chromium/third_party/webrtc/modules/audio_processing/audio_buffer.h

/*
 *  Copyright (c) 2011 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_AUDIO_BUFFER_H_
#define MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_

#include <stddef.h>
#include <stdint.h>

#include <memory>
#include <vector>

#include "api/audio/audio_processing.h"
#include "api/audio/audio_view.h"
#include "common_audio/channel_buffer.h"
#include "common_audio/include/audio_util.h"

namespace webrtc {

class PushSincResampler;
class SplittingFilter;

enum Band { … };

// Stores any audio data in a way that allows the audio processing module to
// operate on it in a controlled manner.
class AudioBuffer { … };

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_