions return unique_ptr by kwiberg · 8 years ago
  89. 4466782 StartTimestamp generated randomly in RtpSender constructor by Danil Chapovalov · 8 years ago
  90. 2ae1fb6 Fix get_landmines.py script. by ehmaldonado · 8 years ago
  91. 144dd27 FileRecorderImpl and FilePlayerImpl don't need their own .h and .cc files by kwiberg · 8 years ago
  92. c54071d WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. by ossu · 8 years ago
  93. a93d5ac Don't simulate probing based on rtc event logs since we don't have that info logged. by stefan · 8 years ago
  94. eb680ea CongestionController::SetBweBitrates may now trigger probing. by philipel · 8 years ago
  95. c594aa61 Add a gyp/gn option to use dummy audio file devices. by noahric · 8 years ago
  96. e05bcc2 Do not switch a connection if the new connection is not ready to send packets. by Honghai Zhang · 8 years ago
  97. 49c01d7 Currently there is not way to programmically test whether a ScreenCapturer by zijiehe · 8 years ago
  98. 895e1a9 Change the default backup connection ping interval to 25 seconds. by Honghai Zhang · 8 years ago
  99. 287e548 Cleanup RtcpReceiver::TMMBRReceived function by danilchap · 8 years ago
  100. f095012 Revert of Adding audio to video_quality_test. (patchset #10 id:230001 of https://codereview.webrtc.org/2136573002/ ) by minyue · 8 years ago