/* * 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 #include #include "absl/memory/memory.h" #include "api/audio/echo_canceller3_factory.h" #include "api/task_queue/default_task_queue_factory.h" #include "modules/audio_processing/aec_dump/aec_dump_factory.h" #include "modules/audio_processing/include/audio_processing.h" #include "modules/audio_processing/test/audio_processing_builder_for_testing.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 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(fuzz_data->ReadOrDefaultValue(true)); static_cast(fuzz_data->ReadOrDefaultValue(true)); static_cast(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(fuzz_data->ReadOrDefaultValue(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 field_trial_bitmask( fuzz_data->ReadOrDefaultValue(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( 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 echo_control_factory; if (aec3) { echo_control_factory.reset(new EchoCanceller3Factory()); } config.Set(new ExperimentalAgc(exp_agc)); config.Set(new ExperimentalNs(exp_ns)); std::unique_ptr apm( AudioProcessingBuilderForTesting() .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.pipeline.multi_channel_render = true; apm_config.pipeline.multi_channel_capture = true; 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_config.level_estimation.enabled = use_le; apm->ApplyConfig(apm_config); return apm; } TaskQueueFactory* GetTaskQueueFactory() { static TaskQueueFactory* const factory = CreateDefaultTaskQueueFactory().release(); return factory; } } // namespace void FuzzOneInput(const uint8_t* data, size_t size) { if (size > 400000) { return; } test::FuzzDataHelper fuzz_data(rtc::ArrayView(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 = ""; rtc::TaskQueue worker_queue(GetTaskQueueFactory()->CreateTaskQueue( "rtc-low-prio", rtc::TaskQueue::Priority::LOW)); auto apm = CreateApm(&fuzz_data, &field_trial_string, &worker_queue); if (apm) { FuzzAudioProcessing(&fuzz_data, std::move(apm)); } } } // namespace webrtc