mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-20 09:07:52 +01:00
35c773dad6
The fuzzer figured out that 3 bytes is enough to fuzz a package. 2 bytes for packet length, and 1 byte of actual packet. A 20K test case can generate > 6000 packets. It does not seem like efficient fuzzing. This CL simply stops execution when 200 packets have been generated. That corresponds to 4 seconds of 20 ms packets. Bug: chromium:840115 Change-Id: Id2742a6f8021134bacd8a6e8c71b32f20c7f1086 Reviewed-on: https://webrtc-review.googlesource.com/88566 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24000}
104 lines
3.9 KiB
C++
104 lines
3.9 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 "test/fuzzers/audio_decoder_fuzzer.h"
|
|
|
|
#include <limits>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "api/audio_codecs/audio_decoder.h"
|
|
#include "modules/rtp_rtcp/source/byte_io.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
template <typename T, unsigned int B = sizeof(T)>
|
|
bool ParseInt(const uint8_t** data, size_t* remaining_size, T* value) {
|
|
static_assert(std::numeric_limits<T>::is_integer, "Type must be an integer.");
|
|
static_assert(sizeof(T) <= sizeof(uint64_t),
|
|
"Cannot read wider than uint64_t.");
|
|
static_assert(B <= sizeof(T), "T must be at least B bytes wide.");
|
|
if (B > *remaining_size)
|
|
return false;
|
|
uint64_t val = ByteReader<uint64_t, B>::ReadBigEndian(*data);
|
|
*data += B;
|
|
*remaining_size -= B;
|
|
*value = static_cast<T>(val);
|
|
return true;
|
|
}
|
|
} // 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. Either AudioDecoder::Decode or
|
|
// AudioDecoder::DecodeRedundant is used, depending on the value of
|
|
// |decode_type|.
|
|
void FuzzAudioDecoder(DecoderFunctionType decode_type,
|
|
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;
|
|
constexpr size_t kMaxNumFuzzedPackets = 200;
|
|
for (size_t num_packets = 0; num_packets < kMaxNumFuzzedPackets;
|
|
++num_packets) {
|
|
if (!(ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) &&
|
|
packet_len <= remaining_size)) {
|
|
break;
|
|
}
|
|
AudioDecoder::SpeechType speech_type;
|
|
switch (decode_type) {
|
|
case DecoderFunctionType::kNormalDecode:
|
|
decoder->Decode(data_ptr, packet_len, sample_rate_hz, max_decoded_bytes,
|
|
decoded, &speech_type);
|
|
break;
|
|
case DecoderFunctionType::kRedundantDecode:
|
|
decoder->DecodeRedundant(data_ptr, packet_len, sample_rate_hz,
|
|
max_decoded_bytes, decoded, &speech_type);
|
|
break;
|
|
}
|
|
data_ptr += packet_len;
|
|
remaining_size -= packet_len;
|
|
}
|
|
}
|
|
|
|
// This function is similar to FuzzAudioDecoder, but also reads fuzzed data into
|
|
// RTP header values. The fuzzed data and values are sent to the decoder's
|
|
// IncomingPacket method.
|
|
void FuzzAudioDecoderIncomingPacket(const uint8_t* data,
|
|
size_t size,
|
|
AudioDecoder* decoder) {
|
|
const uint8_t* data_ptr = data;
|
|
size_t remaining_size = size;
|
|
size_t packet_len;
|
|
while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len)) {
|
|
uint16_t rtp_sequence_number;
|
|
if (!ParseInt(&data_ptr, &remaining_size, &rtp_sequence_number))
|
|
break;
|
|
uint32_t rtp_timestamp;
|
|
if (!ParseInt(&data_ptr, &remaining_size, &rtp_timestamp))
|
|
break;
|
|
uint32_t arrival_timestamp;
|
|
if (!ParseInt(&data_ptr, &remaining_size, &arrival_timestamp))
|
|
break;
|
|
if (remaining_size < packet_len)
|
|
break;
|
|
decoder->IncomingPacket(data_ptr, packet_len, rtp_sequence_number,
|
|
rtp_timestamp, arrival_timestamp);
|
|
data_ptr += packet_len;
|
|
remaining_size -= packet_len;
|
|
}
|
|
}
|
|
} // namespace webrtc
|