APM has historically allowed sample rates not divisible by 100, but there is also code that explicitly states that such rates are not supported. It is unclear how well rates like 22050 are handled in practice. This CL adds support for fuzzing more sample rates, to help find issues. We usually preserve fuzzer data reads to avoid invalidating unresolved fuzzer-found issues, but to make the code a little more readable this CL removes the discarded reads. This renders the only currently open bug non-reproducible, crbug.com/1299393. Bug: webrtc:9413, chromium:1299393 Change-Id: I98ac1c653627c20adc73b8edede02f1526d80d9d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264504 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37114}
148 lines
5.7 KiB
C++
148 lines
5.7 KiB
C++
/*
|
|
* 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-Audio-Agc2ForceExtraSaturationMargin",
|
|
"WebRTC-Audio-Agc2ForceInitialSaturationMargin",
|
|
"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
|