Gitiles
Code Review Sign In
gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b4bc1a60992b0811edcf1aae780dcd1e79b7e948 / test
  1. 1bc4fa6 Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
  2. 11fa357 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  3. 838c9da Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
  4. 976ce98 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 10 years ago
  5. c54ff69 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  6. b0079ed Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  7. b991cd0 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  8. 32e7755 Remove use of tmpnam. by kjellander@webrtc.org · 10 years ago
  9. 6b1114a Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  10. 5f804f8 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  11. f39df52 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
  12. 9420a1f Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  13. af634a2 Remove platform-specific code from new-API tests. by pbos@webrtc.org · 10 years ago
  14. 10b8135 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 10 years ago
  15. a51b238 Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 10 years ago
  16. ee03b3b Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 10 years ago
  17. 65bf249 Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 10 years ago
  18. ae50521 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  19. 3f3e951 Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 10 years ago
  20. c766775 Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  21. 9c8f391 Removes usage of ListWrapper from several files. by henrike@webrtc.org · 10 years ago
  22. cd117d2 Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  23. 0d8474d Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
  24. 3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  25. 9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  26. cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  27. 7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  28. 894dab9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  29. 090f37f Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
  30. e4d591a Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
  31. 6dccf13 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered. by sprang@webrtc.org · 11 years ago
  32. 3d70641 Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
  33. da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  34. 309b2c8 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  35. 9105cbd Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
  36. 4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  37. 3c3a953 Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
  38. f3b4602 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
  39. 8d2354a Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
  40. 26a736f Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
  41. f8c47a1 Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
  42. e0df4d7 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
  43. 685e91a Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
  44. b581c90 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
  45. 2ba95be Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  46. a19dab9 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
  47. 0b7aefe Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  48. 9670be6 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  49. 3de1b22 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  50. 109108e Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  51. 88bcc98 Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
  52. 4bb3362 Disable tests for TSan v2 by kjellander@webrtc.org · 11 years ago
  53. 29fce82 To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 11 years ago
  54. e8eaed8 Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 11 years ago
  55. 1e88712 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 11 years ago
  56. 744235e Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  57. 5566bbd Fix fileutils.cc for tests running under Win memory tools. by kjellander@webrtc.org · 11 years ago
  58. d09d996 Fix metrics_unittests on Android. by kjellander@webrtc.org · 11 years ago
  59. fd9b155 Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
  60. d09ee87 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  61. ac38916 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
  62. 12e3ee7 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  63. 80882f3 Replace MapWrapper with std::map<>. by pbos@webrtc.org · 11 years ago
  64. 4ab008f Remove include_dirs from test/test.gyp. by pbos@webrtc.org · 11 years ago
  65. 50ff6a5 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  66. 9d939ee Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  67. 81e21c6 Added libjingle_peerconnection_java_unittest to buildbot_tests.py by phoglund@webrtc.org · 11 years ago
  68. ae6d494 Fix some chromium-style warnings in webrtc/test/ by pbos@webrtc.org · 11 years ago
  69. 0114e3d Add root_path_android.cc to webrtc/test/Android.mk. by pbos@webrtc.org · 11 years ago
  70. 5ca7ffd Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 11 years ago
  71. 6f44ab3 Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
  72. 0642536 Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
  73. ad63306 Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
  74. c82b35c Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 11 years ago
  75. 5ab7b93 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  76. 1df6cc7 Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  77. b3afc18 Remove #pragma once by pbos@webrtc.org · 11 years ago
  78. 6c9726a Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
  79. 56041ab Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
  80. 6483be5 Drop Virtual webcam check script as moved into buildbot scripts. by kjellander@webrtc.org · 11 years ago
  81. 5187bfa Add script to ensure virtual webcam is running. by kjellander@webrtc.org · 11 years ago
  82. caba49f Add an option to override the TestToStderr trace printout time. by andrew@webrtc.org · 11 years ago
  83. 0a884c0 Revert "Updating test file contents to emmastjernloef" by kjellander@webrtc.org · 11 years ago
  84. da3ad08 Updating test file contents to emmastjernloef by kjellander@webrtc.org · 11 years ago
  85. b28e522 WebRTCDemo: Enable making multiple calls. by fischman@webrtc.org · 11 years ago
  86. 7793e44 Add OWNERS file for channel_transport by kjellander@webrtc.org · 11 years ago
  87. b9ada57 WebRtc_Word32 -> int32_t in test/ by pbos@webrtc.org · 11 years ago
  88. 34dac64 Fix no received audio in tests. by pwestin@webrtc.org · 11 years ago
  89. 5bea712 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 11 years ago
  90. 58a5924 Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
  91. f658278 Refactor unittest trace printouts to a separate class. by andrew@webrtc.org · 11 years ago
  92. 9c3b7bd Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 11 years ago
  93. 680fbc5 Add trace printouts to all unit tests. by andrew@webrtc.org · 11 years ago
  94. ce2d125 Creating a copy of Udp transport under webrtc/test by pwestin@webrtc.org · 11 years ago
  95. 9a7b9f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  96. 66ccc6e Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
  97. 2733e12 Fixed a ton of Python lint errors, enabled python lint checking. by phoglund@webrtc.org · 11 years ago
  98. ad3fd52 1. Updated test pages to include Chrome Frame meta tag by elham@webrtc.org · 11 years ago
  99. 83db9e9 Replace gtest_prod.h include with our own FRIEND_TEST macro. by andrew@webrtc.org · 11 years ago
  100. 9e605b2 Fix Windows x64 errors in video_codecs_test_framework by kjellander@webrtc.org · 11 years ago