| Index: webrtc/test/fuzzers/audio_decoder_fuzzer.cc
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| diff --git a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
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| index fb5adb6cd8e55ca236138a00e762d75b5151cde5..54c56ad62a0375e89cc28978bf365f2da323db13 100644
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| --- a/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
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| +++ b/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
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| @@ -10,23 +10,39 @@
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|  
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|  #include "webrtc/test/fuzzers/audio_decoder_fuzzer.h"
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|  
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| +#include <limits>
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| +
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|  #include "webrtc/base/checks.h"
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| +#include "webrtc/base/optional.h"
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|  #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
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| +#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
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|  
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|  namespace webrtc {
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|  namespace {
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| -size_t PacketSizeFromTwoBytes(const uint8_t* data, size_t size) {
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| -  if (size < 2)
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| -    return 0;
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| -  return static_cast<size_t>((data[0] << 8) + data[1]);
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| +template <typename T, unsigned int B = sizeof(T)>
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| +bool ParseInt(const uint8_t** data, size_t* remaining_size, T* value) {
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| +  static_assert(std::numeric_limits<T>::is_integer, "Type must be an integer.");
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| +  static_assert(sizeof(T) <= sizeof(uint64_t),
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| +                "Cannot read wider than uint64_t.");
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| +  static_assert(B <= sizeof(T), "T must be at least B bytes wide.");
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| +  if (B > *remaining_size)
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| +    return false;
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| +  uint64_t val = ByteReader<uint64_t, B>::ReadBigEndian(*data);
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| +  *data += B;
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| +  *remaining_size -= B;
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| +  *value = static_cast<T>(val);
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| +  return true;
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|  }
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|  }  // namespace
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|  
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|  // This function reads two bytes from the beginning of |data|, interprets them
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|  // as the first packet length, and reads this many bytes if available. The
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|  // payload is inserted into the decoder, and the process continues until no more
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| -// data is available.
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| -void FuzzAudioDecoder(const uint8_t* data,
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| +// data is available. Either AudioDecoder::Decode or
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| +// AudioDecoder::DecodeRedundant is used, depending on the value of
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| +// |decode_type|.
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| +void FuzzAudioDecoder(DecoderFunctionType decode_type,
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| +                      const uint8_t* data,
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|                        size_t size,
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|                        AudioDecoder* decoder,
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|                        int sample_rate_hz,
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| @@ -34,16 +50,50 @@ void FuzzAudioDecoder(const uint8_t* data,
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|                        int16_t* decoded) {
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|    const uint8_t* data_ptr = data;
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|    size_t remaining_size = size;
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| -  size_t packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size);
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| -  while (packet_len != 0 && packet_len <= remaining_size - 2) {
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| -    data_ptr += 2;
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| -    remaining_size -= 2;
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| +  size_t packet_len;
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| +  while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) &&
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| +         packet_len <= remaining_size) {
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|      AudioDecoder::SpeechType speech_type;
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| -    decoder->Decode(data_ptr, packet_len, sample_rate_hz, max_decoded_bytes,
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| -                    decoded, &speech_type);
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| +    switch (decode_type) {
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| +      case DecoderFunctionType::kNormalDecode:
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| +        decoder->Decode(data_ptr, packet_len, sample_rate_hz, max_decoded_bytes,
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| +                        decoded, &speech_type);
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| +        break;
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| +      case DecoderFunctionType::kRedundantDecode:
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| +        decoder->DecodeRedundant(data_ptr, packet_len, sample_rate_hz,
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| +                                 max_decoded_bytes, decoded, &speech_type);
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| +        break;
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| +    }
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| +    data_ptr += packet_len;
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| +    remaining_size -= packet_len;
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| +  }
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| +}
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| +
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| +// This function is similar to FuzzAudioDecoder, but also reads fuzzed data into
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| +// RTP header values. The fuzzed data and values are sent to the decoder's
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| +// IncomingPacket method.
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| +void FuzzAudioDecoderIncomingPacket(const uint8_t* data,
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| +                                    size_t size,
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| +                                    AudioDecoder* decoder) {
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| +  const uint8_t* data_ptr = data;
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| +  size_t remaining_size = size;
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| +  size_t packet_len;
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| +  while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len)) {
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| +    uint16_t rtp_sequence_number;
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| +    if (!ParseInt(&data_ptr, &remaining_size, &rtp_sequence_number))
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| +      break;
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| +    uint32_t rtp_timestamp;
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| +    if (!ParseInt(&data_ptr, &remaining_size, &rtp_timestamp))
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| +      break;
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| +    uint32_t arrival_timestamp;
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| +    if (!ParseInt(&data_ptr, &remaining_size, &arrival_timestamp))
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| +      break;
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| +    if (remaining_size < packet_len)
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| +      break;
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| +    decoder->IncomingPacket(data_ptr, packet_len, rtp_sequence_number,
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| +                            rtp_timestamp, arrival_timestamp);
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|      data_ptr += packet_len;
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|      remaining_size -= packet_len;
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| -    packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size);
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|    }
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|  }
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|  }  // namespace webrtc
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| 
 |