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gerrit-public.fairphone.software / platform / external / webrtc / 4bc4d2747b297b3d770fe4bb2d6bee0786d2164a
  1. 4bc4d27 GN: Fix Windows Clang errors by ehmaldonado · 8 years ago
  2. 3f746ea Fix error when accumulating floats in an int. by maxmorin · 8 years ago
  3. e29352b Refactor certificate stats collection, added SSLCertificateStats. by hbos · 8 years ago
  4. 2ab012c Implement CVO for iOS capturer by magjed · 8 years ago
  5. 19319a3 Add missing "//build/config/sanitizers:deps" to executable targets. by ehmaldonado · 8 years ago
  6. 00e45bb Move InsertZeroColumns and CopyColumn to ::internal. by brandtr · 8 years ago
  7. 7a770e0 GN build rules for four audio processing test executables by kwiberg · 8 years ago
  8. 8a6a600 Make neteq_rtpplay parse RTP header extensions by henrik.lundin · 8 years ago
  9. 5f09980 Removed inline definitions and added destructors to fix chromium-style. by aleloi · 8 years ago
  10. 549d80b NetEq: only update current_rtp_payload_type_ when validated by henrik.lundin · 8 years ago
  11. fe8f489 Fix setting the MTU for SCTP. by deadbeef · 8 years ago
  12. b60a819 Fixing inconsistency with behavior of `ClearGettingPorts`. by deadbeef · 8 years ago
  13. 824f586 Fixing segfault caused by TurnServer. by deadbeef · 8 years ago
  14. 1d7a637 Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 8 years ago
  15. fcada90 Fixing timestamp comparison assert. by deadbeef · 8 years ago
  16. 36a06a9 Increase QP threshold for H.264 encoder QP based scaling. by glaznev · 8 years ago
  17. 1184025 Restart capture session if needed on active. by tkchin · 8 years ago
  18. 5fac3f0 NetEq: Don't check sample rate and frame size upon error by henrik.lundin · 8 years ago
  19. d1a10a0 Make FakeDecodeFromFile handle codec-internal CNG by henrik.lundin · 8 years ago
  20. f02207d MB: Flip Mac bots to GN by default. by kjellander · 8 years ago
  21. b0b0edb Roll chromium_revision e3860bd297..938114be1e (412289:414059) by ehmaldonado · 8 years ago
  22. 28a0ffd GN: Synchronize resources between Android and iOS. by kjellander · 8 years ago
  23. 2df32a3 GN: Override lsan and tsan suppression files. by ehmaldonado · 8 years ago
  24. 5f2e7c4 Added more targets to .gn. by aleloi · 8 years ago
  25. 2ec45b9 Make dependency of audio_device of ApplicationServices explicit. by maxmorin · 8 years ago
  26. 4e7e8d7 Now probe for x3 and x6 of the initial start bitrate and allow for faster receive bitrates when calculating probing estimates. by philipel · 8 years ago
  27. 2c670db Added GN target for webrtc_opus_fec_test. by ivoc · 8 years ago
  28. 7a0ff2f Disable examples for GYP Android bots. by ehmaldonado · 8 years ago
  29. 98468bb Revert of GN build rules for four audio processing test executables (patchset #3 id:40001 of https://codereview.webrtc.org/2267403003/ ) by sakal · 8 years ago
  30. 538b560 GN build rules for four audio processing test executables by kwiberg · 8 years ago
  31. 0561bdf Only use payload size within the know send/receive interval for probing calculations. by philipel · 8 years ago
  32. 619a211 iLBC: Handle a case of bad input data by kwiberg · 8 years ago
  33. 0aa9d18 Set send side bitrate estimate on successful probing attempt. by philipel · 8 years ago
  34. cd8ae61 Add missing dependencies to setup_links. by ehmaldonado · 8 years ago
  35. f944c35 GN: Add resources for webrtc_perf_tests on Android by kjellander · 8 years ago
  36. e51b41a Added GN target for isac_test. by ivoc · 8 years ago
  37. 5d167d6 Removals and renamings in the new audio mixer. by aleloi · 8 years ago
  38. 76f91cd Add ThreadChecker to the TimestampAligner class. by nisse · 8 years ago
  39. 665d181 Increased column width for python tool rtp_analyzer.py. by aleloi · 8 years ago
  40. 30be5d7 Updated mixer unittests and fixed a related bug in the new mixer. by aleloi · 8 years ago
  41. 615d301 RTCStats and RTCStatsReport added (webrtc/stats). by hbos · 8 years ago
  42. 616df1e Added a level indicator to new mixer. by aleloi · 8 years ago
  43. 1f77905 Remove outdated symlink by kthelgason · 8 years ago
  44. a53fa0a Fix AppRTC Android Demo GN build when is_component_build=true. by sakal · 8 years ago
  45. 4c8adb1 MB: Flip Android bots to GN by default. by kjellander · 8 years ago
  46. 24ee050 CQ: Remove android_arm64_rel trybot by kjellander · 8 years ago
  47. 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
  48. 6addf49 Adds function for computing moving average to visualization tool. by terelius · 8 years ago
  49. 5048f57 Add logs and small change in BasicPortAllocator. by Honghai Zhang · 8 years ago
  50. f99a9de ProbingEstimator: Erase history based on time threshold by Irfan Sheriff · 8 years ago
  51. 185ba29 Extract library from the RTCEventLog visualizer by skvlad · 8 years ago
  52. 5bed20f Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video. by Per · 8 years ago
  53. b37c45c GN: Add libjingle_peerconnection_java to peerconnection_unittests. by kjellander · 8 years ago
  54. a246cfb Don't include RTP headers in send-side BWE. by Stefan Holmer · 8 years ago
  55. 9a11784 Migrated GN target :g722_test by aleloi · 8 years ago
  56. 16f55a1 Migrated GN target :g711_test by aleloi · 8 years ago
  57. 649a21a Disable RampUpTest.UpDownUpThreeStreams. by philipel · 8 years ago
  58. 2e48646 RTC_CHECK and RTC_DCHECK macros for C by kwiberg · 8 years ago
  59. 7924697 Refactor WebRtcVideoCapturer. by nisse · 8 years ago
  60. d8dd190 GN: Fix test_support_unittests and MIPS compile issue. by kjellander · 8 years ago
  61. 84c03ba Add rtc_media as a direct dependency of rtc_media_unittests. by nisse · 8 years ago
  62. 0d1ad32 Add histogram for percentage of incoming frames that are limited in resolution due to cpu: by asapersson · 8 years ago
  63. 14cf12b Fixing TSan data race warning in video end-to-end tests. by Taylor Brandstetter · 8 years ago
  64. 23d947d Some cleanup in BaseChannel RTCP mux code. by deadbeef · 8 years ago
  65. b3f1c5d Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine by henrik.lundin · 8 years ago
  66. e131ea5 Adding deadbeef and honghaiz as owners of webrtc/pc. by deadbeef · 8 years ago
  67. 72a5645 Removed the deactivation of the level controller when there is a built-in AGC available by peah · 8 years ago
  68. 8c16520 Method to parse event log directly from a string. by terelius · 8 years ago
  69. 6c46eaa Add gtest as a dependency for neteq_quality_test_support. by ehmaldonado · 8 years ago
  70. d48717b Fix issue where the number of packets reported in tests/simulations sometimes are negative. by stefan · 8 years ago
  71. 4ec01d9 Fix trivial lint errors in FileRecorder and FilePlayer by kwiberg · 8 years ago
  72. 853ecb2 Style cleanup in UpdateTmmbr: by danilchap · 8 years ago
  73. 7f82fc9 WebRtcIlbcfix_Smooth: Fix UBSan fuzzer bug (left shift of 1 by 31 overflows) by kwiberg · 8 years ago
  74. 642e3bc [rtcp] TransportFeedback adjusted to match other rtcp packets: by danilchap · 8 years ago
  75. 4981051 [Reland] Cleanup of the AudioDeviceBuffer class. by henrika · 8 years ago
  76. 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
  77. 4381700 WebRtcVideoFrame constructor without transport_frame_id. by nisse · 8 years ago
  78. e5b4141 Move RTP timestamp calculation from BuildRTPheader to SendOutgoingData by danilchap · 8 years ago
  79. ff101d6 iOS: add PlistBuddy location to path to avoid build errors by vopatop.skam · 8 years ago
  80. 94b9199 Add a copy of gyp_flag_compare from Chromium to WebRTC's webrtc/tools. by ehmaldonado · 8 years ago
  81. 4905f06 Disable the software noise suppressor on iOS devices as that by peah · 8 years ago
  82. abcc3de Add pps id and sps id parsing to the h.264 depacketizer. by stefan · 8 years ago
  83. 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
  84. a7a01df Add field_trial_default dependency to libjingle_peerconnection by arlolra · 8 years ago
  85. 8177452 iOS H264VideoToolBoxEncoder: Stop scaling native CVPixelBuffers by magjed · 8 years ago
  86. d7a89db Revert of Cleanup of the AudioDeviceBuffer class (patchset #6 id:100001 of https://codereview.webrtc.org/2256833003/ ) by henrika · 8 years ago
  87. cf327b4 Cleanup of the AudioDeviceBuffer class. by henrika · 8 years ago
  88. da161d7 Reformat rtcp_receiver git cl format --full by danilchap · 8 years ago
  89. 861da3c Refactor neteq_test_support. by ehmaldonado · 8 years ago
  90. 294fb05 Add a timeout for starting the camera on CameraCapturer. by sakal · 8 years ago
  91. bcba64a GN: Add "//build/config/sanitizers:deps" as a dependency to executable targets. by ehmaldonado · 8 years ago
  92. 4a85abb Add support for more resolutions on iOS/macOS by kthelgason · 8 years ago
  93. ec5c906 GN: Fix errors when some variables are set to non-default values. by kjellander · 8 years ago
  94. 72333d2 Add kjellander@webrtc.org to more BUILD.gn OWNERS files. by kjellander · 8 years ago
  95. 96b6b83 iOS: add type to peer connection local streams by vopatop.skam · 8 years ago
  96. c21560b Remove pbos@webrtc.org from autoroll TBRs. by Peter Boström · 8 years ago
  97. 9b5306c Adding test for unordered, fragmented SCTP message delivery. by Taylor Brandstetter · 8 years ago
  98. b5b3090 Corrected the testvectors for the level controller by peah · 8 years ago
  99. 8df4d0e Add playout_delay_oracle_unittest as gn target by isheriff · 8 years ago
  100. 3a11933 Remove audio_device_test_func. by maxmorin · 8 years ago