Chromium Code Reviews| Index: webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
| diff --git a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
| index fb5adb6cd8e55ca236138a00e762d75b5151cde5..1c4d9d5660720b740ffec28d86e4c5a9be870e06 100644 |
| --- a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
| +++ b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc |
| @@ -10,15 +10,28 @@ |
| #include "webrtc/test/fuzzers/audio_decoder_fuzzer.h" |
| +#include <limits> |
| + |
| #include "webrtc/base/checks.h" |
| +#include "webrtc/base/optional.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
| +#include "webrtc/modules/rtp_rtcp/source/byte_io.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]); |
| +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 |
| @@ -34,16 +47,63 @@ void FuzzAudioDecoder(const uint8_t* data, |
| 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; |
| + size_t packet_len; |
| + while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) && |
| + packet_len <= remaining_size) { |
| 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); |
| + } |
| +} |
| + |
| +// This function is identical to FuzzAudioDecoder above, with the distinction |
| +// that it call DecodeRedundant instead of Decode. |
|
pbos-webrtc
2016/01/26 14:18:12
Can you refactor this into one method with an enum
hlundin-webrtc
2016/01/26 22:32:31
Done.
|
| +void FuzzAudioDecoderRedundant(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; |
| + while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) && |
| + packet_len <= remaining_size) { |
| + AudioDecoder::SpeechType speech_type; |
| + decoder->DecodeRedundant(data_ptr, packet_len, sample_rate_hz, |
| + max_decoded_bytes, decoded, &speech_type); |
| + data_ptr += packet_len; |
| + remaining_size -= packet_len; |
| + } |
| +} |
| + |
| +// This function is similar to FuzzAudioDecoder, but also reads fuzzed data into |
|
pbos-webrtc
2016/01/26 14:18:12
Same as above if possible.
pbos-webrtc
2016/01/26 14:19:08
Not very possible, ignore this. :)
hlundin-webrtc
2016/01/26 22:32:31
Acknowledged.
|
| +// 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 |