Create a fuzzer for the Opus encoder

The fuzzer is very simple. It only considers the default encoder
configuration at this point.

Bug: chromium:826914
Change-Id: Ifa248a1dba80efb231807750e40082ec5580636a
Reviewed-on: https://webrtc-review.googlesource.com/75261
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23192}

test/fuzzers/BUILD.gn

@@ -315,6 +315,19 @@ webrtc_fuzzer_test("audio_decoder_opus_redundant_fuzzer") {
   ]
 }
 
+webrtc_fuzzer_test("audio_encoder_opus_fuzzer") {
+  sources = [
+    "audio_encoder_opus_fuzzer.cc",
+  ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../api:array_view",
+    "../../api/audio_codecs/opus:audio_encoder_opus",
+    "../../rtc_base:checks",
+    "../../rtc_base:rtc_base_approved",
+  ]
+}
+
 webrtc_fuzzer_test("turn_unwrap_fuzzer") {
   sources = [
     "turn_unwrap_fuzzer.cc",

test/fuzzers/audio_encoder_opus_fuzzer.cc

@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2018 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 "api/array_view.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace {
+
+// 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,
+                      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
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  AudioEncoderOpus::Config config;
+  config.frame_size_ms = 20;
+  RTC_CHECK(config.IsOk());
+  constexpr int kPayloadType = 100;
+  std::unique_ptr<AudioEncoder> enc =
+      AudioEncoderOpus::MakeAudioEncoder(config, kPayloadType);
+  FuzzAudioEncoder(rtc::ArrayView<const uint8_t>(data, size), enc.get());
+}
+
+}  // namespace webrtc