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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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408 | 408 |
409 if (rtc_enable_protobuf) { | 409 if (rtc_enable_protobuf) { |
410 public_deps += [ | 410 public_deps += [ |
411 ":audioproc_f", | 411 ":audioproc_f", |
412 ":audioproc_unittest_proto", | 412 ":audioproc_unittest_proto", |
413 ":unpack_aecdump", | 413 ":unpack_aecdump", |
414 ] | 414 ] |
415 } | 415 } |
416 } | 416 } |
417 | 417 |
| 418 rtc_source_set("audio_processing_unittests") { |
| 419 testonly = true |
| 420 sources = [ |
| 421 "aec/echo_cancellation_unittest.cc", |
| 422 "aec/system_delay_unittest.cc", |
| 423 "agc/agc_manager_direct_unittest.cc", |
| 424 "agc/loudness_histogram_unittest.cc", |
| 425 "agc/mock_agc.h", |
| 426 "audio_buffer_unittest.cc", |
| 427 "beamformer/array_util_unittest.cc", |
| 428 "beamformer/complex_matrix_unittest.cc", |
| 429 "beamformer/covariance_matrix_generator_unittest.cc", |
| 430 "beamformer/matrix_unittest.cc", |
| 431 "beamformer/mock_nonlinear_beamformer.h", |
| 432 "beamformer/nonlinear_beamformer_unittest.cc", |
| 433 "config_unittest.cc", |
| 434 "echo_cancellation_impl_unittest.cc", |
| 435 "splitting_filter_unittest.cc", |
| 436 "transient/dyadic_decimator_unittest.cc", |
| 437 "transient/file_utils.cc", |
| 438 "transient/file_utils.h", |
| 439 "transient/file_utils_unittest.cc", |
| 440 "transient/moving_moments_unittest.cc", |
| 441 "transient/transient_detector_unittest.cc", |
| 442 "transient/transient_suppressor_unittest.cc", |
| 443 "transient/wpd_node_unittest.cc", |
| 444 "transient/wpd_tree_unittest.cc", |
| 445 "utility/block_mean_calculator_unittest.cc", |
| 446 "utility/delay_estimator_unittest.cc", |
| 447 "vad/gmm_unittest.cc", |
| 448 "vad/pitch_based_vad_unittest.cc", |
| 449 "vad/pitch_internal_unittest.cc", |
| 450 "vad/pole_zero_filter_unittest.cc", |
| 451 "vad/standalone_vad_unittest.cc", |
| 452 "vad/vad_audio_proc_unittest.cc", |
| 453 "vad/vad_circular_buffer_unittest.cc", |
| 454 "vad/voice_activity_detector_unittest.cc", |
| 455 ] |
| 456 |
| 457 deps = [ |
| 458 ":audio_processing", |
| 459 ":audioproc_debug_proto", |
| 460 ":audioproc_test_utils", |
| 461 "../..:webrtc_common", |
| 462 "../../base:gtest_prod", |
| 463 "../../base:rtc_base", |
| 464 "../../base:rtc_base_approved", |
| 465 "../../common_audio:common_audio", |
| 466 "../../system_wrappers:system_wrappers", |
| 467 "../../test:test_support", |
| 468 "../audio_coding:neteq_unittest_tools", |
| 469 "//testing/gmock", |
| 470 "//testing/gtest", |
| 471 ] |
| 472 |
| 473 defines = [] |
| 474 |
| 475 if (apm_debug_dump) { |
| 476 defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ] |
| 477 } else { |
| 478 defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ] |
| 479 } |
| 480 |
| 481 if (rtc_enable_intelligibility_enhancer) { |
| 482 defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ] |
| 483 sources += [ |
| 484 "intelligibility/intelligibility_enhancer_unittest.cc", |
| 485 "intelligibility/intelligibility_utils_unittest.cc", |
| 486 ] |
| 487 } else { |
| 488 defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ] |
| 489 } |
| 490 |
| 491 if (rtc_prefer_fixed_point) { |
| 492 defines += [ "WEBRTC_AUDIOPROC_FIXED_PROFILE" ] |
| 493 } else { |
| 494 defines += [ "WEBRTC_AUDIOPROC_FLOAT_PROFILE" ] |
| 495 } |
| 496 |
| 497 if (rtc_enable_protobuf) { |
| 498 defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ] |
| 499 deps += [ |
| 500 ":audioproc_protobuf_utils", |
| 501 ":audioproc_unittest_proto", |
| 502 ] |
| 503 sources += [ |
| 504 "aec3/block_framer_unittest.cc", |
| 505 "aec3/block_processor_unittest.cc", |
| 506 "aec3/cascaded_biquad_filter_unittest.cc", |
| 507 "aec3/echo_canceller3_unittest.cc", |
| 508 "aec3/frame_blocker_unittest.cc", |
| 509 "aec3/mock/mock_block_processor.h", |
| 510 "audio_processing_impl_locking_unittest.cc", |
| 511 "audio_processing_impl_unittest.cc", |
| 512 "audio_processing_unittest.cc", |
| 513 "echo_cancellation_bit_exact_unittest.cc", |
| 514 "echo_control_mobile_unittest.cc", |
| 515 "echo_detector/circular_buffer_unittest.cc", |
| 516 "echo_detector/mean_variance_estimator_unittest.cc", |
| 517 "echo_detector/moving_max_unittest.cc", |
| 518 "echo_detector/normalized_covariance_estimator_unittest.cc", |
| 519 "gain_control_unittest.cc", |
| 520 "level_controller/level_controller_unittest.cc", |
| 521 "level_estimator_unittest.cc", |
| 522 "low_cut_filter_unittest.cc", |
| 523 "noise_suppression_unittest.cc", |
| 524 "residual_echo_detector_unittest.cc", |
| 525 "rms_level_unittest.cc", |
| 526 "test/bitexactness_tools.cc", |
| 527 "test/bitexactness_tools.h", |
| 528 "test/debug_dump_replayer.cc", |
| 529 "test/debug_dump_replayer.h", |
| 530 "test/debug_dump_test.cc", |
| 531 "test/test_utils.h", |
| 532 "voice_detection_unittest.cc", |
| 533 ] |
| 534 } |
| 535 |
| 536 if ((!build_with_chromium || is_win) && is_clang) { |
| 537 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 538 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 539 } |
| 540 } |
| 541 |
418 rtc_source_set("audio_processing_perf_tests") { | 542 rtc_source_set("audio_processing_perf_tests") { |
419 # Has problems with autogenerated targets on Android and iOS | 543 # Has problems with autogenerated targets on Android and iOS |
420 # Dependency chain (there may also be others): | 544 # Dependency chain (there may also be others): |
421 # //webrtc/modules/audio_processing:audio_processing_perf_tests --> | 545 # //webrtc/modules/audio_processing:audio_processing_perf_tests --> |
422 # //webrtc/modules:modules_unittests --[private]--> | 546 # //webrtc/modules:modules_unittests --[private]--> |
423 # //webrtc/modules:modules_unittests_apk --> | 547 # //webrtc/modules:modules_unittests_apk --> |
424 # //webrtc/modules:modules_unittests_apk__create --> | 548 # //webrtc/modules:modules_unittests_apk__create --> |
425 # //webrtc/modules:modules_unittests_apk__create__finalize --> | 549 # //webrtc/modules:modules_unittests_apk__create__finalize --> |
426 # //webrtc/modules:modules_unittests_apk__create__package --[private]--> | 550 # //webrtc/modules:modules_unittests_apk__create__package --[private]--> |
427 # //webrtc/modules:_modules_unittests__library | 551 # //webrtc/modules:_modules_unittests__library |
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594 ] | 718 ] |
595 | 719 |
596 deps = [ | 720 deps = [ |
597 ":audioproc_debug_proto", | 721 ":audioproc_debug_proto", |
598 "../..:webrtc_common", | 722 "../..:webrtc_common", |
599 "../../base:rtc_base_approved", | 723 "../../base:rtc_base_approved", |
600 ] | 724 ] |
601 } | 725 } |
602 } | 726 } |
603 } | 727 } |
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