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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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615 outputs = [ | 615 outputs = [ |
616 "{{bundle_resources_dir}}/{{source_file_part}}", | 616 "{{bundle_resources_dir}}/{{source_file_part}}", |
617 ] | 617 ] |
618 } | 618 } |
619 } | 619 } |
620 | 620 |
621 rtc_test("webrtc_perf_tests") { | 621 rtc_test("webrtc_perf_tests") { |
622 testonly = true | 622 testonly = true |
623 configs += [ ":rtc_unittests_config" ] | 623 configs += [ ":rtc_unittests_config" ] |
624 | 624 |
| 625 sources = [ |
| 626 "call/call_perf_tests.cc", |
| 627 "call/rampup_tests.cc", |
| 628 "call/rampup_tests.h", |
| 629 "modules/audio_coding/codecs/opus/opus_complexity_unittest.cc", |
| 630 "modules/audio_coding/neteq/test/neteq_performance_unittest.cc", |
| 631 "modules/audio_processing/audio_processing_performance_unittest.cc", |
| 632 "modules/audio_processing/level_controller/level_controller_complexity_uni
ttest.cc", |
| 633 "modules/audio_processing/residual_echo_detector_complexity_unittest.cc", |
| 634 "modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc", |
| 635 "video/full_stack_tests.cc", |
| 636 ] |
625 deps = [ | 637 deps = [ |
626 "call:call_perf_tests", | 638 "modules/audio_coding:neteq_test_support", |
627 "modules/audio_coding:audio_coding_perf_tests", | 639 "modules/audio_processing", |
628 "modules/audio_processing:audio_processing_perf_tests", | 640 "modules/audio_processing:audioproc_test_utils", |
629 "modules/remote_bitrate_estimator:remote_bitrate_estimator_perf_tests", | 641 "modules/remote_bitrate_estimator:bwe_simulator_lib", |
| 642 "modules/rtp_rtcp", |
| 643 "test:test_common", |
630 "test:test_main", | 644 "test:test_main", |
631 "video:video_full_stack_tests", | 645 "test:test_renderer", |
632 "video:video_quality_test", | 646 "video:video_quality_test", |
| 647 "voice_engine", |
| 648 "//testing/gmock", |
| 649 "//testing/gtest", |
633 ] | 650 ] |
634 | 651 |
| 652 if (rtc_enable_intelligibility_enhancer) { |
| 653 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ] |
| 654 } else { |
| 655 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ] |
| 656 } |
| 657 |
635 data = webrtc_perf_tests_resources | 658 data = webrtc_perf_tests_resources |
636 if (is_android) { | 659 if (is_android) { |
637 deps += [ "//testing/android/native_test:native_test_native_code" ] | 660 deps += [ "//testing/android/native_test:native_test_native_code" ] |
638 shard_timeout = 2700 | 661 shard_timeout = 2700 |
639 } | 662 } |
640 if (is_ios) { | 663 if (is_ios) { |
641 deps += [ ":webrtc_perf_tests_bundle_data" ] | 664 deps += [ ":webrtc_perf_tests_bundle_data" ] |
642 } | 665 } |
| 666 if (!build_with_chromium && is_clang) { |
| 667 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 668 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 669 } |
| 670 if (rtc_use_h264) { |
| 671 defines += [ "WEBRTC_USE_H264" ] |
| 672 } |
643 } | 673 } |
644 | 674 |
645 rtc_test("webrtc_nonparallel_tests") { | 675 rtc_test("webrtc_nonparallel_tests") { |
646 testonly = true | 676 testonly = true |
647 configs += [ ":rtc_unittests_config" ] | 677 configs += [ ":rtc_unittests_config" ] |
648 sources = [ | 678 sources = [ |
649 "base/nullsocketserver_unittest.cc", | 679 "base/nullsocketserver_unittest.cc", |
650 "base/physicalsocketserver_unittest.cc", | 680 "base/physicalsocketserver_unittest.cc", |
651 "base/socket_unittest.cc", | 681 "base/socket_unittest.cc", |
652 "base/socket_unittest.h", | 682 "base/socket_unittest.h", |
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682 ] | 712 ] |
683 | 713 |
684 deps = [ | 714 deps = [ |
685 "//base:base_java_test_support", | 715 "//base:base_java_test_support", |
686 "//webrtc/examples:AppRTCMobile_javalib", | 716 "//webrtc/examples:AppRTCMobile_javalib", |
687 "//webrtc/sdk/android:libjingle_peerconnection_java", | 717 "//webrtc/sdk/android:libjingle_peerconnection_java", |
688 ] | 718 ] |
689 } | 719 } |
690 } | 720 } |
691 } | 721 } |
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