Fuzz AEC3

This fuzzer fuzzes AEC3 with the default configuration and variable sample rates and channel counts.

Bug: None
Change-Id: I0d178a320b75fc4cc389657fa2b99931f359b517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160646
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29967}

test/fuzzers/BUILD.gn

@@ -526,6 +526,24 @@ webrtc_fuzzer_test("aec3_config_json_fuzzer") {
   seed_corpus = "corpora/aec3-config-json-corpus"
 }
 
+webrtc_fuzzer_test("aec3_fuzzer") {
+  defines = []
+  if (apm_debug_dump) {
+    defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ]
+  } else {
+    defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ]
+  }
+  sources = [
+    "aec3_fuzzer.cc",
+  ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../modules/audio_processing:audio_buffer",
+    "../../modules/audio_processing/aec3",
+    "//modules/audio_processing:api",
+  ]
+}
+
 webrtc_fuzzer_test("comfort_noise_decoder_fuzzer") {
   sources = [
     "comfort_noise_decoder_fuzzer.cc",

test/fuzzers/aec3_fuzzer.cc

@@ -0,0 +1,77 @@
+/*
+ *  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 "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(), 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