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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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224 "vad/vad_circular_buffer.h", | 224 "vad/vad_circular_buffer.h", |
225 "vad/voice_activity_detector.cc", | 225 "vad/voice_activity_detector.cc", |
226 "vad/voice_activity_detector.h", | 226 "vad/voice_activity_detector.h", |
227 "vad/voice_gmm_tables.h", | 227 "vad/voice_gmm_tables.h", |
228 "voice_detection_impl.cc", | 228 "voice_detection_impl.cc", |
229 "voice_detection_impl.h", | 229 "voice_detection_impl.h", |
230 ] | 230 ] |
231 | 231 |
232 defines = [] | 232 defines = [] |
233 deps = [ | 233 deps = [ |
| 234 ":aec_dump_interface", |
234 "../..:webrtc_common", | 235 "../..:webrtc_common", |
235 "../../audio/utility:audio_frame_operations", | 236 "../../audio/utility:audio_frame_operations", |
236 "../../base:gtest_prod", | 237 "../../base:gtest_prod", |
237 "../audio_coding:isac", | 238 "../audio_coding:isac", |
238 ] | 239 ] |
239 public_deps = [ | 240 public_deps = [ |
240 ":audio_processing_c", | 241 ":audio_processing_c", |
241 ] | 242 ] |
242 | 243 |
243 if (apm_debug_dump) { | 244 if (apm_debug_dump) { |
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296 # TODO(jschuh): Bug 1348: fix this warning. | 297 # TODO(jschuh): Bug 1348: fix this warning. |
297 configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] | 298 configs += [ "//build/config/compiler:no_size_t_to_int_warning" ] |
298 | 299 |
299 deps += [ | 300 deps += [ |
300 "../../base:rtc_base_approved", | 301 "../../base:rtc_base_approved", |
301 "../../common_audio", | 302 "../../common_audio", |
302 "../../system_wrappers", | 303 "../../system_wrappers", |
303 ] | 304 ] |
304 } | 305 } |
305 | 306 |
| 307 rtc_source_set("aec_dump_interface") { |
| 308 # Includes audio_processing.h. TODO(aleloi): add a dependency for |
| 309 # that when the header is not part of :audio_processing. |
| 310 check_includes = false |
| 311 |
| 312 sources = [ |
| 313 "include/aec_dump.cc", |
| 314 "include/aec_dump.h", |
| 315 ] |
| 316 } |
| 317 |
306 rtc_source_set("audio_processing_c") { | 318 rtc_source_set("audio_processing_c") { |
307 visibility = [ ":*" ] # Only targets in this file can depend on this. | 319 visibility = [ ":*" ] # Only targets in this file can depend on this. |
308 sources = [ | 320 sources = [ |
309 "agc/legacy/analog_agc.c", | 321 "agc/legacy/analog_agc.c", |
310 "agc/legacy/analog_agc.h", | 322 "agc/legacy/analog_agc.h", |
311 "agc/legacy/digital_agc.c", | 323 "agc/legacy/digital_agc.c", |
312 "agc/legacy/digital_agc.h", | 324 "agc/legacy/digital_agc.h", |
313 "agc/legacy/gain_control.h", | 325 "agc/legacy/gain_control.h", |
314 ] | 326 ] |
315 | 327 |
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560 } else { | 572 } else { |
561 defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ] | 573 defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ] |
562 } | 574 } |
563 | 575 |
564 if (rtc_enable_protobuf) { | 576 if (rtc_enable_protobuf) { |
565 defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ] | 577 defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ] |
566 deps += [ | 578 deps += [ |
567 ":audioproc_debug_proto", | 579 ":audioproc_debug_proto", |
568 ":audioproc_protobuf_utils", | 580 ":audioproc_protobuf_utils", |
569 ":audioproc_unittest_proto", | 581 ":audioproc_unittest_proto", |
| 582 "aec_dump:aec_dump_unittests", |
570 ] | 583 ] |
571 sources += [ | 584 sources += [ |
572 "aec3/adaptive_fir_filter_unittest.cc", | 585 "aec3/adaptive_fir_filter_unittest.cc", |
573 "aec3/aec3_fft_unittest.cc", | 586 "aec3/aec3_fft_unittest.cc", |
574 "aec3/aec_state_unittest.cc", | 587 "aec3/aec_state_unittest.cc", |
575 "aec3/block_framer_unittest.cc", | 588 "aec3/block_framer_unittest.cc", |
576 "aec3/block_processor_metrics_unittest.cc", | 589 "aec3/block_processor_metrics_unittest.cc", |
577 "aec3/block_processor_unittest.cc", | 590 "aec3/block_processor_unittest.cc", |
578 "aec3/cascaded_biquad_filter_unittest.cc", | 591 "aec3/cascaded_biquad_filter_unittest.cc", |
579 "aec3/comfort_noise_generator_unittest.cc", | 592 "aec3/comfort_noise_generator_unittest.cc", |
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817 ] | 830 ] |
818 | 831 |
819 deps = [ | 832 deps = [ |
820 ":audioproc_debug_proto", | 833 ":audioproc_debug_proto", |
821 "../..:webrtc_common", | 834 "../..:webrtc_common", |
822 "../../base:rtc_base_approved", | 835 "../../base:rtc_base_approved", |
823 ] | 836 ] |
824 } | 837 } |
825 } | 838 } |
826 } | 839 } |
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