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/*
* Copyright (c) 2019 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 "absl/types/optional.h"
#include "modules/audio_processing/aec3/echo_canceller3.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc {
namespace {
using SampleRate = ::webrtc::AudioProcessing::NativeRate;
void PrepareAudioBuffer(int sample_rate_hz,
test::FuzzDataHelper* fuzz_data,
AudioBuffer* buffer) {
float* const* channels = buffer->channels_f();
for (size_t i = 0; i < buffer->num_channels(); ++i) {
for (size_t j = 0; j < buffer->num_frames(); ++j) {
channels[i][j] =
static_cast<float>(fuzz_data->ReadOrDefaultValue<int16_t>(0));
}
}
if (sample_rate_hz == 32000 || sample_rate_hz == 48000) {
buffer->SplitIntoFrequencyBands();
}
}
} // namespace
void FuzzOneInput(const uint8_t* data, size_t size) {
if (size > 200000) {
return;
}
test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
constexpr int kSampleRates[] = {16000, 32000, 48000};
const int sample_rate_hz =
static_cast<size_t>(fuzz_data.SelectOneOf(kSampleRates));
constexpr int kMaxNumChannels = 9;
const size_t num_render_channels =
1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
const size_t num_capture_channels =
1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
EchoCanceller3 aec3(EchoCanceller3Config(),
/*multichannel_config=*/absl::nullopt, sample_rate_hz,
num_render_channels, num_capture_channels);
AudioBuffer capture_audio(sample_rate_hz, num_capture_channels,
sample_rate_hz, num_capture_channels,
sample_rate_hz, num_capture_channels);
AudioBuffer render_audio(sample_rate_hz, num_render_channels, sample_rate_hz,
num_render_channels, sample_rate_hz,
num_render_channels);
// Fuzz frames while there is still fuzzer data.
while (fuzz_data.BytesLeft() > 0) {
bool is_capture = fuzz_data.ReadOrDefaultValue(true);
bool level_changed = fuzz_data.ReadOrDefaultValue(true);
if (is_capture) {
PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &capture_audio);
aec3.ProcessCapture(&capture_audio, level_changed);
} else {
PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &render_audio);
aec3.AnalyzeRender(&render_audio);
}
}
}
} // namespace webrtc
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