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webrtc / src.git / 593c3a086872c18e1296a61ce112497f56cb0827
  1. 593c3a0 rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. by henrike@webrtc.org · 10 years ago
  2. 4530b2c Revert 7355 "Fix parallelization in libjingle_p2p_unittest." by henrike@webrtc.org · 10 years ago
  3. 36b0c1a Adds PRESUBMIT.py dispensation for depending on rtc_base. by henrike@webrtc.org · 10 years ago
  4. fd29205 Fix parallelization in libjingle_p2p_unittest. by pbos@webrtc.org · 10 years ago
  5. c86e45d Fix parallelizability in modules_tests. by pbos@webrtc.org · 10 years ago
  6. 4cebd84 Reland "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
  7. 4e4fe4f Add support for MSan by kjellander@webrtc.org · 10 years ago
  8. afefed5 Update checkdeps.py rules in DEPS by kjellander@webrtc.org · 10 years ago
  9. 83fe69d Added presubmit protecting against inclusion of rtc_base, while allowing rtc_base_approved. by henrike@webrtc.org · 10 years ago
  10. 3037bc3 GN: Add common configs to tools and test. by kjellander@webrtc.org · 10 years ago
  11. b8caf6a GN: Enable libvpx, add link target and convert some test targets by kjellander@webrtc.org · 10 years ago
  12. d05756f Changed mips_arch_variant variable value corresponding to Chromium code changes. by andrew@webrtc.org · 10 years ago
  13. 79a7148 Revert 7337 "Reland 28629004: adding new AEC dump start interfac..." by xians@webrtc.org · 10 years ago
  14. 7aad5e5 Revert 7338 "Fixed the android build by making the interface pur..." by xians@webrtc.org · 10 years ago
  15. d0bb586 Collecting stats every fixed time in webrtc_video_streaming.js test by houssainy@google.com · 10 years ago
  16. db75a66 Minor code change to fix some warnings in MIPS build. by andrew@webrtc.org · 10 years ago
  17. 90d1979 Fixed the android build by making the interface pure virtual. by xians@webrtc.org · 10 years ago
  18. 14092e0 Reland 28629004: adding new AEC dump start interface for chrome by xians@webrtc.org · 10 years ago
  19. 792d1a0 Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest. by henrike@webrtc.org · 10 years ago
  20. 8752061 Revert 7334 "adding new AEC dump start interface for chrome." by xians@webrtc.org · 10 years ago
  21. 2e417d6 adding new AEC dump start interface for chrome. by xians@webrtc.org · 10 years ago
  22. 38c121c Minor modifications to test::RtpFileReader by henrik.lundin@webrtc.org · 10 years ago
  23. 1795c358 Add default implementation of Add/RemoveObserver. by pbos@webrtc.org · 10 years ago
  24. 65e56db audio_processing/aecm: Added help function for calculating log of energy by bjornv@webrtc.org · 10 years ago
  25. 23ec837 audio_processing: Removed usage of macro WEBRTC_SPL_MUL by bjornv@webrtc.org · 10 years ago
  26. 750423c audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with << by bjornv@webrtc.org · 10 years ago
  27. 8cad943 Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" by kjellander@webrtc.org · 10 years ago
  28. 02cd306 Update isolate.gypi files + link to isolate_driver.py by kjellander@webrtc.org · 10 years ago
  29. 359d720 Allow Android apps to set video renderer scaling type. by glaznev@webrtc.org · 10 years ago
  30. 7dfb7fa Reland disallowing blocking calls on the worker thread. by jiayl@webrtc.org · 10 years ago
  31. ea6c12e Set thread scheduling parameters inside the new thread. by henrike@webrtc.org · 10 years ago
  32. 6266240 Disable flaky tests: by asapersson@webrtc.org · 10 years ago
  33. e794c36 Fix parallel test execution for tools, testsupport and metrics tests. by kjellander@webrtc.org · 10 years ago
  34. d711181 audio_processing: Replaced macro WEBRTC_SPL_LSHIFT_W16 with << by bjornv@webrtc.org · 10 years ago
  35. 7c15510 common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32 by bjornv@webrtc.org · 10 years ago
  36. 24f62e1 Adding getStats function to the exposed PeerConnection in RtcBot by houssainy@google.com · 10 years ago
  37. 730d270 Remove callback from RtpDepacketizer::Parse(). by pbos@webrtc.org · 10 years ago
  38. f21ea91 GN: Add common configs to all targets. by kjellander@webrtc.org · 11 years ago
  39. 34f2a9e Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 11 years ago
  40. 3a10d2f Roll chromium_revision deaf2f7e..c264a056 (295079:297113) by kjellander@webrtc.org · 11 years ago
  41. 6c6680a Cleanup .gclient.bot_entries to avoid sync problems on bots. by kjellander@webrtc.org · 11 years ago
  42. 3902054 Roll chromium_revision 6455c69..deaf2f7 (293954:295079) by kjellander@webrtc.org · 11 years ago
  43. bebc75e Fix the duplicated candidate problem when using multiple STUN servers. by jiayl@webrtc.org · 11 years ago
  44. 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
  45. 5d0071f Build one of NSS or BoringSSL but not both. by pthatcher@webrtc.org · 11 years ago
  46. a21d071 Reverting part of by thorcarpenter@google.com · 11 years ago
  47. 1fd362c Do not assert for blocking call allowed in Thread::Join. by jiayl@webrtc.org · 11 years ago
  48. 384d05f Remove the different block lengths in ns_core by aluebs@webrtc.org · 11 years ago
  49. 5088377 Revert 7297 "Remove the different block lengths in ns_core" by aluebs@webrtc.org · 11 years ago
  50. ca110b8 Mark virtual overrides of ViENetwork and VoENetwork as such. by henrikg@webrtc.org · 11 years ago
  51. 8b2e50c Revert 7302 "Roll chromium revision: 6455c69:2687a76" by marpan@webrtc.org · 11 years ago
  52. bfacaab Add accessors for array of channel pointers in AudioBuffer. They are by claguna@google.com · 11 years ago
  53. b38959e Roll chromium revision: 6455c69:2687a76 by marpan@webrtc.org · 11 years ago
  54. f1d751c Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. by jiayl@webrtc.org · 11 years ago
  55. 0530511 Explicitly initialize SSL for tests. by pbos@webrtc.org · 11 years ago
  56. 61e811f Bump to version 39 by tnakamura@webrtc.org · 11 years ago
  57. 60fbd65 Removing error triggered for disabling FEC on non-opus by minyue@webrtc.org · 11 years ago
  58. 5f39657 Remove the different block lengths in ns_core by aluebs@webrtc.org · 11 years ago
  59. 741711a Revert r7049/r7123, which added unnecessary "u"s to "return 0"s. by henrik.lundin@webrtc.org · 11 years ago
  60. 3156699 Fix typo from RtpPacketizerH264. by pbos@webrtc.org · 11 years ago
  61. 37e1846 Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293). by andresp@webrtc.org · 11 years ago
  62. fe1eafb Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. by jiayl@webrtc.org · 11 years ago
  63. 30be827 Enable render downmixing to mono in AudioProcessing. by andrew@webrtc.org · 11 years ago
  64. e1bba60 Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac by jiayl@webrtc.org · 11 years ago
  65. 3987b6d Fix a problem in Thread::Send. by jiayl@webrtc.org · 11 years ago
  66. a0ce9fa Call NS AnalyzeCaptureAudio before AEC by aluebs@webrtc.org · 11 years ago
  67. 70e2d11 Reduce jitter delay for low fps streams. Enabled by finch flag. by sprang@webrtc.org · 11 years ago
  68. 275dac2 Moved the filter calculation from analyze to process in ns_core by aluebs@webrtc.org · 11 years ago
  69. 634c926 audioproc: Now also writes to output file in simulation mode by bjornv@webrtc.org · 11 years ago
  70. 7ee24a7 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 11 years ago
  71. d60d79a Thread annotation of rtc::CriticalSection. by pbos@webrtc.org · 11 years ago
  72. 38344ed Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 11 years ago
  73. 8166fae Change Android video renderer to maintain video aspect by glaznev@webrtc.org · 11 years ago
  74. 90668b1 Switch HW video decoder to output byte buffers if video by glaznev@webrtc.org · 11 years ago
  75. 1b7dcc1 (Auto)update libjingle 76169599-> 76176062 by buildbot@webrtc.org · 11 years ago
  76. 94ff92c Use VPX_IMG_FMT_*/VPX_PLANE_* defines by johannkoenig@google.com · 11 years ago
  77. 2c1bcea Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 11 years ago
  78. 3987f10 Revert "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 11 years ago
  79. bf7b9e0 Remove DTMF status methods from Voice Engine by henrik.lundin@webrtc.org · 11 years ago
  80. e34a2e7 Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175) by kjellander@webrtc.org · 11 years ago
  81. faf2410 gn: Hide modules/video_capture:video_capture_internal_impl behind an arg by pbos@webrtc.org · 11 years ago
  82. 0e6e4d2 Reland "Converting five tests to use new AudioCoding interface" (r7258) by henrik.lundin@webrtc.org · 11 years ago
  83. 4f6f22f Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface" by andresp@webrtc.org · 11 years ago
  84. ea29787 audio_processing/agc: Solved building with AGC_DEBUG + few style changes by bjornv@webrtc.org · 11 years ago
  85. 0a2087a Skeleton for registering external encoders/decoders. by pbos@webrtc.org · 11 years ago
  86. c569a49 Unit tests for SSLAdapter by tkchin@webrtc.org · 11 years ago
  87. dc0b37d modules_unittests: Turned on ApmTest.Process test for Android by bjornv@webrtc.org · 11 years ago
  88. a3c4d4d Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." by andrew@webrtc.org · 11 years ago
  89. 8c5740b WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t by kwiberg@webrtc.org · 11 years ago
  90. 83f95ba Remove engine-level SetOptions. by pbos@webrtc.org · 11 years ago
  91. 99e404c Revert "Converting five tests to use new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 11 years ago
  92. 35850ff Adding test file path as argument of the rtcBot run command's arguments. by houssainy@google.com · 11 years ago
  93. 64a2f10 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 11 years ago
  94. 07ca949 Adding webrtc_video_streaming test by houssainy@google.com · 11 years ago
  95. c570761 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). by andresp@webrtc.org · 11 years ago
  96. cfe0735 Convert AcmReceiverTest to new AudioCoding interface by henrik.lundin@webrtc.org · 11 years ago
  97. eb1de5c Converting five tests to use new AudioCoding interface by henrik.lundin@webrtc.org · 11 years ago
  98. bdfdc96 Clang-format ns_core by aluebs@webrtc.org · 11 years ago
  99. 759982d Set number of temporal layers for VideoSendStream. by pbos@webrtc.org · 11 years ago
  100. 6121715 Ensure that NetEq recovers after a large timestamp jump by henrik.lundin@webrtc.org · 11 years ago