Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
f7795df4261877af9eaf982cccfafc3fc52aeb4a
/
test
838c9da
Move gflags usage to video_loopback.
by pbos@webrtc.org
· 10 years ago
976ce98
Added include of assert.h for files calling assert but missing the include.
by henrike@webrtc.org
· 10 years ago
c54ff69
Add thread annotations to Call API.
by pbos@webrtc.org
· 10 years ago
b0079ed
Replace scoped_array<T> with scoped_ptr<T[]>.
by andrew@webrtc.org
· 10 years ago
b991cd0
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
by wu@webrtc.org
· 10 years ago
32e7755
Remove use of tmpnam.
by kjellander@webrtc.org
· 10 years ago
6b1114a
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
by fischman@webrtc.org
· 10 years ago
5f804f8
VoE changes to allow forwarding of packets from VoE to ViE BWE.
by solenberg@webrtc.org
· 10 years ago
f39df52
Remove internal codecs from VideoSendStream.
by pbos@webrtc.org
· 10 years ago
9420a1f
Implement minimum transmit bitrate.
by pbos@webrtc.org
· 10 years ago
af634a2
Remove platform-specific code from new-API tests.
by pbos@webrtc.org
· 10 years ago
10b8135
Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed.
by fischman@webrtc.org
· 10 years ago
a51b238
Add SetConfig method to FakeNetworkPipe and to DirectTransport
by henrik.lundin@webrtc.org
· 10 years ago
ee03b3b
Disable libjingle_peerconnection_java_unittest
by kjellander@webrtc.org
· 10 years ago
65bf249
Add RTCP packet class. Adds packet types: sr, rr, bye, fir.
by asapersson@webrtc.org
· 10 years ago
ae50521
Remove external encryption API for VoE.
by solenberg@webrtc.org
· 10 years ago
3f3e951
Incorrect overhead calculation when using FEC + RTP extension headers.
by sprang@webrtc.org
· 10 years ago
c766775
Wire up RTX in VideoReceiveStream.
by pbos@webrtc.org
· 11 years ago
9c8f391
Removes usage of ListWrapper from several files.
by henrike@webrtc.org
· 11 years ago
cd117d2
Integrate fake_network_pipe into direct_transport.
by stefan@webrtc.org
· 11 years ago
0d8474d
Remove metrics_unittests
by kjellander@webrtc.org
· 11 years ago
3a4fc4b
Update talk to 58127566 together with
by wu@webrtc.org
· 11 years ago
9b3d2bf
Revert 5274 "Update talk to 58113193 together with https://webrt..."
by wu@webrtc.org
· 11 years ago
cde78d6
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
by wu@webrtc.org
· 11 years ago
7123a80
Add SwapFrame() to VideoSendStreamInput.
by pbos@webrtc.org
· 11 years ago
894dab9
Roll chromium_revision 232627:238260
by kjellander@webrtc.org
· 11 years ago
090f37f
Improve VideoSendStreamTest::MaxPacketSize
by sprang@webrtc.org
· 11 years ago
e4d591a
Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..."
by andrew@webrtc.org
· 11 years ago
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
3d70641
Move implementation files out of the webrtc/ root.
by pbos@webrtc.org
· 11 years ago
da3ae7c
Adds support for sending redundant payloads over RTX.
by stefan@webrtc.org
· 11 years ago
309b2c8
Set local SSRC for VideoReceiveStream.
by pbos@webrtc.org
· 11 years ago
9105cbd
Set up SSRCs correctly after switching codec.
by pbos@webrtc.org
· 11 years ago
4a9843f
Implement and test EncodedImageCallback in new ViE API.
by sprang@webrtc.org
· 11 years ago
3c3a953
Add -Wnon-virtual-dtor warning for C++ code.
by pbos@webrtc.org
· 11 years ago
f3b4602
Rename newapi::Transport::SendRTP()->SendRtp().
by pbos@webrtc.org
· 11 years ago
8d2354a
Fix test broken with r5128.
by stefan@webrtc.org
· 11 years ago
26a736f
Hook up audio/video sync to Call.
by stefan@webrtc.org
· 11 years ago
f8c47a1
Improve Call tests for RTX.
by stefan@webrtc.org
· 11 years ago
e0df4d7
Remove update_resources.py as it's no longer used.
by kjellander@webrtc.org
· 11 years ago
685e91a
Update getUserMedia W3C conformance tests.
by kjellander@webrtc.org
· 11 years ago
b581c90
Separate Call API/build files from video_engine/.
by pbos@webrtc.org
· 11 years ago
2ba95be
Upgrade scoped_ptr to Chromium's latest version.
by andrew@webrtc.org
· 11 years ago
a19dab9
Revert "Disable tests for TSan v2"
by kjellander@webrtc.org
· 11 years ago
0b7aefe
Reorganize GYP targets to make webrtc.gyp more usable.
by kjellander@webrtc.org
· 11 years ago
9670be6
Add APK and isolate target for video_engine_tests
by kjellander@webrtc.org
· 11 years ago
3de1b22
Fix include of isolate.gypi
by kjellander@webrtc.org
· 11 years ago
109108e
Remove include_dirs from test.
by pbos@webrtc.org
· 11 years ago
88bcc98
Add libjingle_peerconnection_objc_test to buildbot_tests.py
by kjellander@webrtc.org
· 11 years ago
4bb3362
Disable tests for TSan v2
by kjellander@webrtc.org
· 11 years ago
29fce82
To use the channel_transport on the iOS platform, some #if directives are changed.
by sjlee@webrtc.org
· 11 years ago
e8eaed8
Call AllowCommandLineReparsing in unit tests.
by andrew@webrtc.org
· 11 years ago
1e88712
Make unittest log printouts opt-in with a --logs flag.
by andrew@webrtc.org
· 11 years ago
744235e
Restore severity precondition to logging.h.
by andrew@webrtc.org
· 11 years ago
5566bbd
Fix fileutils.cc for tests running under Win memory tools.
by kjellander@webrtc.org
· 11 years ago
d09d996
Fix metrics_unittests on Android.
by kjellander@webrtc.org
· 11 years ago
fd9b155
Add isolate configuration for Android for all tests.
by kjellander@webrtc.org
· 11 years ago
d09ee87
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 11 years ago
ac38916
Revert 4547 "Isolate GYP target and .isolate files for tests"
by kjellander@webrtc.org
· 11 years ago
12e3ee7
Isolate GYP target and .isolate files for tests
by kjellander@webrtc.org
· 11 years ago
80882f3
Replace MapWrapper with std::map<>.
by pbos@webrtc.org
· 11 years ago
4ab008f
Remove include_dirs from test/test.gyp.
by pbos@webrtc.org
· 11 years ago
50ff6a5
Switch C++-style C headers with their C equivalents.
by pbos@webrtc.org
· 11 years ago
9d939ee
Adds all unittests to android NDK-APK framework.
by henrike@webrtc.org
· 11 years ago
81e21c6
Added libjingle_peerconnection_java_unittest to buildbot_tests.py
by phoglund@webrtc.org
· 11 years ago
ae6d494
Fix some chromium-style warnings in webrtc/test/
by pbos@webrtc.org
· 11 years ago
0114e3d
Add root_path_android.cc to webrtc/test/Android.mk.
by pbos@webrtc.org
· 11 years ago
5ca7ffd
Arguments need to be separated when implementing gyp-actions.
by henrike@webrtc.org
· 11 years ago
6f44ab3
Disables unit tests that don't work on Android for Android.
by henrike@webrtc.org
· 11 years ago
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
ad63306
Revert 4298 "Makes it possible to find files used by some unit t..."
by pbos@webrtc.org
· 11 years ago
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
5ab7b93
Proper spacing for end-of-namespace comments.
by pbos@webrtc.org
· 11 years ago
1df6cc7
Reorganize test targets in WebRTC
by kjellander@webrtc.org
· 11 years ago
b3afc18
Remove #pragma once
by pbos@webrtc.org
· 11 years ago
6c9726a
Include files from webrtc/.. paths in test/channel_transport/
by pbos@webrtc.org
· 11 years ago
56041ab
Include files from webrtc/.. paths in test/
by pbos@webrtc.org
· 11 years ago
6483be5
Drop Virtual webcam check script as moved into buildbot scripts.
by kjellander@webrtc.org
· 11 years ago
5187bfa
Add script to ensure virtual webcam is running.
by kjellander@webrtc.org
· 11 years ago
caba49f
Add an option to override the TestToStderr trace printout time.
by andrew@webrtc.org
· 11 years ago
0a884c0
Revert "Updating test file contents to emmastjernloef"
by kjellander@webrtc.org
· 11 years ago
da3ad08
Updating test file contents to emmastjernloef
by kjellander@webrtc.org
· 11 years ago
b28e522
WebRTCDemo: Enable making multiple calls.
by fischman@webrtc.org
· 11 years ago
7793e44
Add OWNERS file for channel_transport
by kjellander@webrtc.org
· 11 years ago
b9ada57
WebRtc_Word32 -> int32_t in test/
by pbos@webrtc.org
· 11 years ago
34dac64
Fix no received audio in tests.
by pwestin@webrtc.org
· 11 years ago
5bea712
Two more sleep calls converted to use SleepMs().
by hta@webrtc.org
· 11 years ago
58a5924
Add some VoE and AudioProcessing mocks.
by andrew@webrtc.org
· 11 years ago
f658278
Refactor unittest trace printouts to a separate class.
by andrew@webrtc.org
· 11 years ago
9c3b7bd
Move the VIE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
680fbc5
Add trace printouts to all unit tests.
by andrew@webrtc.org
· 11 years ago
ce2d125
Creating a copy of Udp transport under webrtc/test
by pwestin@webrtc.org
· 11 years ago
9a7b9f7
Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004
by pwestin@webrtc.org
· 11 years ago
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
2733e12
Fixed a ton of Python lint errors, enabled python lint checking.
by phoglund@webrtc.org
· 11 years ago
ad3fd52
1. Updated test pages to include Chrome Frame meta tag
by elham@webrtc.org
· 11 years ago
83db9e9
Replace gtest_prod.h include with our own FRIEND_TEST macro.
by andrew@webrtc.org
· 11 years ago
9e605b2
Fix Windows x64 errors in video_codecs_test_framework
by kjellander@webrtc.org
· 11 years ago
cc895d1
Fixing/disabling Windows x64 warnings
by kjellander@webrtc.org
· 11 years ago
3824adf
Fix audio_e2e_test command line arguments
by kjellander@webrtc.org
· 11 years ago
Next »