| 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..54c56ad62a0375e89cc28978bf365f2da323db13 100644
|
| --- a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
|
| +++ b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
|
| @@ -10,23 +10,39 @@
|
|
|
| #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
|
|
|
| // 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,
|
| +// 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,
|
| @@ -34,16 +50,50 @@ 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);
|
| + 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;
|
| - packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size);
|
| }
|
| }
|
| } // namespace webrtc
|
|
|