Bug: webrtc:9413 Change-Id: Iccf861453c1c49c306ad18542074a792592491a9 Reviewed-on: https://webrtc-review.googlesource.com/c/113501 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25928}
202 lines
8.3 KiB
C++
202 lines
8.3 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 <bitset>
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#include <string>
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#include "absl/memory/memory.h"
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#include "api/audio/echo_canceller3_factory.h"
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#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/arraysize.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/task_queue.h"
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#include "system_wrappers/include/field_trial.h"
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#include "test/fuzzers/audio_processing_fuzzer_helper.h"
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#include "test/fuzzers/fuzz_data_helper.h"
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namespace webrtc {
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namespace {
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const std::string kFieldTrialNames[] = {
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"WebRTC-Aec3AdaptErleOnLowRenderKillSwitch",
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"WebRTC-Aec3AgcGainChangeResponseKillSwitch",
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"WebRTC-Aec3BoundedNearendKillSwitch",
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"WebRTC-Aec3EarlyShadowFilterJumpstartKillSwitch",
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"WebRTC-Aec3EnableAdaptiveEchoReverbEstimation",
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"WebRTC-Aec3EnableLegacyDominantNearend",
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"WebRTC-Aec3EnableUnityInitialRampupGain",
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"WebRTC-Aec3EnableUnityNonZeroRampupGain",
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"WebRTC-Aec3EnforceSkewHysteresis1",
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"WebRTC-Aec3EnforceSkewHysteresis2",
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"WebRTC-Aec3FilterAnalyzerPreprocessorKillSwitch",
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"WebRTC-Aec3MisadjustmentEstimatorKillSwitch",
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"WebRTC-Aec3NewFilterParamsKillSwitch",
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"WebRTC-Aec3NewRenderBufferingKillSwitch",
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"WebRTC-Aec3OverrideEchoPathGainKillSwitch",
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"WebRTC-Aec3RapidAgcGainRecoveryKillSwitch",
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"WebRTC-Aec3ResetErleAtGainChangesKillSwitch",
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"WebRTC-Aec3ReverbBasedOnRenderKillSwitch",
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"WebRTC-Aec3ReverbModellingKillSwitch",
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"WebRTC-Aec3ShadowFilterBoostedJumpstartKillSwitch",
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"WebRTC-Aec3ShadowFilterJumpstartKillSwitch",
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"WebRTC-Aec3ShortReverbKillSwitch",
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"WebRTC-Aec3SmoothSignalTransitionsKillSwitch",
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"WebRTC-Aec3SmoothUpdatesTailFreqRespKillSwitch",
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"WebRTC-Aec3SoftTransparentModeKillSwitch",
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"WebRTC-Aec3StandardNonlinearReverbModelKillSwitch",
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"WebRTC-Aec3StrictDivergenceCheckKillSwitch",
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"WebRTC-Aec3UseLegacyNormalSuppressorTuning",
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"WebRTC-Aec3UseOffsetBlocks",
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"WebRTC-Aec3UseShortDelayEstimatorWindow",
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"WebRTC-Aec3UseStationarityPropertiesKillSwitch",
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"WebRTC-Aec3UtilizeShadowFilterOutputKillSwitch",
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"WebRTC-Aec3ZeroExternalDelayHeadroomKillSwitch",
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"WebRTC-Aec3EarlyDelayDetectionKillSwitch",
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"WebRTC-Aec3FilterQualityStateKillSwitch",
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"WebRTC-Aec3NewSaturationBehaviorKillSwitch",
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"WebRTC-Aec3GainLimiterDeactivationKillSwitch",
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"WebRTC-Aec3EnableErleUpdatesDuringReverbKillSwitch",
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};
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std::unique_ptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
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std::string* field_trial_string,
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rtc::TaskQueue* worker_queue) {
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// Parse boolean values for optionally enabling different
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// configurable public components of APM.
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bool exp_agc = fuzz_data->ReadOrDefaultValue(true);
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bool exp_ns = fuzz_data->ReadOrDefaultValue(true);
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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bool ef = fuzz_data->ReadOrDefaultValue(true);
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bool raf = fuzz_data->ReadOrDefaultValue(true);
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
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bool red = fuzz_data->ReadOrDefaultValue(true);
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bool hpf = fuzz_data->ReadOrDefaultValue(true);
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bool aec3 = fuzz_data->ReadOrDefaultValue(true);
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bool use_aec = fuzz_data->ReadOrDefaultValue(true);
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bool use_aecm = fuzz_data->ReadOrDefaultValue(true);
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bool use_agc = fuzz_data->ReadOrDefaultValue(true);
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bool use_ns = fuzz_data->ReadOrDefaultValue(true);
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bool use_le = fuzz_data->ReadOrDefaultValue(true);
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bool use_vad = fuzz_data->ReadOrDefaultValue(true);
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bool use_agc_limiter = fuzz_data->ReadOrDefaultValue(true);
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bool use_agc2 = fuzz_data->ReadOrDefaultValue(true);
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// Read an int8 value, but don't let it be too large or small.
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const float gain_controller2_gain_db =
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rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(0), -40, 40);
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constexpr size_t kNumFieldTrials = arraysize(kFieldTrialNames);
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// Verify that the read data type has enough bits to fuzz the field trials.
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using FieldTrialBitmaskType = uint64_t;
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static_assert(kNumFieldTrials <= sizeof(FieldTrialBitmaskType) * 8,
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"FieldTrialBitmaskType is not large enough.");
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std::bitset<kNumFieldTrials> field_trial_bitmask(
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fuzz_data->ReadOrDefaultValue<FieldTrialBitmaskType>(0));
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for (size_t i = 0; i < kNumFieldTrials; ++i) {
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if (field_trial_bitmask[i]) {
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*field_trial_string += kFieldTrialNames[i] + "/Enabled/";
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}
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}
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field_trial::InitFieldTrialsFromString(field_trial_string->c_str());
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bool use_agc2_adaptive_digital = fuzz_data->ReadOrDefaultValue(true);
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bool use_agc2_adaptive_digital_rms_estimator =
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fuzz_data->ReadOrDefaultValue(true);
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bool use_agc2_adaptive_digital_saturation_protector =
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fuzz_data->ReadOrDefaultValue(true);
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// Ignore a few bytes. Bytes from this segment will be used for
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// future config flag changes. We assume 40 bytes is enough for
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// configuring the APM.
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constexpr size_t kSizeOfConfigSegment = 40;
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RTC_DCHECK(kSizeOfConfigSegment >= fuzz_data->BytesRead());
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static_cast<void>(
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fuzz_data->ReadByteArray(kSizeOfConfigSegment - fuzz_data->BytesRead()));
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// Filter out incompatible settings that lead to CHECK failures.
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if ((use_aecm && use_aec) || // These settings cause CHECK failure.
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(use_aecm && aec3 && use_ns) // These settings trigger webrtc:9489.
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) {
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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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std::unique_ptr<EchoControlFactory> echo_control_factory;
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if (aec3) {
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echo_control_factory.reset(new EchoCanceller3Factory());
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}
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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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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(true));
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std::unique_ptr<AudioProcessing> apm(
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AudioProcessingBuilder()
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.SetEchoControlFactory(std::move(echo_control_factory))
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.Create(config));
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#ifdef WEBRTC_LINUX
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apm->AttachAecDump(AecDumpFactory::Create("/dev/null", -1, worker_queue));
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#endif
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webrtc::AudioProcessing::Config apm_config;
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apm_config.echo_canceller.enabled = use_aec || use_aecm;
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apm_config.echo_canceller.mobile_mode = use_aecm;
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apm_config.residual_echo_detector.enabled = red;
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apm_config.high_pass_filter.enabled = hpf;
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apm_config.gain_controller2.enabled = use_agc2;
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apm_config.gain_controller2.fixed_digital.gain_db = gain_controller2_gain_db;
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apm_config.gain_controller2.adaptive_digital.enabled =
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use_agc2_adaptive_digital;
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apm_config.gain_controller2.adaptive_digital.level_estimator =
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use_agc2_adaptive_digital_rms_estimator
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? webrtc::AudioProcessing::Config::GainController2::LevelEstimator::
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kRms
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: webrtc::AudioProcessing::Config::GainController2::LevelEstimator::
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kPeak;
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apm_config.gain_controller2.adaptive_digital.use_saturation_protector =
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use_agc2_adaptive_digital_saturation_protector;
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apm->ApplyConfig(apm_config);
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apm->gain_control()->Enable(use_agc);
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apm->noise_suppression()->Enable(use_ns);
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apm->level_estimator()->Enable(use_le);
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apm->voice_detection()->Enable(use_vad);
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apm->gain_control()->enable_limiter(use_agc_limiter);
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return apm;
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}
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} // namespace
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void FuzzOneInput(const uint8_t* data, size_t size) {
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test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
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// This string must be in scope during execution, according to documentation
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// for field_trial.h. Hence it's created here and not in CreateApm.
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std::string field_trial_string = "";
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std::unique_ptr<rtc::TaskQueue> worker_queue(
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new rtc::TaskQueue("rtc-low-prio", rtc::TaskQueue::Priority::LOW));
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auto apm = CreateApm(&fuzz_data, &field_trial_string, worker_queue.get());
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if (apm) {
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FuzzAudioProcessing(&fuzz_data, std::move(apm));
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}
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}
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} // namespace webrtc
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