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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 <cmath>
#include <limits>
#include <memory>
#include <vector>
#include "api/array_view.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "modules/audio_coding/neteq/tools/audio_checksum.h"
#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/random.h"
#include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc {
namespace test {
namespace {
// Generate a mixture of sine wave and gaussian noise.
class SineAndNoiseGenerator : public EncodeNetEqInput::Generator {
public:
// The noise generator is seeded with a value from the fuzzer data, but 0 is
// avoided (since it is not allowed by the Random class).
SineAndNoiseGenerator(int sample_rate_hz, FuzzDataHelper* fuzz_data)
: sample_rate_hz_(sample_rate_hz),
fuzz_data_(*fuzz_data),
noise_generator_(fuzz_data_.ReadOrDefaultValueNotZero<uint64_t>(1)) {}
// Generates num_samples of the sine-gaussian mixture.
rtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
if (samples_.size() < num_samples) {
samples_.resize(num_samples);
}
rtc::ArrayView<int16_t> output(samples_.data(), num_samples);
// Randomize an amplitude between 0 and 32768; use 65000/2 if we are out of
// fuzzer data.
const float amplitude = fuzz_data_.ReadOrDefaultValue<uint16_t>(65000) / 2;
// Randomize a noise standard deviation between 0 and 1999.
const float noise_std = fuzz_data_.ReadOrDefaultValue<uint16_t>(0) % 2000;
for (auto& x : output) {
x = rtc::saturated_cast<int16_t>(amplitude * std::sin(phase_) +
noise_generator_.Gaussian(0, noise_std));
phase_ += 2 * kPi * kFreqHz / sample_rate_hz_;
}
return output;
}
private:
static constexpr int kFreqHz = 300; // The sinewave frequency.
const int sample_rate_hz_;
const double kPi = std::acos(-1);
std::vector<int16_t> samples_;
double phase_ = 0.0;
FuzzDataHelper& fuzz_data_;
Random noise_generator_;
};
class FuzzSignalInput : public NetEqInput {
public:
explicit FuzzSignalInput(FuzzDataHelper* fuzz_data,
int sample_rate,
uint8_t payload_type)
: fuzz_data_(*fuzz_data) {
AudioEncoderPcm16B::Config config;
config.payload_type = payload_type;
config.sample_rate_hz = sample_rate;
std::unique_ptr<AudioEncoder> encoder(new AudioEncoderPcm16B(config));
std::unique_ptr<EncodeNetEqInput::Generator> generator(
new SineAndNoiseGenerator(config.sample_rate_hz, fuzz_data));
input_.reset(new EncodeNetEqInput(std::move(generator), std::move(encoder),
std::numeric_limits<int64_t>::max()));
packet_ = input_->PopPacket();
// Select an output event period. This is how long time we wait between each
// call to NetEq::GetAudio. 10 ms is nominal, 9 and 11 ms will both lead to
// clock drift (in different directions).
constexpr int output_event_periods[] = {9, 10, 11};
output_event_period_ms_ = fuzz_data_.SelectOneOf(output_event_periods);
}
absl::optional<int64_t> NextPacketTime() const override {
return packet_->time_ms;
}
absl::optional<int64_t> NextOutputEventTime() const override {
return next_output_event_ms_;
}
absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
return input_->NextSetMinimumDelayInfo();
}
std::unique_ptr<PacketData> PopPacket() override {
RTC_DCHECK(packet_);
std::unique_ptr<PacketData> packet_to_return = std::move(packet_);
do {
packet_ = input_->PopPacket();
// If the next value from the fuzzer input is 0, the packet is discarded
// and the next one is pulled from the source.
} while (fuzz_data_.CanReadBytes(1) && fuzz_data_.Read<uint8_t>() == 0);
if (fuzz_data_.CanReadBytes(1)) {
// Generate jitter by setting an offset for the arrival time.
const int8_t arrival_time_offset_ms = fuzz_data_.Read<int8_t>();
// The arrival time can not be before the previous packets.
packet_->time_ms = std::max(packet_to_return->time_ms,
packet_->time_ms + arrival_time_offset_ms);
} else {
// Mark that we are at the end of the test. However, the current packet is
// still valid (but it may not have been fuzzed as expected).
ended_ = true;
}
return packet_to_return;
}
void AdvanceOutputEvent() override {
next_output_event_ms_ += output_event_period_ms_;
}
void AdvanceSetMinimumDelay() override {
return input_->AdvanceSetMinimumDelay();
}
bool ended() const override { return ended_; }
absl::optional<RTPHeader> NextHeader() const override {
RTC_DCHECK(packet_);
return packet_->header;
}
private:
bool ended_ = false;
FuzzDataHelper& fuzz_data_;
std::unique_ptr<EncodeNetEqInput> input_;
std::unique_ptr<PacketData> packet_;
int64_t next_output_event_ms_ = 0;
int64_t output_event_period_ms_ = 10;
};
template <class T>
bool MapHas(const std::map<int, T>& m, int key, const T& value) {
const auto it = m.find(key);
return (it != m.end() && it->second == value);
}
} // namespace
void FuzzOneInputTest(const uint8_t* data, size_t size) {
if (size < 1 || size > 65000) {
return;
}
FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
// Allowed sample rates and payload types used in the test.
std::pair<int, uint8_t> rate_types[] = {
{8000, 93}, {16000, 94}, {32000, 95}, {48000, 96}};
const auto rate_type = fuzz_data.SelectOneOf(rate_types);
const int sample_rate = rate_type.first;
const uint8_t payload_type = rate_type.second;
// Set up the input signal generator.
std::unique_ptr<FuzzSignalInput> input(
new FuzzSignalInput(&fuzz_data, sample_rate, payload_type));
// Output sink for the test.
std::unique_ptr<AudioChecksum> output(new AudioChecksum);
// Configure NetEq and the NetEqTest object.
NetEqTest::Callbacks callbacks;
NetEq::Config config;
config.enable_fast_accelerate = true;
auto codecs = NetEqTest::StandardDecoderMap();
// rate_types contains the payload types that will be used for encoding.
// Verify that they all are included in the standard decoder map, and that
// they point to the expected decoder types.
RTC_CHECK(
MapHas(codecs, rate_types[0].second, SdpAudioFormat("l16", 8000, 1)));
RTC_CHECK(
MapHas(codecs, rate_types[1].second, SdpAudioFormat("l16", 16000, 1)));
RTC_CHECK(
MapHas(codecs, rate_types[2].second, SdpAudioFormat("l16", 32000, 1)));
RTC_CHECK(
MapHas(codecs, rate_types[3].second, SdpAudioFormat("l16", 48000, 1)));
NetEqTest test(config, CreateBuiltinAudioDecoderFactory(), codecs,
/*text_log=*/nullptr, /*neteq_factory=*/nullptr,
std::move(input), std::move(output), callbacks);
test.Run();
}
} // namespace test
void FuzzOneInput(const uint8_t* data, size_t size) {
test::FuzzOneInputTest(data, size);
}
} // namespace webrtc
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