Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / d873540101d51d3b8dd5dd486b381f88c8ea5c07
  1. d873540 Roll chromium 282462:282879. by fgalligan@google.com · 11 years ago
  2. 92a9bac Rebase webrtc/base with r6682 version of talk/base: by henrike@webrtc.org · 11 years ago
  3. 1b84116 Add a facility to the Thread class to catch blocking regressions. by henrike@webrtc.org · 11 years ago
  4. b038c72 Enable SCTP compile for iOS. by tkchin@webrtc.org · 11 years ago
  5. aac1497 (Auto)update libjingle 71116846-> 71117224 by buildbot@webrtc.org · 11 years ago
  6. 5be649f Add a facility to the Thread class to catch blocking regressions. by tommi@webrtc.org · 11 years ago
  7. 242068d A step towards changing StatsReport::Value::name to an enum. by tommi@webrtc.org · 11 years ago
  8. 03505bc Make StatsCollector depend on always having a valid session pointer. by tommi@webrtc.org · 11 years ago
  9. b5348c6 Minor refactoring of the session classes. by tommi@webrtc.org · 11 years ago
  10. d852434 (Auto)update libjingle 71107853-> 71115715 by buildbot@webrtc.org · 11 years ago
  11. b92f6f9 (Auto)update libjingle 71099685-> 71107853 by buildbot@webrtc.org · 11 years ago
  12. a4da771 Fix deadlock in Android stopCapture() call. by glaznev@webrtc.org · 11 years ago
  13. 5f43ce6 Fix a type cast issue for compiling webrtc with BoringSSL. by jiayl@webrtc.org · 11 years ago
  14. e04cb0e (Auto)update libjingle 70948025-> 70959275 by buildbot@webrtc.org · 11 years ago
  15. 9bef551 GN: Fix include paths for WebRTC in Chromium build. by kjellander@webrtc.org · 11 years ago
  16. 9e1acc8 Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 . by tommi@webrtc.org · 11 years ago
  17. dd6780d Remove always-true expression. by tommi@webrtc.org · 11 years ago
  18. eec6ecd Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 11 years ago
  19. 180e516 Thread annotate RTCPSender. by pbos@webrtc.org · 11 years ago
  20. 336e8e8 Fixing memcheck leak suppressions for XMPPClient tests. by pbos@webrtc.org · 11 years ago
  21. 168f23f Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 11 years ago
  22. ccbed3b Implement unittest SetRecvCodecsAcceptDefaultCodecs. by pbos@webrtc.org · 11 years ago
  23. a1bfcad Cast payload types to int for logging. by pbos@webrtc.org · 11 years ago
  24. fb2e7c2 Document that channels are stored contiguously in AudioBuffer by aluebs@webrtc.org · 11 years ago
  25. d212ffc Remove unnecessary build message. by tommi@webrtc.org · 11 years ago
  26. 4ef438e Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 11 years ago
  27. 0f42668 Roll chromium_revision 280876:282462 by henrikg@webrtc.org · 11 years ago
  28. cb97368 roll libyuv to r1033 for clang-cl support on windows. by fbarchard@google.com · 11 years ago
  29. b614d06 Rebase webrtc/base with r6655 version of talk/base: by henrike@webrtc.org · 11 years ago
  30. 72491b9 Count total bytes sent in RTPSender::Bytes(). by pbos@webrtc.org · 11 years ago
  31. 0422100 Fix data race in VCMTiming::ResetDecodeTime. by pbos@webrtc.org · 11 years ago
  32. bd9c092 Skip encoding in fake VP8 encoder. by pbos@webrtc.org · 11 years ago
  33. 7ae9108 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. by andresp@webrtc.org · 11 years ago
  34. 91f1752 Support VP8 encoder settings in VideoSendStream. by pbos@webrtc.org · 11 years ago
  35. 8f15121 Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 11 years ago
  36. 5bde66e audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h by bjornv@webrtc.org · 11 years ago
  37. 555fc78 Neon version of SubbandCoherence() by bjornv@webrtc.org · 11 years ago
  38. ac800c8 Neon version of rftbsub_128() by bjornv@webrtc.org · 11 years ago
  39. 5ac876b Revert "Remove remains of WEBRTC_NO_STL." (rev 6641). by andresp@webrtc.org · 11 years ago
  40. e91ba26 Revert 6643 "Revert 6637 "Revert 6636 "Roll chromium_revision 28..." by henrikg@webrtc.org · 11 years ago
  41. 02dce51 Revert 6637 "Revert 6636 "Roll chromium_revision 280876:281479"" by henrikg@webrtc.org · 11 years ago
  42. 7267020 (Auto)update libjingle 70813271-> 70818369 by buildbot@webrtc.org · 11 years ago
  43. 47d1c98 Remove remains of WEBRTC_NO_STL. by andresp@webrtc.org · 11 years ago
  44. 10ef8fe Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault. by jiayl@webrtc.org · 11 years ago
  45. 4b1f330 Fix a bug in SocketAddress where "a.b.c.d:1" and "b.b.c.d:1" are incorrectly considered equal. by jiayl@webrtc.org · 11 years ago
  46. 7af12be Thread annotations for vie_encoder.cc/.h by stefan@webrtc.org · 11 years ago
  47. e7771d0 Revert 6636 "Roll chromium_revision 280876:281479" by henrikg@webrtc.org · 11 years ago
  48. 543da99 Roll chromium_revision 280876:281479 by henrikg@webrtc.org · 11 years ago
  49. 045a9b1 Remove unnecessary race suppressions copied from chromium. by andresp@webrtc.org · 11 years ago
  50. b8e9e44 Add full stack test cases with a fake network pipe. by stefan@webrtc.org · 11 years ago
  51. e9cefde Improve libjingle's ASSERT and VERIFY macros on Windows. by tommi@webrtc.org · 11 years ago
  52. 01bda20 Fixed the stats problem when new track is using the same ssrc as the previous track. by xians@webrtc.org · 11 years ago
  53. b753762 delay_estimator: Increases test coverage and makes input spectrum const by bjornv@webrtc.org · 11 years ago
  54. 12b4efe Implement a work around for Chrome full-screen tab switch on Mac. by jiayl@webrtc.org · 11 years ago
  55. e55641d Neon version of rftfsub_128() by bjornv@webrtc.org · 11 years ago
  56. 55535d4 (Auto)update libjingle 70711261-> 70733822 by buildbot@webrtc.org · 11 years ago
  57. d11bec4 Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 11 years ago
  58. 3d7da88 Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable. by stefan@webrtc.org · 11 years ago
  59. ecb8723 Change Timing::WallTimeNow to be static. by tommi@webrtc.org · 11 years ago
  60. 62bafae Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 11 years ago
  61. 241a9b0 Fixing compile error. by phoglund@webrtc.org · 11 years ago
  62. 22292df Adding explicit check for using dummy file devices. by phoglund@webrtc.org · 11 years ago
  63. 33d110d Tight data race suppressions around thread_posix. by andresp@webrtc.org · 11 years ago
  64. af38f4e Extract RTP-header SSRC inline in Call. by pbos@webrtc.org · 11 years ago
  65. a70be68 Disabling shared socket mode for TURN ports. This is done as currently when by mallinath@webrtc.org · 11 years ago
  66. 3c637cd Clean data races from system_wrappers_unittests. by andresp@webrtc.org · 11 years ago
  67. 285e9bc Fix potential deadlock in webrtc/system_wrappers/source/logging_unittest.cc. by andresp@webrtc.org · 11 years ago
  68. 5f2c81c webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition. by henrike@webrtc.org · 11 years ago
  69. ba93f9a drmemory flaky: EndToEndTest.RestartingSendStreamPreservesRtpState[WithRtx] suppressed on drMemory. by henrike@webrtc.org · 11 years ago
  70. 161f808 Add test for VideoEncoder setup/teardown. by pbos@webrtc.org · 11 years ago
  71. 2bb1bda Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 11 years ago
  72. 73823ca Add initial gn build files for video_coding and video_processing. by stefan@webrtc.org · 11 years ago
  73. 03c817e Fix pacer to accept duplicate sequence numbers on different SSRCs. by pbos@webrtc.org · 11 years ago
  74. b941fe8 Fix data races related with traces in bitrate estimator test. by andresp@webrtc.org · 11 years ago
  75. bd249bc Remove GetDefaultConfigs() from Call. by pbos@webrtc.org · 11 years ago
  76. 7832648 Add missing break introduced in r6603. by stefan@webrtc.org · 11 years ago
  77. bee164a Fix test issues and a win compile error introduced with r6605. by stefan@webrtc.org · 11 years ago
  78. 875ad49 Revert conversion from TickTime to int64_t in paced sender. by stefan@webrtc.org · 11 years ago
  79. 8faa5db Add pbos@webrtc.org as owner for webrtc/test/. by pbos@webrtc.org · 11 years ago
  80. b9f5453 Add boilerplate code for H.264. by stefan@webrtc.org · 11 years ago
  81. d8440f7 Have Opus follow Chromium revisions by tina.legrand@webrtc.org · 11 years ago
  82. 20c1f56 Configure RTX send status on new modules. by pbos@webrtc.org · 11 years ago
  83. 88e0dda Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. by stefan@webrtc.org · 11 years ago
  84. 614000d Adding pbos as video/ owner and removing persons never working with this folder. by mflodman@webrtc.org · 11 years ago
  85. c5e53dd Revert 6597 "Roll chromium_revision 280876:281094" by kjellander@webrtc.org · 11 years ago
  86. cb1df98 Roll chromium_revision 280876:281094 by kjellander@webrtc.org · 11 years ago
  87. 720964f Fix memcheck error in r6594. by marpan@webrtc.org · 11 years ago
  88. 11bea89 GN: Implement BUILD.gn for common_video. by kjellander@webrtc.org · 11 years ago
  89. c836453 Fix for FEC decoding with sequence number wrap-around. by marpan@webrtc.org · 11 years ago
  90. 69ef991 delay_estimator: Allows dynamically used history sizes by bjornv@webrtc.org · 11 years ago
  91. 224a140 Make experimental NS API not purely virtual by aluebs@webrtc.org · 11 years ago
  92. c0ba439 common_audio: Removes macro WEBRTC_SPL_SHIFT_W16 by bjornv@webrtc.org · 11 years ago
  93. 38214d5 EchoCancellationImpl::ProcessRenderAudio: Use float samples directly by kwiberg@webrtc.org · 11 years ago
  94. a82f9a2 Add Tsan2 to .gitignore by andresp@webrtc.org · 11 years ago
  95. dfdaeb9 Removed old code and default implementations. by asapersson@webrtc.org · 11 years ago
  96. 9c89e93 WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. by braveyao@webrtc.org · 11 years ago
  97. 3ffa1f9 (Auto)update libjingle 70422491-> 70424781 by buildbot@webrtc.org · 11 years ago
  98. b25b08b Remove tools/resources by kjellander@webrtc.org · 11 years ago
  99. 93426cd Implement BUILD.gn for desktop_capture. by jiayl@webrtc.org · 11 years ago
  100. 33586c8 Make deadlock suppressions less generic. by andresp@webrtc.org · 11 years ago