1 files changed, 77 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.cc
new file mode 100644
index 0000000000..1db332eeb5
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.cc
@@ -0,0 +1,77 @@
+/*
+ *  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;
+  }
+}
+
+}  // namespace webrtc