webrtc_m130/webrtc/webrtc_tests.gypi
Bjorn Terelius c159b046d7 Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly.
Placed the protobuf structures in the namespace webrtc::rtclog. Removed the message field from the DebugEvent structure, since it was not used.

Added an interface to set config information for VideoReceiveStream and VideoSendStream in the event log.

Added function to log full RTCP packets and changed RTP-logging to only log headers.

Significantly extended the unit tests for RtcEventLog.

R=ivoc@webrtc.org, minyue@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1230973005 .

Cr-Commit-Position: refs/heads/master@{#9656}
2015-07-30 09:06:09 +00:00

301 lines
8.8 KiB
Python

# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'rtc_unittests',
'type': '<(gtest_target_type)',
'dependencies': [
'base/base.gyp:rtc_base',
'base/base_tests.gyp:rtc_base_tests_utils',
'base/base_tests.gyp:rtc_base_tests',
'libjingle/xmllite/xmllite.gyp:rtc_xmllite',
'libjingle/xmpp/xmpp.gyp:rtc_xmpp',
'p2p/p2p.gyp:rtc_p2p',
'p2p/p2p.gyp:libstunprober',
'rtc_p2p_unittest',
'rtc_sound_tests',
'rtc_xmllite_unittest',
'rtc_xmpp_unittest',
'sound/sound.gyp:rtc_sound',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/testing/gmock.gyp:gmock',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
],
},
{
'target_name': 'webrtc_tests',
'type': 'none',
'dependencies': [
'video_engine_tests',
'video_loopback',
'video_replay',
'webrtc_perf_tests',
],
},
{
'target_name': 'loopback_base',
'type': 'static_library',
'sources': [
'video/loopback.cc',
'video/loopback.h',
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
'<(webrtc_root)/modules/modules.gyp:video_render',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'webrtc',
],
},
{
'target_name': 'video_loopback',
'type': 'executable',
'sources': [
'test/mac/run_test.mm',
'test/run_test.cc',
'test/run_test.h',
'video/video_loopback.cc',
],
'conditions': [
['OS=="mac"', {
'sources!': [
'test/run_test.cc',
],
}],
],
'dependencies': [
'loopback_base',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'test/webrtc_test_common.gyp:webrtc_test_common',
'test/webrtc_test_common.gyp:webrtc_test_renderer',
'test/test.gyp:test_main',
'webrtc',
],
},
{
'target_name': 'screenshare_loopback',
'type': 'executable',
'sources': [
'test/mac/run_test.mm',
'test/run_test.cc',
'test/run_test.h',
'video/screenshare_loopback.cc',
],
'conditions': [
['OS=="mac"', {
'sources!': [
'test/run_test.cc',
],
}],
],
'dependencies': [
'loopback_base',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'test/webrtc_test_common.gyp:webrtc_test_common',
'test/webrtc_test_common.gyp:webrtc_test_renderer',
'test/test.gyp:test_main',
'webrtc',
],
},
{
'target_name': 'video_replay',
'type': 'executable',
'sources': [
'test/mac/run_test.mm',
'test/run_test.cc',
'test/run_test.h',
'video/replay.cc',
],
'conditions': [
['OS=="mac"', {
'sources!': [
'test/run_test.cc',
],
}],
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'test/webrtc_test_common.gyp:webrtc_test_common',
'test/webrtc_test_common.gyp:webrtc_test_renderer',
'<(webrtc_root)/modules/modules.gyp:video_capture',
'<(webrtc_root)/modules/modules.gyp:video_render',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'webrtc',
],
},
{
# TODO(pbos): Rename target to webrtc_tests or rtc_tests, this target is
# not meant to only include video.
'target_name': 'video_engine_tests',
'type': '<(gtest_target_type)',
'sources': [
'test/common_unittest.cc',
'test/testsupport/metrics/video_metrics_unittest.cc',
'tools/agc/agc_manager_unittest.cc',
'video/bitrate_estimator_tests.cc',
'video/end_to_end_tests.cc',
'video/packet_injection_tests.cc',
'video/send_statistics_proxy_unittest.cc',
'video/video_capture_input_unittest.cc',
'video/video_decoder_unittest.cc',
'video/video_encoder_unittest.cc',
'video/video_send_stream_tests.cc',
],
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/modules/modules.gyp:video_capture',
'<(webrtc_root)/modules/modules.gyp:video_render',
'<(webrtc_root)/test/test.gyp:channel_transport',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'test/metrics.gyp:metrics',
'test/test.gyp:test_main',
'test/webrtc_test_common.gyp:webrtc_test_common',
'tools/tools.gyp:agc_manager',
'webrtc',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
['enable_protobuf==1', {
'defines': [
'ENABLE_RTC_EVENT_LOG',
],
'dependencies': [
'webrtc.gyp:rtc_event_log',
'webrtc.gyp:rtc_event_log_proto',
],
'sources': [
'video/rtc_event_log_unittest.cc',
],
}],
],
},
{
'target_name': 'webrtc_perf_tests',
'type': '<(gtest_target_type)',
'sources': [
'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
'tools/agc/agc_manager_integrationtest.cc',
'video/call_perf_tests.cc',
'video/full_stack.cc',
'video/rampup_tests.cc',
'video/rampup_tests.h',
],
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/modules/modules.gyp:video_capture',
'<(webrtc_root)/test/test.gyp:channel_transport',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'modules/modules.gyp:neteq_test_support',
'modules/modules.gyp:bwe_simulator',
'modules/modules.gyp:rtp_rtcp',
'test/test.gyp:test_main',
'test/webrtc_test_common.gyp:webrtc_test_common',
'tools/tools.gyp:agc_manager',
'webrtc',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
],
},
],
'conditions': [
['OS=="android"', {
'targets': [
{
'target_name': 'rtc_unittests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):rtc_unittests_apk',
],
},
{
'target_name': 'video_engine_tests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):video_engine_tests_apk',
],
},
{
'target_name': 'webrtc_perf_tests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):webrtc_perf_tests_apk',
],
},
],
}],
['test_isolation_mode != "noop"', {
'targets': [
{
'target_name': 'rtc_unittests_run',
'type': 'none',
'dependencies': [
'rtc_unittests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'rtc_unittests.isolate',
],
},
{
'target_name': 'video_engine_tests_run',
'type': 'none',
'dependencies': [
'video_engine_tests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'video_engine_tests.isolate',
],
},
{
'target_name': 'webrtc_perf_tests_run',
'type': 'none',
'dependencies': [
'webrtc_perf_tests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'webrtc_perf_tests.isolate',
],
},
],
}],
],
}