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gerrit-public.fairphone.software / platform / external / webrtc / 4165f7aa226584f6865df5ce37b0cb0bae4150d8
  1. 4165f7a Add a variable for deciding when to use openmax_dl. by andrew@webrtc.org · 11 years ago
  2. f71785c audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> by bjornv@webrtc.org · 11 years ago
  3. 575d126 Protect send_/recv_streams_ in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  4. 9c6dc46 CHECK/DCHECK: Explicitly state whether the condition can have side effects by kwiberg@webrtc.org · 11 years ago
  5. 5e3d7c7 Change name of a NetEq internal member variable by henrik.lundin@webrtc.org · 11 years ago
  6. 742922b Make the media content send only if offerToReceive is false while local streams exist. by jiayl@webrtc.org · 11 years ago
  7. d6bda09 Initialize sctp_paddrparams in OpenSctpSocket(). by pbos@webrtc.org · 11 years ago
  8. 27e5898 Explicitly unpoison FDs for MSan. by pbos@webrtc.org · 11 years ago
  9. 46ffc70 Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder. by glaznev@webrtc.org · 11 years ago
  10. 963b979 Remove potential deadlock in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  11. a9e363e Roll chromium_revision c264a05..fc668e2 (297113:298195) by kjellander@webrtc.org · 11 years ago
  12. 77d5a57 Revert "Only configure the SSL library in one place." by pbos@webrtc.org · 11 years ago
  13. 6ed1cf4 Isolate: Remove use of --ignore_broken_items by kjellander@webrtc.org · 11 years ago
  14. 9103953 Fix neteq_rtpplay so that empty SSRC is valid by henrik.lundin@webrtc.org · 11 years ago
  15. 7cbc4f9 Set NetEq playout mode through the Config struct by henrik.lundin@webrtc.org · 11 years ago
  16. 8b65d51 Add an SSRC filter to neteq_rtpplay by henrik.lundin@webrtc.org · 11 years ago
  17. 532ed43 Prevent reading outside iSAC bitstream, if the stream is corrupted. by turaj@webrtc.org · 11 years ago
  18. 8234fa6 Only configure the SSL library in one place. by henrike@webrtc.org · 11 years ago
  19. 2fe5893 Mac: adds missing _DEBUG flag to mac debug builds. by henrike@webrtc.org · 11 years ago
  20. 528fc65 Fixing build issue with L-sdk by henrike@webrtc.org · 11 years ago
  21. 9a742b4 talk: removes empty directories base and sound. by henrike@webrtc.org · 11 years ago
  22. 5d3e7ac Check on the existence of report directory by houssainy@google.com · 11 years ago
  23. 42684be Wire up CPU adaptation in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  24. 31b75ea Moves xmllite's unittests to rtc_unittest. by henrike@webrtc.org · 11 years ago
  25. 25cc745 Switch to SW video decoder on Android after getting 2 or more by glaznev@webrtc.org · 11 years ago
  26. 4b133da Let RtpFileSource use RtpFileReader by henrik.lundin@webrtc.org · 11 years ago
  27. 348eac6 audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >> by bjornv@webrtc.org · 11 years ago
  28. 5fa8c45 Remove mouse cursor capturer from the ScreenCapturer interface by sergeyu@chromium.org · 11 years ago
  29. 6138f0f Revert "Remove mouse cursor capturer from the ScreenCapturer interface" by sergeyu@chromium.org · 11 years ago
  30. 1fced0f Remove mouse cursor capturer from the ScreenCapturer interface by sergeyu@chromium.org · 11 years ago
  31. 76819d3 Add error trap for XFixesGetCursorImage() by sergeyu@chromium.org · 11 years ago
  32. 325cff0 Import LappedTransform and friends. by andrew@webrtc.org · 11 years ago
  33. 593c3a0 rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. by henrike@webrtc.org · 11 years ago
  34. 4530b2c Revert 7355 "Fix parallelization in libjingle_p2p_unittest." by henrike@webrtc.org · 11 years ago
  35. 36b0c1a Adds PRESUBMIT.py dispensation for depending on rtc_base. by henrike@webrtc.org · 11 years ago
  36. fd29205 Fix parallelization in libjingle_p2p_unittest. by pbos@webrtc.org · 11 years ago
  37. c86e45d Fix parallelizability in modules_tests. by pbos@webrtc.org · 11 years ago
  38. 4cebd84 Reland "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 11 years ago
  39. 4e4fe4f Add support for MSan by kjellander@webrtc.org · 11 years ago
  40. afefed5 Update checkdeps.py rules in DEPS by kjellander@webrtc.org · 11 years ago
  41. 83fe69d Added presubmit protecting against inclusion of rtc_base, while allowing rtc_base_approved. by henrike@webrtc.org · 11 years ago
  42. 3037bc3 GN: Add common configs to tools and test. by kjellander@webrtc.org · 11 years ago
  43. b8caf6a GN: Enable libvpx, add link target and convert some test targets by kjellander@webrtc.org · 11 years ago
  44. d05756f Changed mips_arch_variant variable value corresponding to Chromium code changes. by andrew@webrtc.org · 11 years ago
  45. 79a7148 Revert 7337 "Reland 28629004: adding new AEC dump start interfac..." by xians@webrtc.org · 11 years ago
  46. 7aad5e5 Revert 7338 "Fixed the android build by making the interface pur..." by xians@webrtc.org · 11 years ago
  47. d0bb586 Collecting stats every fixed time in webrtc_video_streaming.js test by houssainy@google.com · 11 years ago
  48. db75a66 Minor code change to fix some warnings in MIPS build. by andrew@webrtc.org · 11 years ago
  49. 90d1979 Fixed the android build by making the interface pure virtual. by xians@webrtc.org · 11 years ago
  50. 14092e0 Reland 28629004: adding new AEC dump start interface for chrome by xians@webrtc.org · 11 years ago
  51. 792d1a0 Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest. by henrike@webrtc.org · 11 years ago
  52. 8752061 Revert 7334 "adding new AEC dump start interface for chrome." by xians@webrtc.org · 11 years ago
  53. 2e417d6 adding new AEC dump start interface for chrome. by xians@webrtc.org · 11 years ago
  54. 38c121c Minor modifications to test::RtpFileReader by henrik.lundin@webrtc.org · 11 years ago
  55. 1795c35 Add default implementation of Add/RemoveObserver. by pbos@webrtc.org · 11 years ago
  56. 65e56db audio_processing/aecm: Added help function for calculating log of energy by bjornv@webrtc.org · 11 years ago
  57. 23ec837 audio_processing: Removed usage of macro WEBRTC_SPL_MUL by bjornv@webrtc.org · 11 years ago
  58. 750423c audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with << by bjornv@webrtc.org · 11 years ago
  59. 8cad943 Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" by kjellander@webrtc.org · 11 years ago
  60. 02cd306 Update isolate.gypi files + link to isolate_driver.py by kjellander@webrtc.org · 11 years ago
  61. 359d720 Allow Android apps to set video renderer scaling type. by glaznev@webrtc.org · 11 years ago
  62. 7dfb7fa Reland disallowing blocking calls on the worker thread. by jiayl@webrtc.org · 11 years ago
  63. ea6c12e Set thread scheduling parameters inside the new thread. by henrike@webrtc.org · 11 years ago
  64. 6266240 Disable flaky tests: by asapersson@webrtc.org · 11 years ago
  65. e794c36 Fix parallel test execution for tools, testsupport and metrics tests. by kjellander@webrtc.org · 11 years ago
  66. d711181 audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with << by bjornv@webrtc.org · 11 years ago
  67. 7c15510 common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32 by bjornv@webrtc.org · 11 years ago
  68. 24f62e1 Adding getStats function to the exposed PeerConnection in RtcBot by houssainy@google.com · 11 years ago
  69. 730d270 Remove callback from RtpDepacketizer::Parse(). by pbos@webrtc.org · 11 years ago
  70. f21ea91 GN: Add common configs to all targets. by kjellander@webrtc.org · 11 years ago
  71. 34f2a9e Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 11 years ago
  72. 3a10d2f Roll chromium_revision deaf2f7e..c264a056 (295079:297113) by kjellander@webrtc.org · 11 years ago
  73. 6c6680a Cleanup .gclient.bot_entries to avoid sync problems on bots. by kjellander@webrtc.org · 11 years ago
  74. 3902054 Roll chromium_revision 6455c69..deaf2f7 (293954:295079) by kjellander@webrtc.org · 11 years ago
  75. bebc75e Fix the duplicated candidate problem when using multiple STUN servers. by jiayl@webrtc.org · 11 years ago
  76. 0a256ac Getting orientation is not working properly. VideoCaptureImpl::RotationFromDegrees returns -1 in case fails not 0. So we need to change the if statement. by braveyao@webrtc.org · 11 years ago
  77. 5d0071f Build one of NSS or BoringSSL but not both. by pthatcher@webrtc.org · 11 years ago
  78. a21d071 Reverting part of by thorcarpenter@google.com · 11 years ago
  79. 1fd362c Do not assert for blocking call allowed in Thread::Join. by jiayl@webrtc.org · 11 years ago
  80. 384d05f Remove the different block lengths in ns_core by aluebs@webrtc.org · 11 years ago
  81. 5088377 Revert 7297 "Remove the different block lengths in ns_core" by aluebs@webrtc.org · 11 years ago
  82. ca110b8 Mark virtual overrides of ViENetwork and VoENetwork as such. by henrikg@webrtc.org · 11 years ago
  83. 8b2e50c Revert 7302 "Roll chromium revision: 6455c69:2687a76" by marpan@webrtc.org · 11 years ago
  84. bfacaab Add accessors for array of channel pointers in AudioBuffer. They are by claguna@google.com · 11 years ago
  85. b38959e Roll chromium revision: 6455c69:2687a76 by marpan@webrtc.org · 11 years ago
  86. f1d751c Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. by jiayl@webrtc.org · 11 years ago
  87. 0530511 Explicitly initialize SSL for tests. by pbos@webrtc.org · 11 years ago
  88. 61e811f Bump to version 39 by tnakamura@webrtc.org · 11 years ago
  89. 60fbd65 Removing error triggered for disabling FEC on non-opus by minyue@webrtc.org · 11 years ago
  90. 5f39657 Remove the different block lengths in ns_core by aluebs@webrtc.org · 11 years ago
  91. 741711a Revert r7049/r7123, which added unnecessary "u"s to "return 0"s. by henrik.lundin@webrtc.org · 11 years ago
  92. 3156699 Fix typo from RtpPacketizerH264. by pbos@webrtc.org · 11 years ago
  93. 37e1846 Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293). by andresp@webrtc.org · 11 years ago
  94. fe1eafb Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. by jiayl@webrtc.org · 11 years ago
  95. 30be827 Enable render downmixing to mono in AudioProcessing. by andrew@webrtc.org · 11 years ago
  96. e1bba60 Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac by jiayl@webrtc.org · 11 years ago
  97. 3987b6d Fix a problem in Thread::Send. by jiayl@webrtc.org · 11 years ago
  98. a0ce9fa Call NS AnalyzeCaptureAudio before AEC by aluebs@webrtc.org · 11 years ago
  99. 70e2d11 Reduce jitter delay for low fps streams. Enabled by finch flag. by sprang@webrtc.org · 11 years ago
  100. 275dac2 Moved the filter calculation from analyze to process in ns_core by aluebs@webrtc.org · 11 years ago