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2013-06-19Add dummy Android test APK to be used for buildbot automation testing.kjellander@webrtc.org
Until we have WebRTC test targets created for Android, this test makes it possible to move forward for buildbot automation. TEST=Android NDK buildbot and local execution of: source build/android/envsetup.sh gclient runhooks ninjar -C out/Debug verified the out/Debug/simple_apk dir exists and has the files. BUG=1882 R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1688005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4245 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-18Chromium Android tools for test execution.kjellander@webrtc.org
The md5sum and forwarder2 binaries from Chromium's src/tools/android are needed to be able to run tests using the test framework launched by build/android/run_tests.py. Since they depend on Chromium's base, we're using a precompiled copy for WebRTC's purposes. Linux works out of the box if Chromium's Android build instructions at https://code.google.com/p/chromium/wiki/AndroidBuildInstructions are used. Mac runs into problems earlier in the build toolchain, but as Mac is not a supported Android development platform in Chrome, the files will have to be copied manually on that platform for now. TEST=Synced, built and ran a test APK using run_tests.py. BUG=1882 R=andrew@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1679005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4241 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14Merge more tests into modules_{unit,integration}tests.kjellander@webrtc.org
A new test target named 'modules_integrationtests' is created and the following test targets were merged into it: * audio_coding_module_test * test_fec * video_coding_integrationtests * vp8_integrationtests A couple of other targets were merged into modules_unittests: * audio_coding_unittests * audioproc_unittest * common_unittests * video_coding_unittests * video_processing_unittests * vp8_unittests I wasn't able to merge audio_decoder_unittests and neteq_unittests due to conflicts with different defines in these tests. Some tests that have special requirements aren't merged into modules_integrationtests yet. I took the opportunity to rename them since the bot configs will need to be update anyway: * audio_device_test_api -> audio_device_integrationtests * video_capture_module_test -> video_capture_integrationtests * video_render_module_test -> video_render_integrationtests Exclude files were added for modules_integrationtests to make sure the memcheck and tsan bots doesn't tests that are too slow (audio_coding_module_test and vp8_integrationtests were previously disabled on those bots). Suppressions for AudioCodingModuleTest needed to be added to get modules_integrationtests to pass memcheck (even if the test is excluded from execution). BUG=1843 TEST=local execution on Linux and trybots (passing except the merged tests of course) R=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1656004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-11Reorganize test targets in WebRTCkjellander@webrtc.org
This CL will lower the number of test targets in WebRTC by: Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006): * resampler_unittests * signal_processing_unittests * vad_unittests Merge into modules_unittests: * bitrate_controller_unittests * desktop_capture_unittests * media_file_unittests * remote_bitrate_estimator_unittests * rtp_rtcp_unittests * paced_sender_unittests Merge into test_support_unittests: * channel_transport_unittests channel_transport.gyp was also removed in favor for test.gyp. I had to remove a main method from rtcp_format_remb_unittest.cc since it caused the fileutils.h code to not be able to find the right project root path in ordrer to provide correct paths to test files. Buildbot configuration update will be synced with the commit of this CL. TEST=trybots BUG=1843 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17Moved command line parsing to internal tools and moved back the mic volume ↵phoglund@webrtc.org
thingie. BUG= R=henrika@webrtc.org, kjellander@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1491004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4054 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16Moved force_volume_max to its own gyp file to avoid a circular dependency.phoglund@webrtc.org
BUG= TBR=tlegrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4046 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-13Added a config class to ease passing a set of options across webrtc.andresp@webrtc.org
Its main design reason is to expose control of experimental webrtc features. R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1450009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4004 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-22Add trace printouts to all unit tests.andrew@webrtc.org
Unfortunately, this requires splitting system_wrappers_unittests out of system_wrappers.gyp to avoid a cyclic dependency. TESTED=ran a few unit tests and observed printouts Review URL: https://webrtc-codereview.appspot.com/1221006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3711 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-21Creating a copy of Udp transport under webrtc/testpwestin@webrtc.org
Adding a test namespace, updating the include paths and renamed folder name. Review URL: https://webrtc-codereview.appspot.com/1203004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3701 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13Revert r3667 and r3665pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13Removed the engine API:s related to transport such as SetSendDestination, ↵pwestin@webrtc.org
the functionality is now provided via the test frame work. Review URL: https://webrtc-codereview.appspot.com/1029004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22Move src/ -> webrtc/andrew@webrtc.org
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-22Moving frame_analyzer and rgba_to_i420_converter to src/tools.vspasova@webrtc.org
It might be useful to have these under src/tools as this way they will automatically sync in Chrome. BUG= Review URL: https://webrtc-codereview.appspot.com/740004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2653 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-10Move include_tests to a higher variable nesting.andrew@webrtc.org
TBR=wu@webrtc.org BUG=none TEST=trybots Review URL: https://webrtc-codereview.appspot.com/734004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2587 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-09Make some dependencies more flexible.andrew@webrtc.org
BUG=none TEST=trybot Review URL: https://webrtc-codereview.appspot.com/728005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2583 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-08-09Don't target top-level test gyps if include_tests==0andrew@webrtc.org
Also add the possibility to pass a flag to the gyp run during gclient hooks. Review URL: https://webrtc-codereview.appspot.com/729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2582 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-27Reorganize the vp8 directory.andrew@webrtc.org
The usual changes: vp8/main/source -> vp8/ vp8/main/test -> vp8/test vp8/main/interface -> vp8/include All include paths etc. updated as needed. BUG=none TEST=trybots Review URL: https://webrtc-codereview.appspot.com/704004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2536 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-23Make some build settings more flexible.andrew@webrtc.org
BUG=issue676 TEST=trybots Review URL: https://webrtc-codereview.appspot.com/700006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2524 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-27Move test to src/test.andrew@webrtc.org
- Refer to top-level directories by <(DEPTH), e.g. <(DEPTH)/testing. - Remove now unneeded third_party_root. TBR=henrike@webrtc.org BUG=none TEST=trybots Review URL: https://webrtc-codereview.appspot.com/669007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2446 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-10Add a framework for audio end-to-end quality testing.andrew@webrtc.org
The quality comparison step is still to be done. BUG=issue502 TEST=manual Review URL: https://webrtc-codereview.appspot.com/577005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2220 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-09Removing PeerConnection sample client and libjingle from webrtc.wu@webrtc.org
The new PeerConnection and sample client can be found in libjingle. http://code.google.com/p/libjingle/ BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/389005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1658 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04Changing all PSNR/SSIM calculations to use libyuv.kjellander@webrtc.org
Removed old PSNR/SSIM implementations in: * test/testsupport/metrics/video_metrics.cc * src/modules/video_coding/codecs/test_framework/test.cc The functions in video_metrics.cc is now using code in libyuv instead. Old code in test.cc is using the same functions. The code for video_metrics.h had to be moved into a separate GYP file to avoid circular dependency error on Mac (see issue 160 for more details). The reason for this is that libyuv's unittest target depends on test_support_main. BUG= TEST=metrics_unittests in Debug+Release on Linux, Mac and Windows. Review URL: http://webrtc-codereview.appspot.com/333025 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1325 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09Workaround for libyuv libjingle breakage.andrew@webrtc.org
libjingle depends on ConvertFromI420. This was previously available through vplib. libjingle still has access to the vplib header, but the implementation is no longer built. Fortunately, the libyuv wrapper can supply the implementation, if we hack the signature to return to the unsigned int types. We'll remove this once libjingle has been updated to use libyuv directly. Also, roll libyuv to r100 which fixes a gyp warning on Windows. TEST=build Review URL: http://webrtc-codereview.appspot.com/323004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1151 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09The change http://webrtc-codereview.appspot.com/299001 (commit 1062) does ↵henrikg@webrtc.org
not do what it intends (exclude codecs from Chromium build). This is a fix for that. webrtc.gyp is not pulled in Chromium, hence it has no effect putting a define there. Moving it to src/build/common.gypi. Review URL: http://webrtc-codereview.appspot.com/315002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1143 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-09Give peerconnection its own gyp and disable.andrew@webrtc.org
r1140 broke the libjingle revision we're pulling. The fix in libjingle is pending; rather than reverting r1140, this temporarily disables peerconnection in the default build. TBR=tommi@webrtc.org TEST=build Review URL: http://webrtc-codereview.appspot.com/323002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1141 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-05Fixing crash when calculating SSIM and PSNR with empty video files in ↵kjellander@webrtc.org
video_metrics.cc There were previously a dependency on system_wrappers that is now removed (uses defines to check for SEE2 instructions during compilation time instead). Review URL: http://webrtc-codereview.appspot.com/296008 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1102 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30To be able to get webrtc into chrome, we need to reduce the size of the ↵xians@webrtc.org
binary and the usage of memory. This patch disbale some codecs which are not considered necessary. Review URL: http://webrtc-codereview.appspot.com/299001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1062 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-23Fixing Release compilation errorskjellander@webrtc.org
Review URL: http://webrtc-codereview.appspot.com/267026 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1000 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16Remove module targets due to Linux-Release errors.andrew@webrtc.org
TBR=phoglund Review URL: http://webrtc-codereview.appspot.com/277006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@958 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-16Included modules in webrtc.gyp and fixed build errors.phoglund@webrtc.org
Removed TODO from webrtc.gyp since it is done. Tabs -> spaces. Tabs -> spaces. Tabs -> spaces. Fixed compilation on Windows. Added missing file. Merge branch 'master' into fix_mac_modules Fixed compilation errors for the modules.gyp on Mac. This included some pretty large refactorings. Please enter the commit message for your changes. Lines starting BUG= TEST= Review URL: http://webrtc-codereview.appspot.com/269005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@957 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-04Enabling all common_video targets in webrtc.gyp.andrew@webrtc.org
Review URL: http://webrtc-codereview.appspot.com/268001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@888 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-31Fixing system_wrappers unittests.kjellander@webrtc.org
Not complete, but enough to include them in the build again. Review URL: http://webrtc-codereview.appspot.com/241008 git-svn-id: http://webrtc.googlecode.com/svn/trunk@842 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06Move merge_libs targets to their own gyp.andrew@webrtc.org
The main reason is to depend on all ("*") targets in voice_engine.gyp and video_engine.gyp. We don't want the merge_lib targets building by default, since they do funny stuff like delete some libraries. Review URL: http://webrtc-codereview.appspot.com/191003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@699 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-14Add a unit testing framework.andrew@webrtc.org
Populate it with the beginnings of a resampler unit test to have it do someting. Also fix a bug in resampler caught with the test ;) Review URL: http://webrtc-codereview.appspot.com/135019 git-svn-id: http://webrtc.googlecode.com/svn/trunk@595 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-09-12refactor the gyp file to gypi file.xians@google.com
Basically, the gypi file is a copy of gyp file, but has some difference on the path of the dependencies. Review URL: http://webrtc-codereview.appspot.com/137020 git-svn-id: http://webrtc.googlecode.com/svn/trunk@581 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-25First version of the peerconnection client application for Linux.tommi@webrtc.org
I made several updates to the Windows version as well so that both implementations share a big portion of the code. The underlying PeerConnection notifications have changed a bit since the last update so that there's still a known issue that I plan to fix in my next change: // TODO(tommi): There's a problem now with terminating connections: // When ending a conversation, both peers now send a signaling message // that indicates that their ports are closed (port=0). The trouble this // causes us here is that we can interpret such a message as an invite // to a new conversation. So, currently there is a bug that ending // a conversation can immediately start a new one. // To fix this I plan to change how conversations start and have a special // notification message via the server that prepares a client for a // conversation instead of automatically recognizing the first signaling // message as an invite. Review URL: http://webrtc-codereview.appspot.com/112008 git-svn-id: http://webrtc.googlecode.com/svn/trunk@446 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-10Mock implementation for the UI of the linux version of the peerconnection ↵tommi@google.com
client. At this point, there's not a lot too it as it only shows what the UI will look like and basically mimics what the Windows version does presently. Review URL: http://webrtc-codereview.appspot.com/92018 git-svn-id: http://webrtc.googlecode.com/svn/trunk@344 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-05* Point the webrtc libjingle dependency to third_party_mods.ronghuawu@google.com
* For unchanged files, change the third_party_mods libjingle.gyp to point to the original version of libjingle. Review URL: http://webrtc-codereview.appspot.com/89015 git-svn-id: http://webrtc.googlecode.com/svn/trunk@318 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07git-svn-id: http://webrtc.googlecode.com/svn/trunk@170 ↵niklase@google.com
4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-20add command line test app to gyp buildleozwang@google.com
Review URL: http://webrtc-codereview.appspot.com/24017 git-svn-id: http://webrtc.googlecode.com/svn/trunk@105 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-05-31Change the conditions for the peerconnectin_client project.tommi@google.com
Now the project is completely within a "win" condition which should fix the mac build problem. Review URL: http://webrtc-codereview.appspot.com/20021 git-svn-id: http://webrtc.googlecode.com/svn/trunk@20 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-05-30git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 ↵niklase@google.com
4adac7df-926f-26a2-2b94-8c16560cd09d
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