Fuzzer tests for AudioDecoder's DecodeRedundant and IncomingPacket

webrtc/test/fuzzers/audio_decoder_fuzzer.cc

 /*
  *  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 <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,
                       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;
+  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;
-    packet_len = PacketSizeFromTwoBytes(data_ptr, remaining_size);
+  }
+}
+
+// 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