| /* | 
 |  *  Copyright (c) 2018 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 "api/array_view.h" | 
 | #include "api/audio_codecs/opus/audio_encoder_opus.h" | 
 | #include "rtc_base/buffer.h" | 
 | #include "rtc_base/checks.h" | 
 | #include "test/fuzzers/fuzz_data_helper.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace { | 
 |  | 
 | // 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, | 
 |                       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. | 
 |     return; | 
 |   } | 
 |  | 
 |   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(input_aligned.data(), byte_array.data(), byte_array.size()); | 
 |     auto info = encoder->Encode(timestamp, input_aligned, &encoded); | 
 |   } | 
 | } | 
 |  | 
 | }  // namespace | 
 |  | 
 | void FuzzOneInput(const uint8_t* data, size_t size) { | 
 |   AudioEncoderOpus::Config config; | 
 |   config.frame_size_ms = 20; | 
 |   RTC_CHECK(config.IsOk()); | 
 |   constexpr int kPayloadType = 100; | 
 |   std::unique_ptr<AudioEncoder> enc = | 
 |       AudioEncoderOpus::MakeAudioEncoder(config, kPayloadType); | 
 |   FuzzAudioEncoder(rtc::ArrayView<const uint8_t>(data, size), enc.get()); | 
 | } | 
 |  | 
 | }  // namespace webrtc |