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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 07dc4bec7f186dbb85d9b8abc7b8cc7df2c9684a / test
  1. 88b558f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  2. 8ade059 Removing W3C conformance tests after move to web-platform-tests. by phoglund@webrtc.org · 10 years ago
  3. eb67a6b Refactor Call-based tests. by pbos@webrtc.org · 10 years ago
  4. 7eec1dd Add RTCP packet types to packet builder: by asapersson@webrtc.org · 10 years ago
  5. 38a2d46 Updated W3C getusermedia tests to the latest version of the spec. by phoglund@webrtc.org · 10 years ago
  6. 847dfa5 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 10 years ago
  7. e82b71d Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 10 years ago
  8. 00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  9. f006e8d Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
  10. f03a4a6 Updated conformance tests and w3c-ified them. by phoglund@webrtc.org · 10 years ago
  11. bdfcddf Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
  12. 81f8df9 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
  13. 59a001f Adding back platform specific renderer to video loopback test. by mflodman@webrtc.org · 10 years ago
  14. 00d9c49 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 10 years ago
  15. 774b3d3 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  16. 0a9ed7c Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
  17. 28b7c07 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  18. b5b8648 Add NACK and RPSI packet types to RTCP packet builder. by asapersson@webrtc.org · 10 years ago
  19. 22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
  20. bd49ac2 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 · 10 years ago
  21. bc57e0f Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
  22. 0b8a1c4 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  23. 11de507 Move gflags usage to video_loopback. by pbos@webrtc.org · 10 years ago
  24. 151f6f2 Added include of assert.h for files calling assert but missing the include. by henrike@webrtc.org · 10 years ago
  25. c476e64 Add thread annotations to Call API. by pbos@webrtc.org · 10 years ago
  26. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  27. 093fc0b Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase. by wu@webrtc.org · 10 years ago
  28. 1ed7008 Remove use of tmpnam. by kjellander@webrtc.org · 10 years ago
  29. 98f8320 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  30. fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
  31. e2a7a77 Remove internal codecs from VideoSendStream. by pbos@webrtc.org · 10 years ago
  32. 3f83f9c Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  33. ee86b90 Remove platform-specific code from new-API tests. by pbos@webrtc.org · 10 years ago
  34. f9747a8 Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed. by fischman@webrtc.org · 10 years ago
  35. 9900e37 Add SetConfig method to FakeNetworkPipe and to DirectTransport by henrik.lundin@webrtc.org · 10 years ago
  36. b1a7102 Disable libjingle_peerconnection_java_unittest by kjellander@webrtc.org · 10 years ago
  37. 663ba07 Add RTCP packet class. Adds packet types: sr, rr, bye, fir. by asapersson@webrtc.org · 10 years ago
  38. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  39. 3e4cdec Incorrect overhead calculation when using FEC + RTP extension headers. by sprang@webrtc.org · 10 years ago
  40. c71929d Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 10 years ago
  41. 083049f Removes usage of ListWrapper from several files. by henrike@webrtc.org · 10 years ago
  42. aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  43. a07c56f Remove metrics_unittests by kjellander@webrtc.org · 11 years ago
  44. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  45. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  46. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  47. c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  48. ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  49. b589c65 Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
  50. e388f9e Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
  51. 532b8f7 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
  52. d05597a Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
  53. 47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  54. 4b50db1 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  55. 66f4394 Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
  56. 2e98d45 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  57. e1e050e Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
  58. 3009c81 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
  59. 09f84e5 Fix test broken with r5128. by stefan@webrtc.org · 11 years ago
  60. e028410 Hook up audio/video sync to Call. by stefan@webrtc.org · 11 years ago
  61. 4985c7b Improve Call tests for RTX. by stefan@webrtc.org · 11 years ago
  62. b4db9c3 Remove update_resources.py as it's no longer used. by kjellander@webrtc.org · 11 years ago
  63. 626d764 Update getUserMedia W3C conformance tests. by kjellander@webrtc.org · 11 years ago
  64. 24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
  65. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  66. 9398252 Revert "Disable tests for TSan v2" by kjellander@webrtc.org · 11 years ago
  67. 8e70108 Reorganize GYP targets to make webrtc.gyp more usable. by kjellander@webrtc.org · 11 years ago
  68. bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  69. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  70. b655adf Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
  71. 3bd659f Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
  72. 032f731 Disable tests for TSan v2 by kjellander@webrtc.org · 11 years ago
  73. d8a5b00 To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 11 years ago
  74. b0fb1d6 Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 11 years ago
  75. e41c6b2 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 11 years ago
  76. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  77. 27103e5 Fix fileutils.cc for tests running under Win memory tools. by kjellander@webrtc.org · 11 years ago
  78. e879919 Fix metrics_unittests on Android. by kjellander@webrtc.org · 11 years ago
  79. 8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
  80. 3540c82 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  81. e155918 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
  82. 298bbdb Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  83. 52c5c70 Replace MapWrapper with std::map<>. by pbos@webrtc.org · 11 years ago
  84. 705b38d Remove include_dirs from test/test.gyp. by pbos@webrtc.org · 11 years ago
  85. 3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  86. acb00f5 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  87. 87ae00a Added libjingle_peerconnection_java_unittest to buildbot_tests.py by phoglund@webrtc.org · 11 years ago
  88. 77c6d71 Fix some chromium-style warnings in webrtc/test/ by pbos@webrtc.org · 11 years ago
  89. f9c7018 Add root_path_android.cc to webrtc/test/Android.mk. by pbos@webrtc.org · 11 years ago
  90. 609e332 Arguments need to be separated when implementing gyp-actions. by henrike@webrtc.org · 11 years ago
  91. 7537dde Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
  92. e25e28f 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
  93. 00d566e Revert 4298 "Makes it possible to find files used by some unit t..." by pbos@webrtc.org · 11 years ago
  94. 222efdc 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
  95. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  96. 46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  97. 50a4d9f Remove #pragma once by pbos@webrtc.org · 11 years ago
  98. 96001c8 Include files from webrtc/.. paths in test/channel_transport/ by pbos@webrtc.org · 11 years ago
  99. 20a5c46 Include files from webrtc/.. paths in test/ by pbos@webrtc.org · 11 years ago
  100. 41b55fa Drop Virtual webcam check script as moved into buildbot scripts. by kjellander@webrtc.org · 11 years ago