Added AudioProcessing fuzzer

webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc

Index: webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc
diff --git a/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc b/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc
new file mode 100644
index 0000000000000000000000000000000000000000..9878da98b3d13b06d913323185c543cd45039710
--- /dev/null
+++ b/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2017 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 "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/test/fuzzers/audio_processing_fuzzer.h"
+
+#include "webrtc/base/optional.h"
+
+namespace webrtc {
+
+std::unique_ptr<AudioProcessing> CreateAPM(const uint8_t** data,
+                                           size_t* remaining_size) {
+  // Parse boolean values for optionally enabling different
+  // configurable public components of APM.
+  auto exp_agc = ParseBool(data, remaining_size);
+  auto exp_ns = ParseBool(data, remaining_size);
+  auto bf = ParseBool(data, remaining_size);
+  auto ef = ParseBool(data, remaining_size);
+  auto raf = ParseBool(data, remaining_size);
+  auto da = ParseBool(data, remaining_size);
+  auto ie = ParseBool(data, remaining_size);
+  auto red = ParseBool(data, remaining_size);
+  auto lc = ParseBool(data, remaining_size);
+  auto hpf = ParseBool(data, remaining_size);
+  auto aec3 = ParseBool(data, remaining_size);
+
+  if (!(exp_agc && exp_ns && bf && ef && raf && da && ie && red && lc && hpf &&
+        aec3)) {
+    return nullptr;
+  }
+
+  // Components can be enabled through webrtc::Config and
+  // webrtc::AudioProcessingConfig.
+  Config config;
+
+  config.Set<ExperimentalAgc>(new ExperimentalAgc(*exp_agc));
+  config.Set<ExperimentalNs>(new ExperimentalNs(*exp_ns));
+  if (*bf) {
+    config.Set<Beamforming>(new Beamforming());
+  }
+  config.Set<ExtendedFilter>(new ExtendedFilter(*ef));
+  config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(*raf));
+  config.Set<DelayAgnostic>(new DelayAgnostic(*da));
+  config.Set<Intelligibility>(new Intelligibility(*ie));
+
+  std::unique_ptr<AudioProcessing> apm(AudioProcessing::Create(config));
+
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.residual_echo_detector.enabled = *red;
+  apm_config.level_controller.enabled = *lc;
+  apm_config.high_pass_filter.enabled = *hpf;
+  apm_config.echo_canceller3.enabled = *aec3;
+
+  apm->ApplyConfig(apm_config);
+
+  return apm;
+}
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  auto apm = CreateAPM(&data, &size);
+  FuzzAudioProcessing(data, size, std::move(apm));
+}
+}  // namespace webrtc