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gerrit-public.fairphone.software / platform / external / webrtc / 034a6b8a4cdf151ae7377c87c5b2b6156d658173
  1. 034a6b8 Revert "Adding PRESUBMIT check to avoid mixing C, C++ and Objc-C/Obj-C++." by Mirko Bonadei · 7 years ago
  2. 0c15c53 Adding PRESUBMIT check to avoid mixing C, C++ and Objc-C/Obj-C++. by Mirko Bonadei · 7 years ago
  3. 7de1eb7 Adding Test at the end of test classes. by Mirko Bonadei · 7 years ago
  4. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  5. 2572404 Removing useless include_dirs entry. by Mirko Bonadei · 7 years ago
  6. a5f043f Change ForwardErrorCorrection class to accept one received packet at a time. by nisse · 7 years ago
  7. dd3abbb Remove VoERTP_RTCP. by solenberg · 7 years ago
  8. c5267d2 Simplify ReceiveStatistics: merge GetActiveStatisticians into RtcpReportBlocks by Danil Chapovalov · 7 years ago
  9. aa568a6 Android: Add interface for getting native EGL context by Magnus Jedvert · 7 years ago
  10. 6dc2038 Remove VoECodec. by solenberg · 7 years ago
  11. 6ac1552 Cleaning checkdeps configuration. by Mirko Bonadei · 7 years ago
  12. a730c1c Enabling and fixing CheckNewlineAtTheEndOfProtoFiles by Mirko Bonadei · 7 years ago
  13. cb728ea Fix Gn Untracked headers in webrtc/modules/video_coding. by charujain · 7 years ago
  14. b63310a Remove VoEFile and things it uses. by solenberg · 7 years ago
  15. 2352ce3 Remove backwards compatibilty header for ArrayView by kwiberg · 7 years ago
  16. ef6ee98 Pass environment variable as string in autoroll script. by Edward Lemur · 7 years ago
  17. 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
  18. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  19. 35dee81 Clean out unused methods from VoiceEngine and VoEBase. by solenberg · 7 years ago
  20. 2c9ac29 Autoroller: don't use GCE auth pathway by Aaron Gable · 7 years ago
  21. 4354725 Delete deprecated metod RtpRtcp::SetMaxTransferUnit. by nisse · 7 years ago
  22. cb3b1c1 video_quality_loopback_test.py: Fix relative path to root. by Henrik Kjellander · 7 years ago
  23. 2782904 RtcEventLog::Create() no longer a friend of RtcEventLogImpl by Elad Alon · 7 years ago
  24. d45aea8 Serialize "a=x-google-flag:conference". by deadbeef · 7 years ago
  25. 5ada7ac If SRTP sessions exist, don't create new ones when applying answer. by deadbeef · 7 years ago
  26. 1c5e6d0 Remove BasicPortAllocator::EnableProtocol. by deadbeef · 7 years ago
  27. 7f1563f Making BasicPortAllocator tests slightly less fragile. by deadbeef · 7 years ago
  28. 930021d Eliminating the risk of sustained echo during capture data loss in AEC3. by Per Åhgren · 7 years ago
  29. f491c52 Move log_count_ out of RtcEventLogImpl by Elad Alon · 7 years ago
  30. a96fd7f Make rtc_event_log2text handle all events [2/2] by Elad Alon · 7 years ago
  31. a7567a9 Implement DesktopCapturerWrapper and CaptureResultDesktopCapturerWrapper by Zijie He · 7 years ago
  32. 6c17057 Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target by Danil Chapovalov · 7 years ago
  33. 58b0316 Expose new video codec factories in the PeerConnectionFactory API by Magnus Jedvert · 7 years ago
  34. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago
  35. 8b891ce Remove unused function - RtcEventLog::ParseRtcEventLog() by Elad Alon · 7 years ago
  36. 8ecfd80 Fix public Obj-C header import. by Anders Carlsson · 7 years ago
  37. 8d75ac7 Add stats for forced software encoder fallback for VP8. by asapersson · 7 years ago
  38. 34f303c Make rtc_event_log2text handle all events [1/2] by Elad Alon · 7 years ago
  39. 48d96c0 Corrected upper limits of NetEq minimum and maximum delay. by Gustaf Ullberg · 7 years ago
  40. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  41. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  42. 743117f Disable FullStackTest.LargeRoomVP8_*thumb on iOS by oprypin · 7 years ago
  43. 262d4ff Added logging inside AEC3 for render API buffer under/overruns by Per Åhgren · 7 years ago
  44. 6ef2002 Use current user instead of kjellander@ in the roll script. by Edward Lemur · 7 years ago
  45. 9d8ce7c video_quality_loopback_test.py: Fix paths to root. by Henrik Kjellander · 7 years ago
  46. 866d337 Re-enabling CheckOrphanHeaders by Mirko Bonadei · 7 years ago
  47. 9a45116 Fix Gn Untracked headers in webrtc/common_audio by charujain · 7 years ago
  48. 59b2854 gitignore: Remove webrtc/ by Henrik Kjellander · 7 years ago
  49. a18a3bf Fix Python shebang and license for presubmit_test.py by Henrik Kjellander · 7 years ago
  50. 5a6aa4f Fix path to root in low_bandwidth_audio_test.py by Henrik Kjellander · 7 years ago
  51. e4f900d gradle_project_test.py: Fix calculation of checkout root dir. by Henrik Kjellander · 7 years ago
  52. 0e8f053 iOS: Fix license path in API framework script by Henrik Kjellander · 7 years ago
  53. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  54. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago
  55. 6674846 Fix Gn untracked headers in webrtc/media by charujain · 7 years ago
  56. 8b92dce Whitespace change to test rietveld cq by agable · 7 years ago
  57. 2a5e426 Reject the descriptions that attempt to change the order of m= sections by Zhi Huang · 7 years ago
  58. 9c66aee Adding some checkdeps rules by Mirko Bonadei · 7 years ago
  59. 9cad501 Implement IsOccluded() in each DesktopCapturer implementation by Zijie He · 7 years ago
  60. 6564ea3 FallbackDesktopCapturerWrapper should set permanent_error if main_capturer_->SelectSource() failed by Zijie He · 7 years ago
  61. 6d01f6d Fix PRESUBMIT.py by Edward Lemur · 7 years ago
  62. f1977b3 Roll chromium_revision 946d80b37f..f52ff80530 (501605:501932) by Edward Lemur · 7 years ago
  63. 8769e17 Rename rtp::Packet to RtpPacket by Danil Chapovalov · 7 years ago
  64. eea063f Reject configs with ridiculously many channels instead of crashing by Karl Wiberg · 7 years ago
  65. 033a1bf Update autoroller to use Gerrit. by Edward Lemur · 7 years ago
  66. 73276ad - Removes voe_conference_test. by Fredrik Solenberg · 7 years ago
  67. 7d1f493 Make toVideoFrame in I420Frame return the original frame. by Sami Kalliomäki · 7 years ago
  68. 9893e25 PRESUBMIT: Enforce tracker prefix for all BUG entries by charujain · 7 years ago
  69. 66ca7e3 Add a check that sanitizers cause a fatal failure by Oleh Prypin · 7 years ago
  70. 0a2ed5f Removing webrtc/config.h by Mirko Bonadei · 7 years ago
  71. d4b0c05 Add new video codec factories by Magnus Jedvert · 7 years ago
  72. a9d4da8 Update URLs for deps/third_party/* in DEPS by Henrik Kjellander · 7 years ago
  73. 2db9c28 Update README.md and codereview.settings for new source location by Aaron Gable · 7 years ago
  74. 80c1f12 Whitespace change. by Edward Lemur · 7 years ago
  75. 866e417 Update webrtc CQ to commit to new location by Aaron Gable · 7 years ago
  76. 479d3d7 Drop return value from RtpRtcp::IncomingRtcpPacket. by nisse · 7 years ago
  77. 9a85f07 Enhance RTCUIApplicationStatusObserver thread safety. by andersc · 7 years ago
  78. f54573b Reland of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #1 id:1 of https://codereview.webrtc.org/3011093002/ ) by nisse · 7 years ago
  79. 4e15e67 Android AppRTCMobile: Transition local render to new VideoSink interface by magjed · 7 years ago
  80. c434e96 Roll chromium_revision 2107279323..946d80b37f (501577:501605) by buildbot · 7 years ago
  81. fb2fa3f Fixed the overflow in the AGC by peah · 7 years ago
  82. 2ab9879 Android: Improve handling of RGB texture frames by magjed · 7 years ago
  83. dc80abe Reland of move deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3010553002/ ) by kthelgason · 7 years ago
  84. d57f9dd Separate build targets for APM bit exactness tools from unittests. by Sam Zackrisson · 7 years ago
  85. 6ac77aa Roll chromium_revision 0ff0ce95c4..2107279323 (501558:501577) by buildbot · 7 years ago
  86. 772bd8b Fix no_size_t_to_int_warning in rtp_rtcp:rtp_rtcp_format target by danilchap · 7 years ago
  87. 87443ee Make rtp::Packet's destructor and constructors public by eladalon · 7 years ago
  88. 4090f38 Fix retain cycles in RTCUIApplicationStatusObserver. by andersc · 7 years ago
  89. 47791cf Remove definition of thread annotation macros without RTC_ prefix by danilchap · 7 years ago
  90. c33ab4a Roll chromium_revision 4e1bce7498..0ff0ce95c4 (501489:501558) by buildbot · 7 years ago
  91. 6d64e9a Remove JsepSessionDescription's string Initialize method by Steve Anton · 7 years ago
  92. db45ca8 Change PeerConnection test helpers to take unique_ptr by Steve Anton · 7 years ago
  93. 6ba5e2c Roll chromium_revision 2cb11be9c1..4e1bce7498 (501411:501489) by buildbot · 7 years ago
  94. 1da9ad2 Roll chromium_revision 988ef967a4..2cb11be9c1 (501290:501411) by buildbot · 7 years ago
  95. 43697f6 Add javadoc comment for PeerConnection.dispose. by deadbeef · 7 years ago
  96. e997381 Move reencode logic for screenshare bitrate overshoot from generic by ilnik · 7 years ago
  97. 005b5ec Roll chromium_revision 60110d6972..988ef967a4 (501246:501290) by buildbot · 7 years ago
  98. 09f4481 Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies by danilchap · 7 years ago
  99. 3fe3e3b Add assignment operator to AudioEncoderRuntimeConfig by eladalon · 7 years ago
  100. 5cec256 Roll chromium_revision c06541dae0..60110d6972 (501227:501246) by buildbot · 7 years ago