Extension and refactoring of the audioproc_f tool to be a fully fledged tool for audio processing m…

webrtc/modules/audio_processing/test/aec_dump_based_simulator.h

Index: webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
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+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_
+
+#include "webrtc/modules/audio_processing/test/audio_processing_simulator.h"
+
+#include "webrtc/base/constructormagic.h"
+
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "webrtc/modules/audio_processing/debug.pb.h"
+#endif
+
+namespace webrtc {
+namespace test {
+
+// Used to perform an audio processing simulation from an aec dump.
+class AecDumpBasedSimulator final : public AudioProcessingSimulator {
+ public:
+  explicit AecDumpBasedSimulator(const SimulationSettings& settings)
+      : AudioProcessingSimulator(settings) {}
+  virtual ~AecDumpBasedSimulator() {}
+
+  // Processes the messages in the aecdump file.
+  void Process() override;
+
+ private:
+  void HandleMessage(const webrtc::audioproc::Init& msg);
+  void HandleMessage(const webrtc::audioproc::Stream& msg);
+  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
+  void HandleMessage(const webrtc::audioproc::Config& msg);
+  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg);
+  void PrepareReverseProcessStreamCall(
+      const webrtc::audioproc::ReverseStream& msg);
+  void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);
+
+  enum InterfaceType {
+    kFixedInterface,
+    kFloatInterface,
+    kNotSpecified,
+  };
+
+  FILE* dump_input_file_;
+  InterfaceType interface_used_ = InterfaceType::kNotSpecified;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_