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/*****************************************************************************
*
* Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
* while maintaining the original pitch by using a time domain WSOLA-like method
* with several performance-increasing tweaks.
*
* Anti-alias filter is used to prevent folding of high frequencies when
* transposing the sample rate with interpolation.
*
* Author : Copyright (c) Olli Parviainen
* Author e-mail : oparviai @ iki.fi
* File created : 13-Jan-2002
*
* Last changed : $Date: 2004/11/05 03:28:09 $
* File revision : $Revision: 1.1.1.1.2.1 $
*
* $Id: AAFilter.h,v 1.1.1.1.2.1 2004/11/05 03:28:09 mbrubeck Exp $
*
* License :
*
* SoundTouch sound processing library
* Copyright (c) Olli Parviainen
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*****************************************************************************/
#ifndef AAFilter_H
#define AAFilter_H
#include "STTypes.h"
class AAFilter
{
protected:
class FIRFilter *pFIR;
/// Low-pass filter cut-off frequency, negative = invalid
double cutoffFreq;
/// num of filter taps
uint length;
/// Calculate the FIR coefficients realizing the given cutoff-frequency
void calculateCoeffs();
public:
AAFilter(uint length);
~AAFilter();
/// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
/// frequency (nyquist frequency = 0.5). The filter will cut off the
/// frequencies than that.
void setCutoffFreq(double newCutoffFreq);
/// Sets number of FIR filter taps, i.e. ~filter complexity
void setLength(uint newLength);
uint getLength() const;
/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint evaluate(soundtouch::SAMPLETYPE *dest,
const soundtouch::SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
};
#endif
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