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 MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_

#define MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_

#include <fstream>

#include <string>

#include "modules/audio_processing/test/audio_processing_simulator.h"

#ifdef WEBRTC_ANDROID_PLATFORM_BUILD

#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"

#else

#include "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:

  AecDumpBasedSimulator(const SimulationSettings& settings,

                        rtc::scoped_refptr<AudioProcessing> audio_processing,

                        std::unique_ptr<AudioProcessingBuilder> ap_builder);

  AecDumpBasedSimulator() = delete;

  AecDumpBasedSimulator(const AecDumpBasedSimulator&) = delete;

  AecDumpBasedSimulator& operator=(const AecDumpBasedSimulator&) = delete;

  ~AecDumpBasedSimulator() override;

  // Processes the messages in the aecdump file.

  void Process() override;

  // Analyzes the data in the aecdump file and reports the resulting statistics.

  void Analyze() override;

 private:

  void HandleEvent(const webrtc::audioproc::Event& event_msg,

                   int& num_forward_chunks_processed,

                   int& init_index);

  void HandleMessage(const webrtc::audioproc::Init& msg, int init_index);

  void HandleMessage(const webrtc::audioproc::Stream& msg);

  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);

  void HandleMessage(const webrtc::audioproc::Config& msg);

  void HandleMessage(const webrtc::audioproc::RuntimeSetting& msg);

  void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg);

  void PrepareReverseProcessStreamCall(

      const webrtc::audioproc::ReverseStream& msg);

  void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg);

  void MaybeOpenCallOrderFile();

  enum InterfaceType {

    kFixedInterface,

    kFloatInterface,

    kNotSpecified,

};

  FILE* dump_input_file_;

  std::unique_ptr<ChannelBuffer<float>> artificial_nearend_buf_;

  std::unique_ptr<ChannelBufferWavReader> artificial_nearend_buffer_reader_;

  bool artificial_nearend_eof_reported_ = false;

  InterfaceType interface_used_ = InterfaceType::kNotSpecified;

  std::unique_ptr<std::ofstream> call_order_output_file_;

  bool finished_processing_specified_init_block_ = false;

};

}  // namespace test

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_