webrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
Sam Zackrisson f0d1c03c31 Add replacement interface for webrtc::GainConrol
The pointer-to-submodule interfaces are being removed.
This CL:
1) introduces AudioProcessing::Config::GainController1 with most config,
2) adds functions to APM for setting and getting analog gain,
3) creates a temporary GainControlConfigProxy to support the transition
   to the new config.
4) Moves the lock references in GainControlForExperimentalAgc and
   GainControlImpl into the GainControlConfigProxy, as it becomes the
   sole AGC object with functionality exposed to the client.

Bug: webrtc:9947, webrtc:9878
Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27316}
2019-03-27 15:19:41 +00:00

170 lines
6.8 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 "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