blob: 837c26df564561d20a0e96acafd55f5be51ea109 [file] [log] [blame]
* Copyright (c) 2020 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_encoder_fuzzer.h"
#include <cstring>
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc {
// This function reads bytes from `data_view`, interprets them as RTP timestamp
// and input samples, and sends them for encoding. The process continues until
// no more data is available.
void FuzzAudioEncoder(rtc::ArrayView<const uint8_t> data_view,
std::unique_ptr<AudioEncoder> encoder) {
test::FuzzDataHelper data(data_view);
const size_t block_size_samples =
encoder->SampleRateHz() / 100 * encoder->NumChannels();
const size_t block_size_bytes = block_size_samples * sizeof(int16_t);
if (data_view.size() / block_size_bytes > 1000) {
// If the size of the fuzzer data is more than 1000 input blocks (i.e., more
// than 10 seconds), then don't fuzz at all for the fear of timing out.
rtc::BufferT<int16_t> input_aligned(block_size_samples);
rtc::Buffer encoded;
// Each round in the loop below will need one block of samples + a 32-bit
// timestamp from the fuzzer input.
const size_t bytes_to_read = block_size_bytes + sizeof(uint32_t);
while (data.CanReadBytes(bytes_to_read)) {
const uint32_t timestamp = data.Read<uint32_t>();
auto byte_array = data.ReadByteArray(block_size_bytes);
// Align the data by copying to another array.
RTC_DCHECK_EQ(input_aligned.size() * sizeof(int16_t),
byte_array.size() * sizeof(uint8_t));
memcpy(,, byte_array.size());
auto info = encoder->Encode(timestamp, input_aligned, &encoded);
} // namespace webrtc