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gerrit-public.fairphone.software / platform / external / webrtc / 599df8523382e0687f09dcd966d71f7cf9c9d058
  1. 599df85 Resolve cyclic dependency in remote bitrate estimator by Danil Chapovalov · 7 years ago
  2. fb08994 Adding time profiling support to AudioFrame by henrika · 7 years ago
  3. b22184e Roll chromium_revision 533744aa77..29ce025c90 (504014:504034) by Autoroller · 7 years ago
  4. e21be1d Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ ) by philipel · 7 years ago
  5. b0573bc Reorganize config of RTP header extensions for video receive streams. by Niels Möller · 7 years ago
  6. 0bf6071 Remove internal methods from global namespace. by Kári Tristan Helgason · 7 years ago
  7. ac0a503 NetEq/Stats: Don't let concealed_samples decrease by Henrik Lundin · 7 years ago
  8. b4aeb5b Roll chromium_revision 300d540fe1..533744aa77 (503992:504014) by Autoroller · 7 years ago
  9. 5ab6854 Revert "Remove remaining mentions of gflags" by Oleh Prypin · 7 years ago
  10. 90ce84e Remove remaining mentions of gflags by Oleh Prypin · 7 years ago
  11. 5c53641 Roll chromium_revision 7877db2873..300d540fe1 (503976:503992) by Autoroller · 7 years ago
  12. d93ffb0 Roll chromium_revision c91bb42405..7877db2873 (503967:503976) by Autoroller · 7 years ago
  13. e9ad322 Roll chromium_revision 569732c0a9..c91bb42405 (503964:503967) by Autoroller · 7 years ago
  14. fc63a7f Roll chromium_revision fe71bbc1fb..569732c0a9 (503807:503964) by Autoroller · 7 years ago
  15. 86021d9 Roll chromium_revision 008fb813c3..fe71bbc1fb (503673:503807) + Android fix by Oleh Prypin · 7 years ago
  16. b3547fa Revert "Added logging inside AEC3 for render API buffer under/overruns" by Per Åhgren · 7 years ago
  17. 835cc0c Remove unnecessary audio references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
  18. 73c8175 Fix autoroller script. by Edward Lemur · 7 years ago
  19. 2397b9a Remove voe::OutputMixer and AudioConferenceMixer. by solenberg · 7 years ago
  20. 4652e86 Disable flaky AudioStats.NoLoss test. by solenberg · 7 years ago
  21. c3d0da0 Avoids crash in AudioTrack when audio starts in background mode by henrika · 7 years ago
  22. 2c30120 Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ ) by brandtr · 7 years ago
  23. 55c7ede VideoProcessorIntegrationTest: Group member variables into two structs containing target/actual rates. by asapersson · 7 years ago
  24. 22d3da9 Delete unused Atomic32 overloads of binary +/- operators. by Niels Möller · 7 years ago
  25. 2cefac6 Add full stack tests for MediaCodec encoder. by brandtr · 7 years ago
  26. 433b11e Roll chromium_revision d4c912f989..008fb813c3 (503644:503673) by Autoroller · 7 years ago
  27. 7cd28b9 Set protected_by_flexfec flag properly in tests. by brandtr · 7 years ago
  28. 44eb7a2 Roll chromium_revision cbba184f42..d4c912f989 (503620:503644) by Autoroller · 7 years ago
  29. 401a52b Roll chromium_revision aa5131c67a..cbba184f42 (503572:503620) by Autoroller · 7 years ago
  30. ec5493e Roll chromium_revision 5aca6dd699..aa5131c67a (503520:503572) by Autoroller · 7 years ago
  31. 73b60b8 Remove the redundant method GetPayloadSpecifics by Karl Wiberg · 7 years ago
  32. 736a98a Roll chromium_revision 0b04bb65d6..5aca6dd699 (503455:503520) by Autoroller · 7 years ago
  33. 92d9dd0 rtp_rtcp_format: Separate public and private source files by Karl Wiberg · 7 years ago
  34. 8e8da13 Roll chromium_revision e242527ec0..0b04bb65d6 (503426:503455) by Autoroller · 7 years ago
  35. e10c01c Roll chromium_revision 6995ce043e..e242527ec0 (503371:503426) by Autoroller · 7 years ago
  36. 946d886 Remove VoENetwork by solenberg · 7 years ago
  37. 99a81b6 Remove #include of rtc_stream_config.h from rtc_event_log.h by Elad Alon · 7 years ago
  38. fc17930 Roll chromium_revision 2d1b3e73b6..6995ce043e (503346:503371) by Autoroller · 7 years ago
  39. 7a2b370 Roll chromium_revision d49a98a3ca..2d1b3e73b6 (503300:503346) by Autoroller · 7 years ago
  40. aa0fe12 Roll chromium_revision f52ff80530..d49a98a3ca (501932:503300) by Autoroller · 7 years ago
  41. 4e2deab Only return stats for the most recent unsignaled audio stream. by deadbeef · 7 years ago
  42. feeb9bf Remove backwards compatibilty header for Optional by kwiberg · 7 years ago
  43. 851481c Revert "Roll chromium_revision f52ff80530..4bafa509ea (501932:502960)" by Charu Jain · 7 years ago
  44. e682935 Roll chromium_revision f52ff80530..4bafa509ea (501932:502960) by Autoroller · 7 years ago
  45. b19012e Remove the support of fallback from DTLS to SDES. by zhihuang · 7 years ago
  46. b335e31 This is a rollback of https://chromium-review.googlesource.com/c/external/webrtc/+/616724 by alexnarest · 7 years ago
  47. 75dd6d4 Drop reference to webrtc_overrides/.../task_queue.h by nisse · 7 years ago
  48. 4dc4e25 Reland "Adding PRESUBMIT check to avoid mixing C, C++ and Objc-C/Obj-C++."" by Mirko Bonadei · 7 years ago
  49. 080832e Moving Obj-C++ code in desktop_capture_objc. by Mirko Bonadei · 7 years ago
  50. 034a6b8 Revert "Adding PRESUBMIT check to avoid mixing C, C++ and Objc-C/Obj-C++." by Mirko Bonadei · 7 years ago
  51. 0c15c53 Adding PRESUBMIT check to avoid mixing C, C++ and Objc-C/Obj-C++. by Mirko Bonadei · 7 years ago
  52. 7de1eb7 Adding Test at the end of test classes. by Mirko Bonadei · 7 years ago
  53. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  54. 2572404 Removing useless include_dirs entry. by Mirko Bonadei · 7 years ago
  55. a5f043f Change ForwardErrorCorrection class to accept one received packet at a time. by nisse · 7 years ago
  56. dd3abbb Remove VoERTP_RTCP. by solenberg · 7 years ago
  57. c5267d2 Simplify ReceiveStatistics: merge GetActiveStatisticians into RtcpReportBlocks by Danil Chapovalov · 7 years ago
  58. aa568a6 Android: Add interface for getting native EGL context by Magnus Jedvert · 7 years ago
  59. 6dc2038 Remove VoECodec. by solenberg · 7 years ago
  60. 6ac1552 Cleaning checkdeps configuration. by Mirko Bonadei · 7 years ago
  61. a730c1c Enabling and fixing CheckNewlineAtTheEndOfProtoFiles by Mirko Bonadei · 7 years ago
  62. cb728ea Fix Gn Untracked headers in webrtc/modules/video_coding. by charujain · 7 years ago
  63. b63310a Remove VoEFile and things it uses. by solenberg · 7 years ago
  64. 2352ce3 Remove backwards compatibilty header for ArrayView by kwiberg · 7 years ago
  65. ef6ee98 Pass environment variable as string in autoroll script. by Edward Lemur · 7 years ago
  66. 1d4db39 Revert of If SRTP sessions exist, don't create new ones when applying answer. (patchset #1 id:1 of https://codereview.webrtc.org/3019443002/ ) by henrika · 7 years ago
  67. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  68. 35dee81 Clean out unused methods from VoiceEngine and VoEBase. by solenberg · 7 years ago
  69. 2c9ac29 Autoroller: don't use GCE auth pathway by Aaron Gable · 7 years ago
  70. 4354725 Delete deprecated metod RtpRtcp::SetMaxTransferUnit. by nisse · 7 years ago
  71. cb3b1c1 video_quality_loopback_test.py: Fix relative path to root. by Henrik Kjellander · 7 years ago
  72. 2782904 RtcEventLog::Create() no longer a friend of RtcEventLogImpl by Elad Alon · 7 years ago
  73. d45aea8 Serialize "a=x-google-flag:conference". by deadbeef · 7 years ago
  74. 5ada7ac If SRTP sessions exist, don't create new ones when applying answer. by deadbeef · 7 years ago
  75. 1c5e6d0 Remove BasicPortAllocator::EnableProtocol. by deadbeef · 7 years ago
  76. 7f1563f Making BasicPortAllocator tests slightly less fragile. by deadbeef · 7 years ago
  77. 930021d Eliminating the risk of sustained echo during capture data loss in AEC3. by Per Åhgren · 7 years ago
  78. f491c52 Move log_count_ out of RtcEventLogImpl by Elad Alon · 7 years ago
  79. a96fd7f Make rtc_event_log2text handle all events [2/2] by Elad Alon · 7 years ago
  80. a7567a9 Implement DesktopCapturerWrapper and CaptureResultDesktopCapturerWrapper by Zijie He · 7 years ago
  81. 6c17057 Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target by Danil Chapovalov · 7 years ago
  82. 58b0316 Expose new video codec factories in the PeerConnectionFactory API by Magnus Jedvert · 7 years ago
  83. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago
  84. 8b891ce Remove unused function - RtcEventLog::ParseRtcEventLog() by Elad Alon · 7 years ago
  85. 8ecfd80 Fix public Obj-C header import. by Anders Carlsson · 7 years ago
  86. 8d75ac7 Add stats for forced software encoder fallback for VP8. by asapersson · 7 years ago
  87. 34f303c Make rtc_event_log2text handle all events [1/2] by Elad Alon · 7 years ago
  88. 48d96c0 Corrected upper limits of NetEq minimum and maximum delay. by Gustaf Ullberg · 7 years ago
  89. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  90. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  91. 743117f Disable FullStackTest.LargeRoomVP8_*thumb on iOS by oprypin · 7 years ago
  92. 262d4ff Added logging inside AEC3 for render API buffer under/overruns by Per Åhgren · 7 years ago
  93. 6ef2002 Use current user instead of kjellander@ in the roll script. by Edward Lemur · 7 years ago
  94. 9d8ce7c video_quality_loopback_test.py: Fix paths to root. by Henrik Kjellander · 7 years ago
  95. 866d337 Re-enabling CheckOrphanHeaders by Mirko Bonadei · 7 years ago
  96. 9a45116 Fix Gn Untracked headers in webrtc/common_audio by charujain · 7 years ago
  97. 59b2854 gitignore: Remove webrtc/ by Henrik Kjellander · 7 years ago
  98. a18a3bf Fix Python shebang and license for presubmit_test.py by Henrik Kjellander · 7 years ago
  99. 5a6aa4f Fix path to root in low_bandwidth_audio_test.py by Henrik Kjellander · 7 years ago
  100. e4f900d gradle_project_test.py: Fix calculation of checkout root dir. by Henrik Kjellander · 7 years ago