chromium/third_party/webrtc/common_audio/resampler/sinc_resampler.h

/*
 *  Copyright (c) 2013 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.
 */

// Modified from the Chromium original here:
// src/media/base/sinc_resampler.h

#ifndef COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
#define COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_

#include <stddef.h>

#include <memory>

#include "rtc_base/gtest_prod_util.h"
#include "rtc_base/memory/aligned_malloc.h"
#include "rtc_base/system/arch.h"

namespace webrtc {

// Callback class for providing more data into the resampler.  Expects `frames`
// of data to be rendered into `destination`; zero padded if not enough frames
// are available to satisfy the request.
class SincResamplerCallback { … };

// SincResampler is a high-quality single-channel sample-rate converter.
class SincResampler { … };

}  // namespace webrtc

#endif  // COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_