webrtc/talk/libjingle_tests.gyp
jbauch ac8869ec5a Report metrics about negotiated ciphers.
This CL adds an API to the metrics observer interface to report negotiated
ciphers for WebRTC sessions. This can be used from Chromium for UMA metrics
later to get an idea which cipher suites are used by clients (e.g. compare
the use of DTLS 1.0 / 1.2).

BUG=428343

Review URL: https://codereview.webrtc.org/1156143005

Cr-Commit-Position: refs/heads/master@{#9537}
2015-07-03 08:36:22 +00:00

445 lines
16 KiB
Python
Executable File

#
# libjingle
# Copyright 2012 Google Inc.
#
# Redistribution and use in source and binary forms, with or without
# modification, are permitted provided that the following conditions are met:
#
# 1. Redistributions of source code must retain the above copyright notice,
# this list of conditions and the following disclaimer.
# 2. Redistributions in binary form must reproduce the above copyright notice,
# this list of conditions and the following disclaimer in the documentation
# and/or other materials provided with the distribution.
# 3. The name of the author may not be used to endorse or promote products
# derived from this software without specific prior written permission.
#
# THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
# MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
# SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
# OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
{
'includes': ['build/common.gypi'],
'targets': [
{
'target_name': 'libjingle_unittest_main',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<@(libjingle_tests_additional_deps)',
],
'direct_dependent_settings': {
'include_dirs': [
'<(libyuv_dir)/include',
'<(DEPTH)/testing/gtest/include',
'<(DEPTH)/testing/gtest',
],
},
'conditions': [
['build_libyuv==1', {
'dependencies': ['<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',],
}],
],
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
'<(DEPTH)/testing/gtest',
],
'sources': [
'media/base/fakecapturemanager.h',
'media/base/fakemediaengine.h',
'media/base/fakemediaprocessor.h',
'media/base/fakenetworkinterface.h',
'media/base/fakertp.h',
'media/base/fakevideocapturer.h',
'media/base/fakevideorenderer.h',
'media/base/nullvideoframe.h',
'media/base/testutils.cc',
'media/base/testutils.h',
'media/devices/fakedevicemanager.h',
'media/webrtc/dummyinstantiation.cc',
'media/webrtc/fakewebrtccall.cc',
'media/webrtc/fakewebrtccall.h',
'media/webrtc/fakewebrtccommon.h',
'media/webrtc/fakewebrtcdeviceinfo.h',
'media/webrtc/fakewebrtcvcmfactory.h',
'media/webrtc/fakewebrtcvideocapturemodule.h',
'media/webrtc/fakewebrtcvideoengine.h',
'media/webrtc/fakewebrtcvoiceengine.h',
],
}, # target libjingle_unittest_main
{
'target_name': 'libjingle_media_unittest',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle_media',
'libjingle_unittest_main',
],
'sources': [
'media/base/capturemanager_unittest.cc',
'media/base/codec_unittest.cc',
'media/base/rtpdataengine_unittest.cc',
'media/base/rtpdump_unittest.cc',
'media/base/rtputils_unittest.cc',
'media/base/streamparams_unittest.cc',
'media/base/testutils.cc',
'media/base/testutils.h',
'media/base/videoadapter_unittest.cc',
'media/base/videocapturer_unittest.cc',
'media/base/videocommon_unittest.cc',
'media/base/videoengine_unittest.h',
'media/devices/dummydevicemanager_unittest.cc',
'media/devices/filevideocapturer_unittest.cc',
'media/sctp/sctpdataengine_unittest.cc',
'media/webrtc/simulcast_unittest.cc',
'media/webrtc/webrtcpassthroughrender_unittest.cc',
'media/webrtc/webrtcvideocapturer_unittest.cc',
'media/base/videoframe_unittest.h',
'media/webrtc/webrtcvideoframe_unittest.cc',
'media/webrtc/webrtcvideoframefactory_unittest.cc',
# Disabled because some tests fail.
# TODO(ronghuawu): Reenable these tests.
# 'media/devices/devicemanager_unittest.cc',
'media/webrtc/webrtcvideoengine2_unittest.cc',
'media/webrtc/webrtcvoiceengine_unittest.cc',
],
'conditions': [
['OS=="win"', {
'conditions': [
['use_openssl==0', {
'dependencies': [
'<(DEPTH)/net/third_party/nss/ssl.gyp:libssl',
'<(DEPTH)/third_party/nss/nss.gyp:nspr',
'<(DEPTH)/third_party/nss/nss.gyp:nss',
],
}],
],
'msvs_settings': {
'VCLinkerTool': {
'AdditionalDependencies': [
# TODO(ronghuawu): Since we've included strmiids in
# libjingle_media target, we shouldn't need this here.
# Find out why it doesn't work without this.
'strmiids.lib',
],
},
},
}],
['OS=="ios"', {
'sources!': [
'media/sctp/sctpdataengine_unittest.cc',
],
}],
],
}, # target libjingle_media_unittest
{
'target_name': 'libjingle_p2p_unittest',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle',
'libjingle.gyp:libjingle_p2p',
'libjingle_unittest_main',
],
'include_dirs': [
'<(DEPTH)/third_party/libsrtp/srtp',
],
'sources': [
'session/media/bundlefilter_unittest.cc',
'session/media/channel_unittest.cc',
'session/media/channelmanager_unittest.cc',
'session/media/currentspeakermonitor_unittest.cc',
'session/media/mediarecorder_unittest.cc',
'session/media/mediasession_unittest.cc',
'session/media/rtcpmuxfilter_unittest.cc',
'session/media/srtpfilter_unittest.cc',
],
'conditions': [
['build_libsrtp==1', {
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
],
}],
['OS=="win"', {
'msvs_settings': {
'VCLinkerTool': {
'AdditionalDependencies': [
'strmiids.lib',
],
},
},
}],
],
}, # target libjingle_p2p_unittest
{
'target_name': 'libjingle_peerconnection_unittest',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/common.gyp:webrtc_common',
'libjingle.gyp:libjingle',
'libjingle.gyp:libjingle_p2p',
'libjingle.gyp:libjingle_peerconnection',
'libjingle_unittest_main',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/testing/gmock/include',
],
},
'sources': [
'app/webrtc/datachannel_unittest.cc',
'app/webrtc/dtlsidentitystore_unittest.cc',
'app/webrtc/dtmfsender_unittest.cc',
'app/webrtc/fakemetricsobserver.cc',
'app/webrtc/fakemetricsobserver.h',
'app/webrtc/jsepsessiondescription_unittest.cc',
'app/webrtc/localaudiosource_unittest.cc',
'app/webrtc/mediastream_unittest.cc',
'app/webrtc/mediastreamhandler_unittest.cc',
'app/webrtc/mediastreamsignaling_unittest.cc',
'app/webrtc/peerconnection_unittest.cc',
'app/webrtc/peerconnectionendtoend_unittest.cc',
'app/webrtc/peerconnectionfactory_unittest.cc',
'app/webrtc/peerconnectioninterface_unittest.cc',
# 'app/webrtc/peerconnectionproxy_unittest.cc',
'app/webrtc/remotevideocapturer_unittest.cc',
'app/webrtc/sctputils.cc',
'app/webrtc/statscollector_unittest.cc',
'app/webrtc/test/fakeaudiocapturemodule.cc',
'app/webrtc/test/fakeaudiocapturemodule.h',
'app/webrtc/test/fakeaudiocapturemodule_unittest.cc',
'app/webrtc/test/fakeconstraints.h',
'app/webrtc/test/fakedatachannelprovider.h',
'app/webrtc/test/fakedtlsidentityservice.h',
'app/webrtc/test/fakemediastreamsignaling.h',
'app/webrtc/test/fakeperiodicvideocapturer.h',
'app/webrtc/test/fakevideotrackrenderer.h',
'app/webrtc/test/mockpeerconnectionobservers.h',
'app/webrtc/test/peerconnectiontestwrapper.h',
'app/webrtc/test/peerconnectiontestwrapper.cc',
'app/webrtc/test/testsdpstrings.h',
'app/webrtc/videosource_unittest.cc',
'app/webrtc/videotrack_unittest.cc',
'app/webrtc/webrtcsdp_unittest.cc',
'app/webrtc/webrtcsession_unittest.cc',
],
'conditions': [
['OS=="android"', {
# We want gmock features that use tr1::tuple, but we currently
# don't support the variadic templates used by libstdc++'s
# implementation. gmock supports this scenario by providing its
# own implementation but we must opt in to it.
'defines': [
'GTEST_USE_OWN_TR1_TUPLE=1',
# GTEST_USE_OWN_TR1_TUPLE only works if GTEST_HAS_TR1_TUPLE is set.
# gmock r625 made it so that GTEST_HAS_TR1_TUPLE is set to 0
# automatically on android, so it has to be set explicitly here.
'GTEST_HAS_TR1_TUPLE=1',
],
}],
],
}, # target libjingle_peerconnection_unittest
],
'conditions': [
['OS=="linux"', {
'variables': {
'junit_jar': '<(DEPTH)/third_party/junit-jar/junit-4.11.jar',
},
'targets': [
{
'target_name': 'libjingle_peerconnection_test_jar',
'type': 'none',
'dependencies': [
'libjingle.gyp:libjingle_peerconnection_jar',
],
'actions': [
{
'variables': {
'java_src_dir': 'app/webrtc/javatests/src',
'java_files': [
'app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java',
'app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java',
],
},
'action_name': 'create_jar',
'inputs': [
'build/build_jar.sh',
'<@(java_files)',
'<(PRODUCT_DIR)/libjingle_peerconnection.jar',
'<(PRODUCT_DIR)/lib/libjingle_peerconnection_so.so',
'<(junit_jar)',
],
'outputs': [
'<(PRODUCT_DIR)/libjingle_peerconnection_test.jar',
],
'action': [
'build/build_jar.sh', '<(java_home)', '<@(_outputs)',
'<(INTERMEDIATE_DIR)',
'<(java_src_dir):<(PRODUCT_DIR)/libjingle_peerconnection.jar:<(junit_jar)',
'<@(java_files)'
],
},
],
},
{
'target_name': 'libjingle_peerconnection_java_unittest',
'type': 'none',
'actions': [
{
'action_name': 'copy libjingle_peerconnection_java_unittest',
'inputs': [
'app/webrtc/javatests/libjingle_peerconnection_java_unittest.sh',
'<(PRODUCT_DIR)/libjingle_peerconnection_test_jar',
'<(junit_jar)',
],
'outputs': [
'<(PRODUCT_DIR)/libjingle_peerconnection_java_unittest',
],
'action': [
'bash', '-c',
'rm -f <(PRODUCT_DIR)/libjingle_peerconnection_java_unittest && '
'sed -e "s@GYP_JAVA_HOME@<(java_home)@" '
'< app/webrtc/javatests/libjingle_peerconnection_java_unittest.sh '
'> <(PRODUCT_DIR)/libjingle_peerconnection_java_unittest && '
'cp <(junit_jar) <(PRODUCT_DIR) && '
'chmod u+x <(PRODUCT_DIR)/libjingle_peerconnection_java_unittest'
],
},
],
},
],
}],
['OS=="android"', {
'targets': [
{
'target_name': 'libjingle_peerconnection_android_unittest',
'type': 'none',
'dependencies': [
'libjingle.gyp:libjingle_peerconnection_java',
],
'variables': {
'apk_name': 'libjingle_peerconnection_android_unittest',
'java_in_dir': 'app/webrtc/androidtests',
'resource_dir': 'app/webrtc/androidtests/res',
'additional_src_dirs': ['app/webrtc/java/testcommon'],
'native_lib_target': 'libjingle_peerconnection_so',
'is_test_apk': 1,
},
'includes': [ '../build/java_apk.gypi' ],
},
], # targets
}], # OS=="android"
['OS=="ios" or (OS=="mac" and target_arch!="ia32" and mac_sdk>="10.7")', {
# The >=10.7 above is required to make ARC link cleanly (e.g. as
# opposed to _compile_ cleanly, which the library under test
# does just fine on 10.6 too).
'targets': [
{
'target_name': 'libjingle_peerconnection_objc_test',
'type': 'executable',
'includes': [ 'build/objc_app.gypi' ],
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'libjingle.gyp:libjingle_peerconnection_objc',
],
'sources': [
'app/webrtc/objctests/RTCPeerConnectionSyncObserver.h',
'app/webrtc/objctests/RTCPeerConnectionSyncObserver.m',
'app/webrtc/objctests/RTCPeerConnectionTest.mm',
'app/webrtc/objctests/RTCSessionDescriptionSyncObserver.h',
'app/webrtc/objctests/RTCSessionDescriptionSyncObserver.m',
# TODO(fischman): figure out if this works for ios or if it
# needs a GUI driver.
'app/webrtc/objctests/mac/main.mm',
],
'conditions': [
['OS=="mac"', {
'xcode_settings': {
# Need to build against 10.7 framework for full ARC support
# on OSX.
'MACOSX_DEPLOYMENT_TARGET' : '10.7',
# common.gypi enables this for mac but we want this to be
# disabled like it is for ios.
'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
},
}],
],
}, # target libjingle_peerconnection_objc_test
{
'target_name': 'apprtc_signaling_gunit_test',
'type': 'executable',
'includes': [ 'build/objc_app.gypi' ],
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(DEPTH)/third_party/ocmock/ocmock.gyp:ocmock',
'libjingle_examples.gyp:apprtc_signaling',
],
'sources': [
'app/webrtc/objctests/mac/main.mm',
'examples/objc/AppRTCDemo/tests/ARDAppClientTest.mm',
],
'conditions': [
['OS=="mac"', {
'xcode_settings': {
'MACOSX_DEPLOYMENT_TARGET' : '10.8',
},
}],
],
}, # target apprtc_signaling_gunit_test
],
}],
['test_isolation_mode != "noop"', {
'targets': [
{
'target_name': 'libjingle_media_unittest_run',
'type': 'none',
'dependencies': [
'libjingle_media_unittest',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'libjingle_media_unittest.isolate',
],
},
{
'target_name': 'libjingle_p2p_unittest_run',
'type': 'none',
'dependencies': [
'libjingle_p2p_unittest',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'libjingle_p2p_unittest.isolate',
],
},
{
'target_name': 'libjingle_peerconnection_unittest_run',
'type': 'none',
'dependencies': [
'libjingle_peerconnection_unittest',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'libjingle_peerconnection_unittest.isolate',
],
},
],
}],
],
}