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Issue 2438973002: GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn (Closed)
Patch Set: Add missing dependency on modules/video_capture:video_capture_module Created 4 years, 1 month ago
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1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
2 # 2 #
3 # Use of this source code is governed by a BSD-style license 3 # Use of this source code is governed by a BSD-style license
4 # that can be found in the LICENSE file in the root of the source 4 # that can be found in the LICENSE file in the root of the source
5 # tree. An additional intellectual property rights grant can be found 5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may 6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree. 7 # be found in the AUTHORS file in the root of the source tree.
8 8
9 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330. 9 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
10 10
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605 if (is_android) { 605 if (is_android) {
606 deps += [ "//testing/android/native_test:native_test_native_code" ] 606 deps += [ "//testing/android/native_test:native_test_native_code" ]
607 shard_timeout = 900 607 shard_timeout = 900
608 } 608 }
609 609
610 if (is_ios) { 610 if (is_ios) {
611 deps += [ ":video_engine_tests_bundle_data" ] 611 deps += [ ":video_engine_tests_bundle_data" ]
612 } 612 }
613 } 613 }
614 614
615 rtc_source_set("video_quality_test") {
616 testonly = true
617 sources = [
618 "video/video_quality_test.cc",
619 "video/video_quality_test.h",
620 ]
621 deps = [
622 ":webrtc",
623 "system_wrappers",
624 "//testing/gtest",
625 ]
626 if (!is_android) {
627 deps += [ "modules/video_capture:video_capture_internal_impl" ]
628 }
629 if (!build_with_chromium && is_clang) {
630 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
631 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
632 }
633 }
634
635 webrtc_perf_tests_resources = [ 615 webrtc_perf_tests_resources = [
636 "//resources/audio_coding/speech_mono_16kHz.pcm", 616 "//resources/audio_coding/speech_mono_16kHz.pcm",
637 "//resources/audio_coding/testfile32kHz.pcm", 617 "//resources/audio_coding/testfile32kHz.pcm",
638 "//resources/ConferenceMotion_1280_720_50.yuv", 618 "//resources/ConferenceMotion_1280_720_50.yuv",
639 "//resources/difficult_photo_1850_1110.yuv", 619 "//resources/difficult_photo_1850_1110.yuv",
640 "//resources/foreman_cif.yuv", 620 "//resources/foreman_cif.yuv",
641 "//resources/google-wifi-3mbps.rx", 621 "//resources/google-wifi-3mbps.rx",
642 "//resources/paris_qcif.yuv", 622 "//resources/paris_qcif.yuv",
643 "//resources/photo_1850_1110.yuv", 623 "//resources/photo_1850_1110.yuv",
644 "//resources/presentation_1850_1110.yuv", 624 "//resources/presentation_1850_1110.yuv",
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659 639
660 rtc_executable("webrtc_tests") { 640 rtc_executable("webrtc_tests") {
661 testonly = true 641 testonly = true
662 deps = [ 642 deps = [
663 ":webrtc", 643 ":webrtc",
664 "modules/video_capture:video_capture_internal_impl", 644 "modules/video_capture:video_capture_internal_impl",
665 "test", 645 "test",
666 ] 646 ]
667 } 647 }
668 648
669 rtc_executable("video_loopback") {
670 testonly = true
671 sources = [
672 "test/run_test.h",
673 "video/video_loopback.cc",
674 ]
675
676 if (is_mac) {
677 sources += [ "test/mac/run_test.mm" ]
678 } else {
679 sources += [ "test/run_test.cc" ]
680 }
681 deps = [
682 ":video_quality_test",
683 "system_wrappers:metrics_default",
684 "test:field_trial",
685 "test:test_common",
686 "test:test_renderer",
687 "//testing/gmock",
688 "//testing/gtest",
689 "//third_party/gflags",
690 ]
691 if (!build_with_chromium && is_clang) {
692 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
693 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
694 }
695 }
696
697 rtc_executable("screenshare_loopback") {
698 testonly = true
699 sources = [
700 "test/run_test.h",
701 "video/screenshare_loopback.cc",
702 ]
703
704 if (is_mac) {
705 sources += [ "test/mac/run_test.mm" ]
706 } else {
707 sources += [ "test/run_test.cc" ]
708 }
709 deps = [
710 ":video_quality_test",
711 "system_wrappers:metrics_default",
712 "test:field_trial",
713 "test:test_common",
714 "test:test_renderer",
715 "//testing/gmock",
716 "//testing/gtest",
717 "//third_party/gflags",
718 ]
719 if (is_clang && !is_nacl) {
720 # Suppress warnings from Chrome's Clang plugins.
721 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
722 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
723 }
724 }
725
726 rtc_test("webrtc_perf_tests") { 649 rtc_test("webrtc_perf_tests") {
727 testonly = true 650 testonly = true
728 configs += [ ":rtc_unittests_config" ] 651 configs += [ ":rtc_unittests_config" ]
729 652
730 sources = [ 653 sources = [
731 "call/call_perf_tests.cc", 654 "call/call_perf_tests.cc",
732 "call/rampup_tests.cc", 655 "call/rampup_tests.cc",
733 "call/rampup_tests.h", 656 "call/rampup_tests.h",
734 "modules/audio_coding/neteq/test/neteq_performance_unittest.cc", 657 "modules/audio_coding/neteq/test/neteq_performance_unittest.cc",
735 "modules/audio_processing/audio_processing_performance_unittest.cc", 658 "modules/audio_processing/audio_processing_performance_unittest.cc",
736 "modules/audio_processing/level_controller/level_controller_complexity_uni ttest.cc", 659 "modules/audio_processing/level_controller/level_controller_complexity_uni ttest.cc",
737 "modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc", 660 "modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc",
738 "video/full_stack.cc", 661 "video/full_stack.cc",
739 ] 662 ]
740 deps = [ 663 deps = [
741 ":video_quality_test",
742 ":webrtc",
743 "modules/audio_coding:neteq_test_support", 664 "modules/audio_coding:neteq_test_support",
744 "modules/audio_processing", 665 "modules/audio_processing",
745 "modules/audio_processing:audioproc_test_utils", 666 "modules/audio_processing:audioproc_test_utils",
746 "modules/remote_bitrate_estimator:bwe_simulator_lib", 667 "modules/remote_bitrate_estimator:bwe_simulator_lib",
747 "modules/rtp_rtcp", 668 "modules/rtp_rtcp",
748 "test:test_common", 669 "test:test_common",
749 "test:test_main", 670 "test:test_main",
750 "test:test_renderer", 671 "test:test_renderer",
672 "video:video_quality_test",
751 "voice_engine", 673 "voice_engine",
752 "//testing/gmock", 674 "//testing/gmock",
753 "//testing/gtest", 675 "//testing/gtest",
754 ] 676 ]
755 677
756 if (rtc_enable_intelligibility_enhancer) { 678 if (rtc_enable_intelligibility_enhancer) {
757 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ] 679 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
758 } else { 680 } else {
759 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ] 681 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
760 } 682 }
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813 735
814 deps = [ 736 deps = [
815 "//base:base_java_test_support", 737 "//base:base_java_test_support",
816 "//webrtc/api:libjingle_peerconnection_java", 738 "//webrtc/api:libjingle_peerconnection_java",
817 "//webrtc/api:libjingle_peerconnection_jni", 739 "//webrtc/api:libjingle_peerconnection_jni",
818 "//webrtc/examples:AppRTCMobile_javalib", 740 "//webrtc/examples:AppRTCMobile_javalib",
819 ] 741 ]
820 } 742 }
821 } 743 }
822 } 744 }
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