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webrtc / src / f2e4a99a39a4f6018d54d1e797da36dc520ac8d2 / webrtc / test
  1. 88fbb2d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 11 years ago
  2. 2fa7f79 Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 11 years ago
  3. 125ffd7 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 11 years ago
  4. a826006 Add NACK and RPSI packet types to RTCP packet builder. by asapersson@webrtc.org · 11 years ago
  5. cb711f77 Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 11 years ago
  6. 60015d2 Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} by andresp@webrtc.org · 11 years ago
  7. caba2d2 Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 11 years ago
  8. a36ad69 Add webrtc field trials API. by andresp@webrtc.org · 11 years ago
  9. 023b101 Move gflags usage to video_loopback. by pbos@webrtc.org · 11 years ago
  10. f2aafe4 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 11 years ago
  11. de1429e Add thread annotations to Call API. by pbos@webrtc.org · 11 years ago
  12. 8f69330 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 11 years ago
  13. cd70119 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 11 years ago
  14. 7de47bc Remove use of tmpnam. by kjellander@webrtc.org · 11 years ago
  15. 2c89b5c Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 11 years ago
  16. b1f5010 VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 11 years ago
  17. f577ae9 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 11 years ago
  18. 3349ae0 Implement minimum transmit bitrate. by pbos@webrtc.org · 11 years ago
  19. 95153cc Remove platform-specific code from new-API tests. by pbos@webrtc.org · 11 years ago
  20. 2bd5944 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 11 years ago
  21. c0e9aeb Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 11 years ago
  22. 55fcd71 Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 11 years ago
  23. 0f2809a Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 11 years ago
  24. a079233 Remove external encryption API for VoE. by solenberg@webrtc.org · 11 years ago
  25. 346094c Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 11 years ago
  26. c279a5d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  27. 79cf3ac Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  28. faada6e Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  29. 5fe2d65 Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
  30. a989080 Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  31. 2018269 Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  32. a129b6c Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  33. 724947b Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  34. f9bdbe3 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  35. 8b88192 Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
  36. 797522f Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
  37. 7104fc1 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
  38. c49d5b7 Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
  39. 7e9315b Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  40. b613b5a Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  41. 13d38a1 Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
  42. 4070935 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  43. b3cc78d Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
  44. 27326b6 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
  45. 28bf50f Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
  46. b082ade Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
  47. 69969e2 Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
  48. 4c828e1 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
  49. c4225b6 Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
  50. 16e03b7 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
  51. 31628aa Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  52. e7009f3 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
  53. 5b3b6b1 Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  54. 3f9288f Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  55. 2a97317 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  56. f8b2966 Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  57. ebd3ab0 Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
  58. d8a9b86 Disable tests for TSan v2 by kjellander@webrtc.org · 11 years ago
  59. c3e51ac To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 11 years ago
  60. 15e979b Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 11 years ago
  61. aa3d1c8 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 12 years ago
  62. 9080518 Restore severity precondition to logging.h. by andrew@webrtc.org · 12 years ago
  63. de49966 Fix fileutils.cc for tests running under Win memory tools. by kjellander@webrtc.org · 12 years ago
  64. f746f4f Fix metrics_unittests on Android. by kjellander@webrtc.org · 12 years ago
  65. e141373 Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 12 years ago
  66. 3365422 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 12 years ago
  67. 4298f73 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 12 years ago
  68. d7a4d23 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 12 years ago
  69. 4ca7d3f Replace MapWrapper with std::map<>. by pbos@webrtc.org · 12 years ago
  70. 2ab209e Remove include_dirs from test/test.gyp. by pbos@webrtc.org · 12 years ago
  71. 12dc1a3 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 12 years ago
  72. 89c6740 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 12 years ago
  73. a96d877 Added libjingle_peerconnection_java_unittest to buildbot_tests.py by phoglund@webrtc.org · 12 years ago
  74. e6c3966 Fix some chromium-style warnings in webrtc/test/ by pbos@webrtc.org · 12 years ago
  75. fc496d9 Add root_path_android.cc to webrtc/test/Android.mk. by pbos@webrtc.org · 12 years ago
  76. 5fc4d34 Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 12 years ago
  77. a950300b Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 12 years ago
  78. 34773d9 Unreverts revert: Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 12 years ago
  79. db7d82f Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 12 years ago
  80. caf2fcc Makes it possible to find files used by some unit tests when running them as Chrome native tests. by henrike@webrtc.org · 12 years ago
  81. d900e8b Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 12 years ago
  82. 6c35e0b Reorganize test targets in WebRTC by kjellander@webrtc.org · 12 years ago
  83. 9aca5b3 Remove #pragma once by pbos@webrtc.org · 12 years ago
  84. 1e50231 Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 12 years ago
  85. 34741c8 Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 12 years ago
  86. 219762a Drop Virtual webcam check script as moved into buildbot scripts. by kjellander@webrtc.org · 12 years ago
  87. 73a4d5a Add script to ensure virtual webcam is running. by kjellander@webrtc.org · 12 years ago
  88. 59aaebc Add an option to override the TestToStderr trace printout time. by andrew@webrtc.org · 12 years ago
  89. f090167 Revert "Updating test file contents to emmastjernloef" by kjellander@webrtc.org · 12 years ago
  90. 11959d3 Updating test file contents to emmastjernloef by kjellander@webrtc.org · 12 years ago
  91. 6f41ca9 WebRTCDemo: Enable making multiple calls. by fischman@webrtc.org · 12 years ago
  92. 59d8889 Add OWNERS file for channel_transport by kjellander@webrtc.org · 12 years ago
  93. a5f1787 WebRtc_Word32 -> int32_t in test/ by pbos@webrtc.org · 12 years ago
  94. 835dbf4 Fix no received audio in tests. by pwestin@webrtc.org · 12 years ago
  95. fcb7c38 Two more sleep calls converted to use SleepMs(). by hta@webrtc.org · 12 years ago
  96. c83a00a Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 12 years ago
  97. b87cc85 Refactor unittest trace printouts to a separate class. by andrew@webrtc.org · 12 years ago
  98. 26e35e1 Move the VIE tests to use external transport instead of the built in udp transport by pwestin@webrtc.org · 12 years ago
  99. c1ffd33 Add trace printouts to all unit tests. by andrew@webrtc.org · 12 years ago
  100. 999e900 Creating a copy of Udp transport under webrtc/test by pwestin@webrtc.org · 12 years ago