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/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <bitset>
#include <string>
#include "absl/memory/memory.h"
#include "api/audio/echo_canceller3_factory.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/task_queue.h"
#include "system_wrappers/include/field_trial.h"
#include "test/fuzzers/audio_processing_fuzzer_helper.h"
#include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc {
namespace {
const std::string kFieldTrialNames[] = {
"WebRTC-Audio-Agc2ForceExtraSaturationMargin",
"WebRTC-Audio-Agc2ForceInitialSaturationMargin",
"WebRTC-Aec3MinErleDuringOnsetsKillSwitch",
"WebRTC-Aec3ShortHeadroomKillSwitch",
};
std::unique_ptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
std::string* field_trial_string,
rtc::TaskQueue* worker_queue) {
// Parse boolean values for optionally enabling different
// configurable public components of APM.
bool exp_agc = fuzz_data->ReadOrDefaultValue(true);
bool exp_ns = fuzz_data->ReadOrDefaultValue(true);
static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
bool ef = fuzz_data->ReadOrDefaultValue(true);
bool raf = fuzz_data->ReadOrDefaultValue(true);
static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
bool red = fuzz_data->ReadOrDefaultValue(true);
bool hpf = fuzz_data->ReadOrDefaultValue(true);
bool aec3 = fuzz_data->ReadOrDefaultValue(true);
bool use_aec = fuzz_data->ReadOrDefaultValue(true);
bool use_aecm = fuzz_data->ReadOrDefaultValue(true);
bool use_agc = fuzz_data->ReadOrDefaultValue(true);
bool use_ns = fuzz_data->ReadOrDefaultValue(true);
bool use_le = fuzz_data->ReadOrDefaultValue(true);
bool use_vad = fuzz_data->ReadOrDefaultValue(true);
bool use_agc_limiter = fuzz_data->ReadOrDefaultValue(true);
bool use_agc2 = fuzz_data->ReadOrDefaultValue(true);
// Read an int8 value, but don't let it be too large or small.
const float gain_controller2_gain_db =
rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(0), -40, 40);
constexpr size_t kNumFieldTrials = arraysize(kFieldTrialNames);
// Verify that the read data type has enough bits to fuzz the field trials.
using FieldTrialBitmaskType = uint64_t;
static_assert(kNumFieldTrials <= sizeof(FieldTrialBitmaskType) * 8,
"FieldTrialBitmaskType is not large enough.");
std::bitset<kNumFieldTrials> field_trial_bitmask(
fuzz_data->ReadOrDefaultValue<FieldTrialBitmaskType>(0));
for (size_t i = 0; i < kNumFieldTrials; ++i) {
if (field_trial_bitmask[i]) {
*field_trial_string += kFieldTrialNames[i] + "/Enabled/";
}
}
field_trial::InitFieldTrialsFromString(field_trial_string->c_str());
bool use_agc2_adaptive_digital = fuzz_data->ReadOrDefaultValue(true);
bool use_agc2_adaptive_digital_rms_estimator =
fuzz_data->ReadOrDefaultValue(true);
bool use_agc2_adaptive_digital_saturation_protector =
fuzz_data->ReadOrDefaultValue(true);
// Ignore a few bytes. Bytes from this segment will be used for
// future config flag changes. We assume 40 bytes is enough for
// configuring the APM.
constexpr size_t kSizeOfConfigSegment = 40;
RTC_DCHECK(kSizeOfConfigSegment >= fuzz_data->BytesRead());
static_cast<void>(
fuzz_data->ReadByteArray(kSizeOfConfigSegment - fuzz_data->BytesRead()));
// Filter out incompatible settings that lead to CHECK failures.
if ((use_aecm && use_aec) || // These settings cause CHECK failure.
(use_aecm && aec3 && use_ns) // These settings trigger webrtc:9489.
) {
return nullptr;
}
// Components can be enabled through webrtc::Config and
// webrtc::AudioProcessingConfig.
Config config;
std::unique_ptr<EchoControlFactory> echo_control_factory;
if (aec3) {
echo_control_factory.reset(new EchoCanceller3Factory());
}
config.Set<ExperimentalAgc>(new ExperimentalAgc(exp_agc));
config.Set<ExperimentalNs>(new ExperimentalNs(exp_ns));
config.Set<ExtendedFilter>(new ExtendedFilter(ef));
config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(raf));
config.Set<DelayAgnostic>(new DelayAgnostic(true));
std::unique_ptr<AudioProcessing> apm(
AudioProcessingBuilder()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create(config));
#ifdef WEBRTC_LINUX
apm->AttachAecDump(AecDumpFactory::Create("/dev/null", -1, worker_queue));
#endif
webrtc::AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = use_aec || use_aecm;
apm_config.echo_canceller.mobile_mode = use_aecm;
apm_config.residual_echo_detector.enabled = red;
apm_config.high_pass_filter.enabled = hpf;
apm_config.gain_controller1.enabled = use_agc;
apm_config.gain_controller1.enable_limiter = use_agc_limiter;
apm_config.gain_controller2.enabled = use_agc2;
apm_config.gain_controller2.fixed_digital.gain_db = gain_controller2_gain_db;
apm_config.gain_controller2.adaptive_digital.enabled =
use_agc2_adaptive_digital;
apm_config.gain_controller2.adaptive_digital.level_estimator =
use_agc2_adaptive_digital_rms_estimator
? webrtc::AudioProcessing::Config::GainController2::LevelEstimator::
kRms
: webrtc::AudioProcessing::Config::GainController2::LevelEstimator::
kPeak;
apm_config.gain_controller2.adaptive_digital.use_saturation_protector =
use_agc2_adaptive_digital_saturation_protector;
apm_config.noise_suppression.enabled = use_ns;
apm_config.voice_detection.enabled = use_vad;
apm->ApplyConfig(apm_config);
apm->level_estimator()->Enable(use_le);
apm->voice_detection()->Enable(use_vad);
return apm;
}
} // namespace
void FuzzOneInput(const uint8_t* data, size_t size) {
if (size > 400000) {
return;
}
test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
// This string must be in scope during execution, according to documentation
// for field_trial.h. Hence it's created here and not in CreateApm.
std::string field_trial_string = "";
std::unique_ptr<rtc::TaskQueue> worker_queue(
new rtc::TaskQueue("rtc-low-prio", rtc::TaskQueue::Priority::LOW));
auto apm = CreateApm(&fuzz_data, &field_trial_string, worker_queue.get());
if (apm) {
FuzzAudioProcessing(&fuzz_data, std::move(apm));
}
}
} // namespace webrtc
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