Files
platform-external-webrtc/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
Henrik Lundin fe32a76d60 Create fuzzer tests for audio decoders
This change adds fuzzer tests for iLBC, iSAC fix and float, and
Opus. The fuzzer function takes a random input vector and splits it
into a number of payloads. The lengths of the payloads is also
determined by the random vector. The payloads are decoded with the
decoders.

BUG=webrtc:5306
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1499093002 .

Cr-Commit-Position: refs/heads/master@{#10932}
2015-12-08 10:27:34 +00:00

50 lines
1.8 KiB
C++

/*
* Copyright (c) 2015 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 "webrtc/test/fuzzers/audio_decoder_fuzzer.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
namespace webrtc {
namespace {
size_t PacketSizeFromTwoBytes(const uint8_t* data, size_t size) {
if (size < 2)
return 0;
return static_cast<size_t>((data[0] << 8) + data[1]);
}
} // namespace
// This function reads two bytes from the beginning of |data|, interprets them
// as the first packet length, and reads this many bytes if available. The
// payload is inserted into the decoder, and the process continues until no more
// data is available.
void FuzzAudioDecoder(const uint8_t* data,
size_t size,
AudioDecoder* decoder,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded) {
const uint8_t* data_ptr = data;
size_t remaining_size = size;
size_t packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size);
while (packet_len != 0 && packet_len <= remaining_size - 2) {
data_ptr += 2;
remaining_size -= 2;
AudioDecoder::SpeechType speech_type;
decoder->Decode(data_ptr, packet_len, sample_rate_hz, max_decoded_bytes,
decoded, &speech_type);
data_ptr += packet_len;
remaining_size -= packet_len;
packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size);
}
}
} // namespace webrtc