2013-10-28 16:32:01 +00:00
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# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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{
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'targets': [
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2014-09-10 17:28:19 +00:00
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{
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'target_name': 'rtc_unittests',
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2015-03-04 08:47:39 +00:00
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'type': '<(gtest_target_type)',
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2014-09-10 17:28:19 +00:00
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'dependencies': [
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'base/base.gyp:rtc_base',
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'base/base_tests.gyp:rtc_base_tests_utils',
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'base/base_tests.gyp:rtc_base_tests',
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2014-10-02 18:43:47 +00:00
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'libjingle/xmllite/xmllite.gyp:rtc_xmllite',
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2014-10-28 22:20:11 +00:00
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'libjingle/xmpp/xmpp.gyp:rtc_xmpp',
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'p2p/p2p.gyp:rtc_p2p',
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2015-05-15 10:26:52 -07:00
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'p2p/p2p.gyp:libstunprober',
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2014-10-28 22:20:11 +00:00
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'rtc_p2p_unittest',
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2014-10-01 16:33:03 +00:00
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'rtc_sound_tests',
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2014-10-02 18:43:47 +00:00
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'rtc_xmllite_unittest',
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2014-10-28 22:20:11 +00:00
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'rtc_xmpp_unittest',
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2014-09-30 14:21:10 +00:00
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'sound/sound.gyp:rtc_sound',
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2014-09-10 17:28:19 +00:00
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'<(DEPTH)/testing/gtest.gyp:gtest',
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2015-05-15 10:26:52 -07:00
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'<(DEPTH)/testing/gmock.gyp:gmock',
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2014-09-10 17:28:19 +00:00
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],
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2015-03-04 08:47:39 +00:00
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'conditions': [
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['OS=="android"', {
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'dependencies': [
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'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
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],
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}],
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2015-12-13 19:58:11 -08:00
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['OS=="ios"', {
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'dependencies': [
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2016-01-27 17:50:11 -08:00
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'api/api_tests.gyp:rtc_api_objc_tests',
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2015-12-13 19:58:11 -08:00
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]
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}]
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2015-03-04 08:47:39 +00:00
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],
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2014-09-10 17:28:19 +00:00
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},
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2013-10-28 16:32:01 +00:00
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{
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'target_name': 'webrtc_tests',
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'type': 'none',
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'dependencies': [
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'video_engine_tests',
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'video_loopback',
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2014-08-06 16:26:56 +00:00
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'video_replay',
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2013-12-13 12:48:05 +00:00
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'webrtc_perf_tests',
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2015-10-29 12:41:56 +01:00
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'webrtc_nonparallel_tests',
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2013-10-28 16:32:01 +00:00
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],
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},
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2015-09-07 06:04:16 -07:00
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{
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2015-09-17 05:30:24 -07:00
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'target_name': 'video_quality_test',
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2015-02-18 12:46:06 +00:00
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'type': 'static_library',
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'sources': [
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2015-09-17 05:30:24 -07:00
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'video/video_quality_test.cc',
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'video/video_quality_test.h',
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2015-02-18 12:46:06 +00:00
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],
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'dependencies': [
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(webrtc_root)/modules/modules.gyp:video_render',
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2015-09-17 05:30:24 -07:00
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'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
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2015-02-18 12:46:06 +00:00
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
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'webrtc',
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],
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2015-09-17 05:30:24 -07:00
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'conditions': [
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['OS=="android"', {
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'dependencies!': [
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'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
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],
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}],
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],
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2015-02-18 12:46:06 +00:00
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},
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2013-10-28 16:32:01 +00:00
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{
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'target_name': 'video_loopback',
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'type': 'executable',
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'sources': [
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2014-06-05 09:32:51 +00:00
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'test/mac/run_test.mm',
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'test/run_test.cc',
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'test/run_test.h',
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2015-02-18 12:46:06 +00:00
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'video/video_loopback.cc',
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2013-10-28 16:32:01 +00:00
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],
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2014-06-05 09:32:51 +00:00
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'conditions': [
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['OS=="mac"', {
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'sources!': [
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'test/run_test.cc',
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],
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}],
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],
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2013-10-28 16:32:01 +00:00
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'dependencies': [
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2015-09-17 05:30:24 -07:00
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'video_quality_test',
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2015-02-18 12:46:06 +00:00
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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'test/webrtc_test_common.gyp:webrtc_test_common',
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'test/webrtc_test_common.gyp:webrtc_test_renderer',
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'test/test.gyp:test_main',
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'webrtc',
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],
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},
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2015-06-26 06:58:16 +02:00
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{
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2015-02-18 12:46:06 +00:00
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'target_name': 'screenshare_loopback',
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'type': 'executable',
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'sources': [
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'test/mac/run_test.mm',
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'test/run_test.cc',
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'test/run_test.h',
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'video/screenshare_loopback.cc',
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],
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'conditions': [
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['OS=="mac"', {
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'sources!': [
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'test/run_test.cc',
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],
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}],
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],
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'dependencies': [
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2015-09-17 05:30:24 -07:00
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'video_quality_test',
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2013-10-28 16:32:01 +00:00
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'<(DEPTH)/testing/gtest.gyp:gtest',
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2014-05-13 11:26:40 +00:00
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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2013-10-28 16:32:01 +00:00
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'test/webrtc_test_common.gyp:webrtc_test_common',
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2014-06-05 09:32:51 +00:00
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'test/webrtc_test_common.gyp:webrtc_test_renderer',
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2014-11-04 14:57:14 +00:00
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'test/test.gyp:test_main',
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2013-10-28 16:32:01 +00:00
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'webrtc',
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],
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},
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2014-08-06 16:26:56 +00:00
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{
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'target_name': 'video_replay',
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'type': 'executable',
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'sources': [
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'test/mac/run_test.mm',
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'test/run_test.cc',
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'test/run_test.h',
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'video/replay.cc',
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],
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'conditions': [
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['OS=="mac"', {
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'sources!': [
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'test/run_test.cc',
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],
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}],
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],
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'dependencies': [
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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'test/webrtc_test_common.gyp:webrtc_test_common',
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'test/webrtc_test_common.gyp:webrtc_test_renderer',
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2015-02-11 07:47:00 +00:00
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'<(webrtc_root)/modules/modules.gyp:video_capture',
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'<(webrtc_root)/modules/modules.gyp:video_render',
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2015-01-14 09:30:52 +00:00
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'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
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2014-08-06 16:26:56 +00:00
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'webrtc',
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],
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},
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2013-10-28 16:32:01 +00:00
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{
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2015-10-16 14:35:07 -07:00
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# TODO(solenberg): Rename to webrtc_call_tests.
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2013-10-28 16:32:01 +00:00
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'target_name': 'video_engine_tests',
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'type': '<(gtest_target_type)',
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'sources': [
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2015-09-25 13:58:30 +02:00
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'audio/audio_receive_stream_unittest.cc',
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2015-10-16 14:35:07 -07:00
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'audio/audio_send_stream_unittest.cc',
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2015-11-06 15:34:49 -08:00
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'audio/audio_state_unittest.cc',
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2015-11-12 21:02:42 -08:00
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'call/bitrate_allocator_unittest.cc',
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2015-09-25 13:58:30 +02:00
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'call/bitrate_estimator_tests.cc',
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2015-10-16 14:35:07 -07:00
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'call/call_unittest.cc',
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2015-09-25 13:58:30 +02:00
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'call/packet_injection_tests.cc',
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2015-01-29 14:29:45 +00:00
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'test/common_unittest.cc',
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'test/testsupport/metrics/video_metrics_unittest.cc',
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2015-12-09 12:13:30 +01:00
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'video/call_stats_unittest.cc',
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'video/encoder_state_feedback_unittest.cc',
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2014-06-27 08:47:52 +00:00
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'video/end_to_end_tests.cc',
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2015-12-09 12:13:30 +01:00
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'video/overuse_frame_detector_unittest.cc',
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'video/payload_router_unittest.cc',
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'video/report_block_stats_unittest.cc',
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2014-01-07 09:54:34 +00:00
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'video/send_statistics_proxy_unittest.cc',
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2015-12-09 12:13:30 +01:00
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'video/stream_synchronization_unittest.cc',
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2015-06-26 06:58:16 +02:00
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'video/video_capture_input_unittest.cc',
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2015-05-18 19:42:03 +02:00
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'video/video_decoder_unittest.cc',
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2015-05-19 23:09:35 +02:00
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'video/video_encoder_unittest.cc',
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2013-10-28 16:32:01 +00:00
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'video/video_send_stream_tests.cc',
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2015-12-09 12:13:30 +01:00
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'video/vie_remb_unittest.cc',
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2013-10-28 16:32:01 +00:00
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],
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'dependencies': [
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2014-12-16 14:48:47 +00:00
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'<(DEPTH)/testing/gmock.gyp:gmock',
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2013-10-28 16:32:01 +00:00
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'<(DEPTH)/testing/gtest.gyp:gtest',
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2015-02-26 13:59:22 +00:00
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'<(webrtc_root)/common.gyp:webrtc_common',
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2014-09-18 08:58:15 +00:00
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'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
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2015-02-11 07:47:00 +00:00
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'<(webrtc_root)/modules/modules.gyp:video_capture',
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'<(webrtc_root)/modules/modules.gyp:video_render',
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2014-12-16 14:48:47 +00:00
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'<(webrtc_root)/test/test.gyp:channel_transport',
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'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
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2013-12-13 14:31:47 +00:00
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'test/metrics.gyp:metrics',
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2014-05-14 12:24:04 +00:00
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'test/test.gyp:test_main',
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2014-12-16 14:48:47 +00:00
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'test/webrtc_test_common.gyp:webrtc_test_common',
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2013-10-28 16:32:01 +00:00
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'webrtc',
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],
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2014-05-27 18:44:59 +00:00
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'conditions': [
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2016-01-27 01:36:03 -08:00
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['rtc_use_h264==1', {
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'defines': [
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'WEBRTC_END_TO_END_H264_TESTS',
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],
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}],
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2014-09-02 08:40:39 +00:00
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['OS=="android"', {
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2014-05-27 18:44:59 +00:00
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'dependencies': [
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'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
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],
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}],
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2015-07-30 12:45:18 +02:00
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['enable_protobuf==1', {
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'defines': [
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'ENABLE_RTC_EVENT_LOG',
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],
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'dependencies': [
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'webrtc.gyp:rtc_event_log',
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'webrtc.gyp:rtc_event_log_proto',
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],
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'sources': [
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2015-09-25 13:58:30 +02:00
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'call/rtc_event_log_unittest.cc',
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2015-07-30 12:45:18 +02:00
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],
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}],
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2014-05-27 18:44:59 +00:00
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],
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2013-10-28 16:32:01 +00:00
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},
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2013-12-13 12:48:05 +00:00
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{
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'target_name': 'webrtc_perf_tests',
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'type': '<(gtest_target_type)',
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'sources': [
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2015-09-25 13:58:30 +02:00
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'call/call_perf_tests.cc',
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2015-12-21 03:14:00 -08:00
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'call/rampup_tests.cc',
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'call/rampup_tests.h',
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2014-06-09 08:10:28 +00:00
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'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
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2015-12-20 23:13:40 -08:00
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'modules/audio_processing/audio_processing_performance_unittest.cc',
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2015-04-16 20:20:36 +02:00
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'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
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2013-12-13 12:48:05 +00:00
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'video/full_stack.cc',
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],
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'dependencies': [
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2014-12-16 14:48:47 +00:00
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'<(DEPTH)/testing/gmock.gyp:gmock',
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2013-12-13 12:48:05 +00:00
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'<(DEPTH)/testing/gtest.gyp:gtest',
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2015-12-20 23:13:40 -08:00
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'<(webrtc_root)/modules/modules.gyp:audio_processing',
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'<(webrtc_root)/modules/modules.gyp:audioproc_test_utils',
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2015-02-11 07:47:00 +00:00
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'<(webrtc_root)/modules/modules.gyp:video_capture',
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2014-12-16 14:48:47 +00:00
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'<(webrtc_root)/test/test.gyp:channel_transport',
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'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
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2015-09-17 05:30:24 -07:00
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'video_quality_test',
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2015-04-16 20:20:36 +02:00
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'modules/modules.gyp:neteq_test_support',
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'modules/modules.gyp:bwe_simulator',
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2013-12-13 12:48:05 +00:00
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'modules/modules.gyp:rtp_rtcp',
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2014-05-14 12:24:04 +00:00
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'test/test.gyp:test_main',
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2014-12-16 14:48:47 +00:00
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'test/webrtc_test_common.gyp:webrtc_test_common',
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2015-09-17 05:30:24 -07:00
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'test/webrtc_test_common.gyp:webrtc_test_renderer',
|
2013-12-13 12:48:05 +00:00
|
|
|
'webrtc',
|
|
|
|
|
],
|
2014-05-30 12:42:10 +00:00
|
|
|
'conditions': [
|
2014-09-01 11:06:37 +00:00
|
|
|
['OS=="android"', {
|
2014-05-30 12:42:10 +00:00
|
|
|
'dependencies': [
|
|
|
|
|
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
|
|
|
|
|
],
|
|
|
|
|
}],
|
|
|
|
|
],
|
2013-12-13 12:48:05 +00:00
|
|
|
},
|
2015-10-29 12:41:56 +01:00
|
|
|
{
|
|
|
|
|
'target_name': 'webrtc_nonparallel_tests',
|
|
|
|
|
'type': '<(gtest_target_type)',
|
|
|
|
|
'sources': [
|
|
|
|
|
'base/nullsocketserver_unittest.cc',
|
|
|
|
|
'base/physicalsocketserver_unittest.cc',
|
|
|
|
|
'base/socket_unittest.cc',
|
|
|
|
|
'base/socket_unittest.h',
|
|
|
|
|
'base/socketaddress_unittest.cc',
|
|
|
|
|
'base/virtualsocket_unittest.cc',
|
|
|
|
|
],
|
|
|
|
|
'defines': [
|
|
|
|
|
'GTEST_RELATIVE_PATH',
|
|
|
|
|
],
|
|
|
|
|
'dependencies': [
|
|
|
|
|
'<(DEPTH)/testing/gtest.gyp:gtest',
|
|
|
|
|
'base/base.gyp:rtc_base',
|
|
|
|
|
'test/test.gyp:test_main',
|
|
|
|
|
],
|
|
|
|
|
'conditions': [
|
|
|
|
|
['OS=="android"', {
|
|
|
|
|
'dependencies': [
|
|
|
|
|
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
|
|
|
|
|
],
|
|
|
|
|
}],
|
|
|
|
|
['OS=="win"', {
|
|
|
|
|
'sources': [
|
|
|
|
|
'base/win32socketserver_unittest.cc',
|
|
|
|
|
],
|
|
|
|
|
'sources!': [
|
|
|
|
|
# TODO(ronghuawu): Fix TestUdpReadyToSendIPv6 on windows bot
|
|
|
|
|
# then reenable these tests.
|
|
|
|
|
# TODO(pbos): Move test disabling to ifdefs within the test files
|
|
|
|
|
# instead of here.
|
|
|
|
|
'base/physicalsocketserver_unittest.cc',
|
|
|
|
|
'base/socket_unittest.cc',
|
|
|
|
|
'base/win32socketserver_unittest.cc',
|
|
|
|
|
],
|
|
|
|
|
}],
|
|
|
|
|
['OS=="mac"', {
|
|
|
|
|
'sources': [
|
|
|
|
|
'base/macsocketserver_unittest.cc',
|
|
|
|
|
],
|
|
|
|
|
}],
|
|
|
|
|
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
|
|
|
|
|
'defines': [
|
|
|
|
|
'CARBON_DEPRECATED=YES',
|
|
|
|
|
],
|
|
|
|
|
}],
|
|
|
|
|
],
|
|
|
|
|
},
|
2013-10-28 16:32:01 +00:00
|
|
|
],
|
|
|
|
|
'conditions': [
|
2014-09-01 11:06:37 +00:00
|
|
|
['OS=="android"', {
|
2013-10-28 16:32:01 +00:00
|
|
|
'targets': [
|
2015-03-04 08:47:39 +00:00
|
|
|
{
|
|
|
|
|
'target_name': 'rtc_unittests_apk_target',
|
|
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
|
|
|
|
'<(apk_tests_path):rtc_unittests_apk',
|
|
|
|
|
],
|
|
|
|
|
},
|
2013-10-28 16:32:01 +00:00
|
|
|
{
|
2014-09-03 14:34:46 +00:00
|
|
|
'target_name': 'video_engine_tests_apk_target',
|
2013-10-28 16:32:01 +00:00
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
2014-09-03 14:34:46 +00:00
|
|
|
'<(apk_tests_path):video_engine_tests_apk',
|
2013-10-28 16:32:01 +00:00
|
|
|
],
|
|
|
|
|
},
|
2013-12-13 12:48:05 +00:00
|
|
|
{
|
2014-09-03 14:34:46 +00:00
|
|
|
'target_name': 'webrtc_perf_tests_apk_target',
|
2013-12-13 12:48:05 +00:00
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
2014-09-03 14:34:46 +00:00
|
|
|
'<(apk_tests_path):webrtc_perf_tests_apk',
|
2013-12-13 12:48:05 +00:00
|
|
|
],
|
|
|
|
|
},
|
2015-10-29 12:41:56 +01:00
|
|
|
{
|
|
|
|
|
'target_name': 'webrtc_nonparallel_tests_apk_target',
|
|
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
|
|
|
|
'<(apk_tests_path):webrtc_nonparallel_tests_apk',
|
|
|
|
|
],
|
|
|
|
|
},
|
2013-10-28 16:32:01 +00:00
|
|
|
],
|
|
|
|
|
}],
|
|
|
|
|
['test_isolation_mode != "noop"', {
|
|
|
|
|
'targets': [
|
2014-09-30 14:21:10 +00:00
|
|
|
{
|
|
|
|
|
'target_name': 'rtc_unittests_run',
|
|
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
|
|
|
|
'rtc_unittests',
|
|
|
|
|
],
|
|
|
|
|
'includes': [
|
|
|
|
|
'build/isolate.gypi',
|
|
|
|
|
],
|
|
|
|
|
'sources': [
|
|
|
|
|
'rtc_unittests.isolate',
|
|
|
|
|
],
|
Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
|
|
|
},
|
|
|
|
|
{
|
2016-02-12 15:06:02 +01:00
|
|
|
'target_name': 'rtc_media_unittests_run',
|
Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
2016-02-12 15:06:02 +01:00
|
|
|
'rtc_media_unittests',
|
Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
|
|
|
],
|
|
|
|
|
'includes': [
|
|
|
|
|
'build/isolate.gypi',
|
|
|
|
|
],
|
|
|
|
|
'sources': [
|
2016-02-12 15:06:02 +01:00
|
|
|
'rtc_media_unittests.isolate',
|
Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
|
|
|
],
|
2016-02-12 06:47:59 +01:00
|
|
|
},
|
|
|
|
|
{
|
2016-02-15 09:36:31 +01:00
|
|
|
'target_name': 'rtc_pc_unittests_run',
|
2016-02-12 06:47:59 +01:00
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
2016-02-15 09:36:31 +01:00
|
|
|
'rtc_pc_unittests',
|
2016-02-12 06:47:59 +01:00
|
|
|
],
|
|
|
|
|
'includes': [
|
|
|
|
|
'build/isolate.gypi',
|
|
|
|
|
],
|
|
|
|
|
'sources': [
|
2016-02-15 09:36:31 +01:00
|
|
|
'rtc_pc_unittests.isolate',
|
2016-02-12 06:47:59 +01:00
|
|
|
],
|
2014-09-30 14:21:10 +00:00
|
|
|
},
|
2013-10-28 16:32:01 +00:00
|
|
|
{
|
|
|
|
|
'target_name': 'video_engine_tests_run',
|
|
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
|
|
|
|
'video_engine_tests',
|
|
|
|
|
],
|
|
|
|
|
'includes': [
|
2013-10-28 18:10:29 +00:00
|
|
|
'build/isolate.gypi',
|
2013-10-28 16:32:01 +00:00
|
|
|
],
|
|
|
|
|
'sources': [
|
|
|
|
|
'video_engine_tests.isolate',
|
|
|
|
|
],
|
|
|
|
|
},
|
2015-10-29 15:29:04 +01:00
|
|
|
{
|
|
|
|
|
'target_name': 'webrtc_nonparallel_tests_run',
|
|
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
|
|
|
|
'webrtc_nonparallel_tests',
|
|
|
|
|
],
|
|
|
|
|
'includes': [
|
|
|
|
|
'build/isolate.gypi',
|
|
|
|
|
],
|
|
|
|
|
'sources': [
|
|
|
|
|
'webrtc_nonparallel_tests.isolate',
|
|
|
|
|
],
|
|
|
|
|
},
|
2013-12-13 12:48:05 +00:00
|
|
|
{
|
|
|
|
|
'target_name': 'webrtc_perf_tests_run',
|
|
|
|
|
'type': 'none',
|
|
|
|
|
'dependencies': [
|
|
|
|
|
'webrtc_perf_tests',
|
|
|
|
|
],
|
|
|
|
|
'includes': [
|
|
|
|
|
'build/isolate.gypi',
|
|
|
|
|
],
|
|
|
|
|
'sources': [
|
|
|
|
|
'webrtc_perf_tests.isolate',
|
|
|
|
|
],
|
|
|
|
|
},
|
2013-10-28 16:32:01 +00:00
|
|
|
],
|
|
|
|
|
}],
|
|
|
|
|
],
|
|
|
|
|
}
|