1 files changed, 53 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.cc
new file mode 100644
index 0000000000..837c26df56
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2020 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_encoder_fuzzer.h"
+
+#include <cstring>
+
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+
+// This function reads bytes from `data_view`, interprets them as RTP timestamp
+// and input samples, and sends them for encoding. The process continues until
+// no more data is available.
+void FuzzAudioEncoder(rtc::ArrayView<const uint8_t> data_view,
+                      std::unique_ptr<AudioEncoder> encoder) {
+  test::FuzzDataHelper data(data_view);
+  const size_t block_size_samples =
+      encoder->SampleRateHz() / 100 * encoder->NumChannels();
+  const size_t block_size_bytes = block_size_samples * sizeof(int16_t);
+  if (data_view.size() / block_size_bytes > 1000) {
+    // If the size of the fuzzer data is more than 1000 input blocks (i.e., more
+    // than 10 seconds), then don't fuzz at all for the fear of timing out.
+    return;
+  }
+
+  rtc::BufferT<int16_t> input_aligned(block_size_samples);
+  rtc::Buffer encoded;
+
+  // Each round in the loop below will need one block of samples + a 32-bit
+  // timestamp from the fuzzer input.
+  const size_t bytes_to_read = block_size_bytes + sizeof(uint32_t);
+  while (data.CanReadBytes(bytes_to_read)) {
+    const uint32_t timestamp = data.Read<uint32_t>();
+    auto byte_array = data.ReadByteArray(block_size_bytes);
+    // Align the data by copying to another array.
+    RTC_DCHECK_EQ(input_aligned.size() * sizeof(int16_t),
+                  byte_array.size() * sizeof(uint8_t));
+    memcpy(input_aligned.data(), byte_array.data(), byte_array.size());
+    auto info = encoder->Encode(timestamp, input_aligned, &encoded);
+  }
+}
+
+}  // namespace webrtc