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webrtc / src / b63310a25690842a40b7cf31f1664d25831d0c70
  1. b63310a Remove VoEFile and things it uses. by solenberg · 8 years ago
  2. 2352ce3 Remove backwards compatibilty header for ArrayView by kwiberg · 8 years ago
  3. ef6ee98 Pass environment variable as string in autoroll script. by Edward Lemur · 8 years ago
  4. 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 · 8 years ago
  5. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 8 years ago
  6. 35dee81 Clean out unused methods from VoiceEngine and VoEBase. by solenberg · 8 years ago
  7. 2c9ac29 Autoroller: don't use GCE auth pathway by Aaron Gable · 8 years ago
  8. 4354725 Delete deprecated metod RtpRtcp::SetMaxTransferUnit. by nisse · 8 years ago
  9. cb3b1c1 video_quality_loopback_test.py: Fix relative path to root. by Henrik Kjellander · 8 years ago
  10. 2782904 RtcEventLog::Create() no longer a friend of RtcEventLogImpl by Elad Alon · 8 years ago
  11. d45aea8 Serialize "a=x-google-flag:conference". by deadbeef · 8 years ago
  12. 5ada7ac If SRTP sessions exist, don't create new ones when applying answer. by deadbeef · 8 years ago
  13. 1c5e6d0 Remove BasicPortAllocator::EnableProtocol. by deadbeef · 8 years ago
  14. 7f1563f Making BasicPortAllocator tests slightly less fragile. by deadbeef · 8 years ago
  15. 930021d Eliminating the risk of sustained echo during capture data loss in AEC3. by Per Åhgren · 8 years ago
  16. f491c52 Move log_count_ out of RtcEventLogImpl by Elad Alon · 8 years ago
  17. a96fd7f Make rtc_event_log2text handle all events [2/2] by Elad Alon · 8 years ago
  18. a7567a9 Implement DesktopCapturerWrapper and CaptureResultDesktopCapturerWrapper by Zijie He · 8 years ago
  19. 6c17057 Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target by Danil Chapovalov · 8 years ago
  20. 58b0316 Expose new video codec factories in the PeerConnectionFactory API by Magnus Jedvert · 8 years ago
  21. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 8 years ago
  22. 8b891ce Remove unused function - RtcEventLog::ParseRtcEventLog() by Elad Alon · 8 years ago
  23. 8ecfd80 Fix public Obj-C header import. by Anders Carlsson · 8 years ago
  24. 8d75ac7 Add stats for forced software encoder fallback for VP8. by asapersson · 8 years ago
  25. 34f303c Make rtc_event_log2text handle all events [1/2] by Elad Alon · 8 years ago
  26. 48d96c0 Corrected upper limits of NetEq minimum and maximum delay. by Gustaf Ullberg · 8 years ago
  27. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 8 years ago
  28. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 8 years ago
  29. 743117f Disable FullStackTest.LargeRoomVP8_*thumb on iOS by oprypin · 8 years ago
  30. 262d4ff Added logging inside AEC3 for render API buffer under/overruns by Per Åhgren · 8 years ago
  31. 6ef2002 Use current user instead of kjellander@ in the roll script. by Edward Lemur · 8 years ago
  32. 9d8ce7c video_quality_loopback_test.py: Fix paths to root. by Henrik Kjellander · 8 years ago
  33. 866d337 Re-enabling CheckOrphanHeaders by Mirko Bonadei · 8 years ago
  34. 9a45116 Fix Gn Untracked headers in webrtc/common_audio by charujain · 8 years ago
  35. 59b2854 gitignore: Remove webrtc/ by Henrik Kjellander · 8 years ago
  36. a18a3bf Fix Python shebang and license for presubmit_test.py by Henrik Kjellander · 8 years ago
  37. 5a6aa4f Fix path to root in low_bandwidth_audio_test.py by Henrik Kjellander · 8 years ago
  38. e4f900d gradle_project_test.py: Fix calculation of checkout root dir. by Henrik Kjellander · 8 years ago
  39. 0e8f053 iOS: Fix license path in API framework script by Henrik Kjellander · 8 years ago
  40. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  41. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago
  42. 6674846 Fix Gn untracked headers in webrtc/media by charujain · 8 years ago
  43. 8b92dce Whitespace change to test rietveld cq by agable · 8 years ago
  44. 2a5e426 Reject the descriptions that attempt to change the order of m= sections by Zhi Huang · 8 years ago
  45. 9c66aee Adding some checkdeps rules by Mirko Bonadei · 8 years ago
  46. 9cad501 Implement IsOccluded() in each DesktopCapturer implementation by Zijie He · 8 years ago
  47. 6564ea3 FallbackDesktopCapturerWrapper should set permanent_error if main_capturer_->SelectSource() failed by Zijie He · 8 years ago
  48. 6d01f6d Fix PRESUBMIT.py by Edward Lemur · 8 years ago
  49. f1977b3 Roll chromium_revision 946d80b37f..f52ff80530 (501605:501932) by Edward Lemur · 8 years ago
  50. 8769e17 Rename rtp::Packet to RtpPacket by Danil Chapovalov · 8 years ago
  51. eea063f Reject configs with ridiculously many channels instead of crashing by Karl Wiberg · 8 years ago
  52. 033a1bf Update autoroller to use Gerrit. by Edward Lemur · 8 years ago
  53. 73276ad - Removes voe_conference_test. by Fredrik Solenberg · 8 years ago
  54. 7d1f493 Make toVideoFrame in I420Frame return the original frame. by Sami Kalliomäki · 8 years ago
  55. 9893e25 PRESUBMIT: Enforce tracker prefix for all BUG entries by charujain · 8 years ago
  56. 66ca7e3 Add a check that sanitizers cause a fatal failure by Oleh Prypin · 8 years ago
  57. 0a2ed5f Removing webrtc/config.h by Mirko Bonadei · 8 years ago
  58. d4b0c05 Add new video codec factories by Magnus Jedvert · 8 years ago
  59. a9d4da8 Update URLs for deps/third_party/* in DEPS by Henrik Kjellander · 8 years ago
  60. 2db9c28 Update README.md and codereview.settings for new source location by Aaron Gable · 8 years ago
  61. 80c1f12 Whitespace change. by Edward Lemur · 8 years ago
  62. 866e417 Update webrtc CQ to commit to new location by Aaron Gable · 8 years ago
  63. 479d3d7 Drop return value from RtpRtcp::IncomingRtcpPacket. by nisse · 8 years ago
  64. 9a85f07 Enhance RTCUIApplicationStatusObserver thread safety. by andersc · 8 years ago
  65. f54573b Reland of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #1 id:1 of https://codereview.webrtc.org/3011093002/ ) by nisse · 8 years ago
  66. 4e15e67 Android AppRTCMobile: Transition local render to new VideoSink interface by magjed · 8 years ago
  67. c434e96 Roll chromium_revision 2107279323..946d80b37f (501577:501605) by buildbot · 8 years ago
  68. fb2fa3f Fixed the overflow in the AGC by peah · 8 years ago
  69. 2ab9879 Android: Improve handling of RGB texture frames by magjed · 8 years ago
  70. dc80abe Reland of move deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3010553002/ ) by kthelgason · 8 years ago
  71. d57f9dd Separate build targets for APM bit exactness tools from unittests. by Sam Zackrisson · 8 years ago
  72. 6ac77aa Roll chromium_revision 0ff0ce95c4..2107279323 (501558:501577) by buildbot · 8 years ago
  73. 772bd8b Fix no_size_t_to_int_warning in rtp_rtcp:rtp_rtcp_format target by danilchap · 8 years ago
  74. 87443ee Make rtp::Packet's destructor and constructors public by eladalon · 8 years ago
  75. 4090f38 Fix retain cycles in RTCUIApplicationStatusObserver. by andersc · 8 years ago
  76. 47791cf Remove definition of thread annotation macros without RTC_ prefix by danilchap · 8 years ago
  77. c33ab4a Roll chromium_revision 4e1bce7498..0ff0ce95c4 (501489:501558) by buildbot · 8 years ago
  78. 6d64e9a Remove JsepSessionDescription's string Initialize method by Steve Anton · 8 years ago
  79. db45ca8 Change PeerConnection test helpers to take unique_ptr by Steve Anton · 8 years ago
  80. 6ba5e2c Roll chromium_revision 2cb11be9c1..4e1bce7498 (501411:501489) by buildbot · 8 years ago
  81. 1da9ad2 Roll chromium_revision 988ef967a4..2cb11be9c1 (501290:501411) by buildbot · 8 years ago
  82. 43697f6 Add javadoc comment for PeerConnection.dispose. by deadbeef · 8 years ago
  83. e997381 Move reencode logic for screenshare bitrate overshoot from generic by ilnik · 8 years ago
  84. 005b5ec Roll chromium_revision 60110d6972..988ef967a4 (501246:501290) by buildbot · 8 years ago
  85. 09f4481 Break rtp_rtcp_format out of rtp_rtcp, to resolve circular dependencies by danilchap · 8 years ago
  86. 3fe3e3b Add assignment operator to AudioEncoderRuntimeConfig by eladalon · 8 years ago
  87. 5cec256 Roll chromium_revision c06541dae0..60110d6972 (501227:501246) by buildbot · 8 years ago
  88. c3d2bfd Remove no- prefix from command line flags in rtc_event_log2text and rtc_event_log2rtp_dump and negate their meaning. by terelius · 8 years ago
  89. 661d949 Only use BBRs pacer if the unit test is running BBR. by terelius · 8 years ago
  90. 0c011d9 Make sure send and receive deltas are positive for remote estimated probe clusters. by philipel · 8 years ago
  91. bcc2176 Decoupling audio_device from Obj-C code by mbonadei · 8 years ago
  92. 2475ae2 Simplify passing video coded factories in media engine by magjed · 8 years ago
  93. 1e7dd31 Break the ANA build-target into ANA and ANA-config by eladalon · 8 years ago
  94. adfc279 Roll chromium_revision f2668a840f..c06541dae0 (501203:501227) by buildbot · 8 years ago
  95. 43467b0 ObjC EncodedImage: Use fixed width integer types by magjed · 8 years ago
  96. 9a2d5c9 Roll chromium_revision 1c19443a45..f2668a840f (501170:501203) by buildbot · 8 years ago
  97. d10bed1 Roll chromium_revision af161574c2..1c19443a45 (501126:501170) by buildbot · 8 years ago
  98. 22581d4 Roll chromium_revision 0e2fc3a0b9..af161574c2 (501051:501126) by buildbot · 8 years ago
  99. 2597889 Roll chromium_revision 307e855af4..0e2fc3a0b9 (500986:501051) by buildbot · 8 years ago
  100. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 8 years ago