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Side by Side Diff: webrtc/modules/audio_processing/BUILD.gn

Issue 2747863003: Loosening the coupling between WebRTC and //third_party/protobuf (Closed)
Patch Set: Fixing missing translation from std::string to ProtoString Created 3 years, 9 months 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 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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243 } else { 243 } else {
244 defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ] 244 defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ]
245 } 245 }
246 246
247 if (aec_untrusted_delay_for_testing) { 247 if (aec_untrusted_delay_for_testing) {
248 defines += [ "WEBRTC_UNTRUSTED_DELAY" ] 248 defines += [ "WEBRTC_UNTRUSTED_DELAY" ]
249 } 249 }
250 250
251 if (rtc_enable_protobuf) { 251 if (rtc_enable_protobuf) {
252 defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ] 252 defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
253 deps += [ ":audioproc_debug_proto" ] 253 deps += [
254 ":audioproc_debug_proto",
255 "../../base:protobuf_utils",
256 ]
254 } 257 }
255 258
256 if (rtc_enable_intelligibility_enhancer) { 259 if (rtc_enable_intelligibility_enhancer) {
257 defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ] 260 defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
258 sources += [ 261 sources += [
259 "intelligibility/intelligibility_enhancer.cc", 262 "intelligibility/intelligibility_enhancer.cc",
260 "intelligibility/intelligibility_enhancer.h", 263 "intelligibility/intelligibility_enhancer.h",
261 "intelligibility/intelligibility_utils.cc", 264 "intelligibility/intelligibility_utils.cc",
262 "intelligibility/intelligibility_utils.h", 265 "intelligibility/intelligibility_utils.h",
263 ] 266 ]
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556 } else { 559 } else {
557 defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ] 560 defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ]
558 } 561 }
559 562
560 if (rtc_enable_protobuf) { 563 if (rtc_enable_protobuf) {
561 defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ] 564 defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
562 deps += [ 565 deps += [
563 ":audioproc_debug_proto", 566 ":audioproc_debug_proto",
564 ":audioproc_protobuf_utils", 567 ":audioproc_protobuf_utils",
565 ":audioproc_unittest_proto", 568 ":audioproc_unittest_proto",
569 "../../base:protobuf_utils",
566 ] 570 ]
567 sources += [ 571 sources += [
568 "aec3/adaptive_fir_filter_unittest.cc", 572 "aec3/adaptive_fir_filter_unittest.cc",
569 "aec3/aec3_fft_unittest.cc", 573 "aec3/aec3_fft_unittest.cc",
570 "aec3/aec_state_unittest.cc", 574 "aec3/aec_state_unittest.cc",
571 "aec3/block_framer_unittest.cc", 575 "aec3/block_framer_unittest.cc",
572 "aec3/block_processor_metrics_unittest.cc", 576 "aec3/block_processor_metrics_unittest.cc",
573 "aec3/block_processor_unittest.cc", 577 "aec3/block_processor_unittest.cc",
574 "aec3/cascaded_biquad_filter_unittest.cc", 578 "aec3/cascaded_biquad_filter_unittest.cc",
575 "aec3/comfort_noise_generator_unittest.cc", 579 "aec3/comfort_noise_generator_unittest.cc",
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649 sources = [ 653 sources = [
650 "audio_processing_performance_unittest.cc", 654 "audio_processing_performance_unittest.cc",
651 "level_controller/level_controller_complexity_unittest.cc", 655 "level_controller/level_controller_complexity_unittest.cc",
652 "residual_echo_detector_complexity_unittest.cc", 656 "residual_echo_detector_complexity_unittest.cc",
653 ] 657 ]
654 deps = [ 658 deps = [
655 ":audio_processing", 659 ":audio_processing",
656 ":audioproc_test_utils", 660 ":audioproc_test_utils",
657 "//testing/gtest", 661 "//testing/gtest",
658 ] 662 ]
663
664 if (rtc_enable_protobuf) {
665 deps += [ "../../base:protobuf_utils" ]
666 }
667
659 if (rtc_enable_intelligibility_enhancer) { 668 if (rtc_enable_intelligibility_enhancer) {
660 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ] 669 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
661 } else { 670 } else {
662 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ] 671 defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ]
663 } 672 }
664 } 673 }
665 674
666 if (rtc_enable_protobuf) { 675 if (rtc_enable_protobuf) {
667 rtc_executable("unpack_aecdump") { 676 rtc_executable("unpack_aecdump") {
668 testonly = true 677 testonly = true
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808 817
809 rtc_static_library("audioproc_protobuf_utils") { 818 rtc_static_library("audioproc_protobuf_utils") {
810 sources = [ 819 sources = [
811 "test/protobuf_utils.cc", 820 "test/protobuf_utils.cc",
812 "test/protobuf_utils.h", 821 "test/protobuf_utils.h",
813 ] 822 ]
814 823
815 deps = [ 824 deps = [
816 ":audioproc_debug_proto", 825 ":audioproc_debug_proto",
817 "../..:webrtc_common", 826 "../..:webrtc_common",
827 "../../base:protobuf_utils",
818 "../../base:rtc_base_approved", 828 "../../base:rtc_base_approved",
819 ] 829 ]
820 } 830 }
821 } 831 }
822 } 832 }
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