chromium/third_party/webrtc/common_audio/audio_converter.h

/*
 *  Copyright (c) 2014 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 COMMON_AUDIO_AUDIO_CONVERTER_H_
#define COMMON_AUDIO_AUDIO_CONVERTER_H_

#include <stddef.h>

#include <memory>

namespace webrtc {

// Format conversion (remixing and resampling) for audio. Only simple remixing
// conversions are supported: downmix to mono (i.e. `dst_channels` == 1) or
// upmix from mono (i.e. |src_channels == 1|).
//
// The source and destination chunks have the same duration in time; specifying
// the number of frames is equivalent to specifying the sample rates.
class AudioConverter { … };

}  // namespace webrtc

#endif  // COMMON_AUDIO_AUDIO_CONVERTER_H_