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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 import("//build/config/arm.gni") | 9 import("//build/config/arm.gni") |
10 import("//third_party/protobuf/proto_library.gni") | 10 import("//third_party/protobuf/proto_library.gni") |
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552 "..:module_api", | 552 "..:module_api", |
553 "../..:webrtc_common", | 553 "../..:webrtc_common", |
554 "../../base:gtest_prod", | 554 "../../base:gtest_prod", |
555 "../../base:protobuf_utils", | 555 "../../base:protobuf_utils", |
556 "../../base:rtc_base", | 556 "../../base:rtc_base", |
557 "../../base:rtc_base_approved", | 557 "../../base:rtc_base_approved", |
558 "../../common_audio:common_audio", | 558 "../../common_audio:common_audio", |
559 "../../system_wrappers:system_wrappers", | 559 "../../system_wrappers:system_wrappers", |
560 "../../test:test_support", | 560 "../../test:test_support", |
561 "../audio_coding:neteq_tools", | 561 "../audio_coding:neteq_tools", |
| 562 "aec_dump:mock_aec_dump_unittests", |
562 "test/conversational_speech:unittest", | 563 "test/conversational_speech:unittest", |
563 "//testing/gmock", | 564 "//testing/gmock", |
564 "//testing/gtest", | 565 "//testing/gtest", |
565 ] | 566 ] |
566 | 567 |
567 defines = [] | 568 defines = [] |
568 | 569 |
569 if (apm_debug_dump) { | 570 if (apm_debug_dump) { |
570 defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ] | 571 defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ] |
571 } else { | 572 } else { |
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859 | 860 |
860 deps = [ | 861 deps = [ |
861 ":audioproc_debug_proto", | 862 ":audioproc_debug_proto", |
862 "../..:webrtc_common", | 863 "../..:webrtc_common", |
863 "../../base:protobuf_utils", | 864 "../../base:protobuf_utils", |
864 "../../base:rtc_base_approved", | 865 "../../base:rtc_base_approved", |
865 ] | 866 ] |
866 } | 867 } |
867 } | 868 } |
868 } | 869 } |
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