1 files changed, 79 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/aec3_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/aec3_fuzzer.cc
new file mode 100644
index 0000000000..a12ca30f63
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/aec3_fuzzer.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2019 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 "absl/types/optional.h"
+#include "modules/audio_processing/aec3/echo_canceller3.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace {
+using SampleRate = ::webrtc::AudioProcessing::NativeRate;
+
+void PrepareAudioBuffer(int sample_rate_hz,
+                        test::FuzzDataHelper* fuzz_data,
+                        AudioBuffer* buffer) {
+  float* const* channels = buffer->channels_f();
+  for (size_t i = 0; i < buffer->num_channels(); ++i) {
+    for (size_t j = 0; j < buffer->num_frames(); ++j) {
+      channels[i][j] =
+          static_cast<float>(fuzz_data->ReadOrDefaultValue<int16_t>(0));
+    }
+  }
+  if (sample_rate_hz == 32000 || sample_rate_hz == 48000) {
+    buffer->SplitIntoFrequencyBands();
+  }
+}
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 200000) {
+    return;
+  }
+
+  test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
+
+  constexpr int kSampleRates[] = {16000, 32000, 48000};
+  const int sample_rate_hz =
+      static_cast<size_t>(fuzz_data.SelectOneOf(kSampleRates));
+
+  constexpr int kMaxNumChannels = 9;
+  const size_t num_render_channels =
+      1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
+  const size_t num_capture_channels =
+      1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
+
+  EchoCanceller3 aec3(EchoCanceller3Config(),
+                      /*multichannel_config=*/absl::nullopt, sample_rate_hz,
+                      num_render_channels, num_capture_channels);
+
+  AudioBuffer capture_audio(sample_rate_hz, num_capture_channels,
+                            sample_rate_hz, num_capture_channels,
+                            sample_rate_hz, num_capture_channels);
+  AudioBuffer render_audio(sample_rate_hz, num_render_channels, sample_rate_hz,
+                           num_render_channels, sample_rate_hz,
+                           num_render_channels);
+
+  // Fuzz frames while there is still fuzzer data.
+  while (fuzz_data.BytesLeft() > 0) {
+    bool is_capture = fuzz_data.ReadOrDefaultValue(true);
+    bool level_changed = fuzz_data.ReadOrDefaultValue(true);
+    if (is_capture) {
+      PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &capture_audio);
+      aec3.ProcessCapture(&capture_audio, level_changed);
+    } else {
+      PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &render_audio);
+      aec3.AnalyzeRender(&render_audio);
+    }
+  }
+}
+}  // namespace webrtc