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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc')
-rw-r--r-- | third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc | 145 |
1 files changed, 145 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc new file mode 100644 index 0000000000..331a373f4e --- /dev/null +++ b/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc @@ -0,0 +1,145 @@ +/* + * 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 "api/audio/echo_detector_creator.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-Aec3MinErleDuringOnsetsKillSwitch", + "WebRTC-Aec3ShortHeadroomKillSwitch", +}; + +rtc::scoped_refptr<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 use_ts = fuzz_data->ReadOrDefaultValue(true); + bool use_red = fuzz_data->ReadOrDefaultValue(true); + bool use_hpf = fuzz_data->ReadOrDefaultValue(true); + bool use_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_agc_limiter = fuzz_data->ReadOrDefaultValue(true); + bool use_agc2 = fuzz_data->ReadOrDefaultValue(true); + bool use_agc2_adaptive_digital = fuzz_data->ReadOrDefaultValue(true); + + // Read a gain value supported by GainController2::Validate(). + const float gain_controller2_gain_db = + fuzz_data->ReadOrDefaultValue<uint8_t>(0) % 50; + + 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()); + + // 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 && use_aec3 && use_ns) // These settings trigger webrtc:9489. + ) { + return nullptr; + } + + std::unique_ptr<EchoControlFactory> echo_control_factory; + if (use_aec3) { + echo_control_factory.reset(new EchoCanceller3Factory()); + } + + 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.high_pass_filter.enabled = use_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.noise_suppression.enabled = use_ns; + apm_config.transient_suppression.enabled = use_ts; + + rtc::scoped_refptr<AudioProcessing> apm = + AudioProcessingBuilderForTesting() + .SetEchoControlFactory(std::move(echo_control_factory)) + .SetEchoDetector(use_red ? CreateEchoDetector() : nullptr) + .SetConfig(apm_config) + .Create(); + +#ifdef WEBRTC_LINUX + apm->AttachAecDump(AecDumpFactory::Create("/dev/null", -1, worker_queue)); +#endif + + 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<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 = ""; + + 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 |