92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
71 lines
2.4 KiB
C++
71 lines
2.4 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/include/audio_processing.h"
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#include "test/fuzzers/audio_processing_fuzzer.h"
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#include "api/optional.h"
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namespace webrtc {
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std::unique_ptr<AudioProcessing> CreateAPM(const uint8_t** data,
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size_t* remaining_size) {
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// Parse boolean values for optionally enabling different
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// configurable public components of APM.
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auto exp_agc = ParseBool(data, remaining_size);
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auto exp_ns = ParseBool(data, remaining_size);
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auto bf = ParseBool(data, remaining_size);
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auto ef = ParseBool(data, remaining_size);
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auto raf = ParseBool(data, remaining_size);
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auto da = ParseBool(data, remaining_size);
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auto ie = ParseBool(data, remaining_size);
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auto red = ParseBool(data, remaining_size);
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auto lc = ParseBool(data, remaining_size);
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auto hpf = ParseBool(data, remaining_size);
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auto aec3 = ParseBool(data, remaining_size);
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if (!(exp_agc && exp_ns && bf && ef && raf && da && ie && red && lc && hpf &&
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aec3)) {
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return nullptr;
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}
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// Components can be enabled through webrtc::Config and
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// webrtc::AudioProcessingConfig.
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Config config;
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config.Set<ExperimentalAgc>(new ExperimentalAgc(*exp_agc));
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config.Set<ExperimentalNs>(new ExperimentalNs(*exp_ns));
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if (*bf) {
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config.Set<Beamforming>(new Beamforming());
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}
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config.Set<ExtendedFilter>(new ExtendedFilter(*ef));
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config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(*raf));
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config.Set<DelayAgnostic>(new DelayAgnostic(*da));
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config.Set<Intelligibility>(new Intelligibility(*ie));
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std::unique_ptr<AudioProcessing> apm(AudioProcessing::Create(config));
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webrtc::AudioProcessing::Config apm_config;
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apm_config.residual_echo_detector.enabled = *red;
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apm_config.level_controller.enabled = *lc;
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apm_config.high_pass_filter.enabled = *hpf;
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apm_config.echo_canceller3.enabled = *aec3;
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apm->ApplyConfig(apm_config);
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return apm;
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}
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void FuzzOneInput(const uint8_t* data, size_t size) {
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auto apm = CreateAPM(&data, &size);
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FuzzAudioProcessing(data, size, std::move(apm));
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}
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} // namespace webrtc
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