/*
 *  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_SIMULATOR_BUFFERS_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_

#include <memory>
#include <vector>

#include "webrtc/base/random.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"

namespace webrtc {
namespace test {

struct SimulatorBuffers {
  SimulatorBuffers(int render_input_sample_rate_hz,
                   int capture_input_sample_rate_hz,
                   int render_output_sample_rate_hz,
                   int capture_output_sample_rate_hz,
                   size_t num_render_input_channels,
                   size_t num_capture_input_channels,
                   size_t num_render_output_channels,
                   size_t num_capture_output_channels);
  ~SimulatorBuffers();

  void CreateConfigAndBuffer(int sample_rate_hz,
                             size_t num_channels,
                             Random* rand_gen,
                             std::unique_ptr<AudioBuffer>* buffer,
                             StreamConfig* config,
                             std::vector<float*>* buffer_data,
                             std::vector<float>* buffer_data_samples);

  void UpdateInputBuffers();

  std::unique_ptr<AudioBuffer> render_input_buffer;
  std::unique_ptr<AudioBuffer> capture_input_buffer;
  std::unique_ptr<AudioBuffer> render_output_buffer;
  std::unique_ptr<AudioBuffer> capture_output_buffer;
  StreamConfig render_input_config;
  StreamConfig capture_input_config;
  StreamConfig render_output_config;
  StreamConfig capture_output_config;
  std::vector<float*> render_input;
  std::vector<float> render_input_samples;
  std::vector<float*> capture_input;
  std::vector<float> capture_input_samples;
  std::vector<float*> render_output;
  std::vector<float> render_output_samples;
  std::vector<float*> capture_output;
  std::vector<float> capture_output_samples;
};

}  // namespace test
}  // namespace webrtc

#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_