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/*
* 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 "test/fuzzers/audio_processing_fuzzer_helper.h"
#include <algorithm>
#include <array>
#include <cmath>
#include <limits>
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/include/audio_frame_proxies.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
bool ValidForApm(float x) {
return std::isfinite(x) && -1.0f <= x && x <= 1.0f;
}
void GenerateFloatFrame(test::FuzzDataHelper* fuzz_data,
int input_rate,
int num_channels,
float* const* float_frames) {
const int samples_per_input_channel =
AudioProcessing::GetFrameSize(input_rate);
RTC_DCHECK_LE(samples_per_input_channel, 480);
for (int i = 0; i < num_channels; ++i) {
std::fill(float_frames[i], float_frames[i] + samples_per_input_channel, 0);
const size_t read_bytes = sizeof(float) * samples_per_input_channel;
if (fuzz_data->CanReadBytes(read_bytes)) {
rtc::ArrayView<const uint8_t> byte_array =
fuzz_data->ReadByteArray(read_bytes);
memmove(float_frames[i], byte_array.begin(), read_bytes);
}
// Sanitize input.
for (int j = 0; j < samples_per_input_channel; ++j) {
if (!ValidForApm(float_frames[i][j])) {
float_frames[i][j] = 0.f;
}
}
}
}
void GenerateFixedFrame(test::FuzzDataHelper* fuzz_data,
int input_rate,
int num_channels,
AudioFrame* fixed_frame) {
const int samples_per_input_channel =
AudioProcessing::GetFrameSize(input_rate);
fixed_frame->samples_per_channel_ = samples_per_input_channel;
fixed_frame->sample_rate_hz_ = input_rate;
fixed_frame->num_channels_ = num_channels;
RTC_DCHECK_LE(samples_per_input_channel * num_channels,
AudioFrame::kMaxDataSizeSamples);
for (int i = 0; i < samples_per_input_channel * num_channels; ++i) {
fixed_frame->mutable_data()[i] = fuzz_data->ReadOrDefaultValue<int16_t>(0);
}
}
} // namespace
void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
rtc::scoped_refptr<AudioProcessing> apm) {
AudioFrame fixed_frame;
// Normal usage is up to 8 channels. Allowing to fuzz one beyond this allows
// us to catch implicit assumptions about normal usage.
constexpr int kMaxNumChannels = 9;
std::array<std::array<float, 480>, kMaxNumChannels> float_frames;
std::array<float*, kMaxNumChannels> float_frame_ptrs;
for (int i = 0; i < kMaxNumChannels; ++i) {
float_frame_ptrs[i] = float_frames[i].data();
}
float* const* ptr_to_float_frames = &float_frame_ptrs[0];
constexpr int kSampleRatesHz[] = {8000, 11025, 16000, 22050,
32000, 44100, 48000};
// We may run out of fuzz data in the middle of a loop iteration. In
// that case, default values will be used for the rest of that
// iteration.
while (fuzz_data->CanReadBytes(1)) {
const bool is_float = fuzz_data->ReadOrDefaultValue(true);
// Decide input/output rate for this iteration.
const int input_rate = fuzz_data->SelectOneOf(kSampleRatesHz);
const int output_rate = fuzz_data->SelectOneOf(kSampleRatesHz);
const uint8_t stream_delay = fuzz_data->ReadOrDefaultValue<uint8_t>(0);
// API call needed for AECM to run.
apm->set_stream_delay_ms(stream_delay);
const bool key_pressed = fuzz_data->ReadOrDefaultValue(true);
apm->set_stream_key_pressed(key_pressed);
// Make the APM call depending on capture/render mode and float /
// fix interface.
const bool is_capture = fuzz_data->ReadOrDefaultValue(true);
// Fill the arrays with audio samples from the data.
int apm_return_code = AudioProcessing::Error::kNoError;
if (is_float) {
const int num_channels =
fuzz_data->ReadOrDefaultValue<uint8_t>(1) % kMaxNumChannels;
GenerateFloatFrame(fuzz_data, input_rate, num_channels,
ptr_to_float_frames);
if (is_capture) {
apm_return_code = apm->ProcessStream(
ptr_to_float_frames, StreamConfig(input_rate, num_channels),
StreamConfig(output_rate, num_channels), ptr_to_float_frames);
} else {
apm_return_code = apm->ProcessReverseStream(
ptr_to_float_frames, StreamConfig(input_rate, num_channels),
StreamConfig(output_rate, num_channels), ptr_to_float_frames);
}
} else {
const int num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
GenerateFixedFrame(fuzz_data, input_rate, num_channels, &fixed_frame);
if (is_capture) {
apm_return_code = ProcessAudioFrame(apm.get(), &fixed_frame);
} else {
apm_return_code = ProcessReverseAudioFrame(apm.get(), &fixed_frame);
}
}
// Cover stats gathering code paths.
static_cast<void>(apm->GetStatistics(true /*has_remote_tracks*/));
RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
}
}
} // namespace webrtc
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