michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: ///
michael@0: /// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo 
michael@0: /// while maintaining the original pitch by using a time domain WSOLA-like method 
michael@0: /// with several performance-increasing tweaks.
michael@0: ///
michael@0: /// Anti-alias filter is used to prevent folding of high frequencies when 
michael@0: /// transposing the sample rate with interpolation.
michael@0: ///
michael@0: /// Author        : Copyright (c) Olli Parviainen
michael@0: /// Author e-mail : oparviai 'at' iki.fi
michael@0: /// SoundTouch WWW: http://www.surina.net/soundtouch
michael@0: ///
michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: //
michael@0: // Last changed  : $Date: 2014-01-07 13:41:23 -0600 (Tue, 07 Jan 2014) $
michael@0: // File revision : $Revision: 4 $
michael@0: //
michael@0: // $Id: AAFilter.h 187 2014-01-07 19:41:23Z oparviai $
michael@0: //
michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: //
michael@0: // License :
michael@0: //
michael@0: //  SoundTouch audio processing library
michael@0: //  Copyright (c) Olli Parviainen
michael@0: //
michael@0: //  This library is free software; you can redistribute it and/or
michael@0: //  modify it under the terms of the GNU Lesser General Public
michael@0: //  License as published by the Free Software Foundation; either
michael@0: //  version 2.1 of the License, or (at your option) any later version.
michael@0: //
michael@0: //  This library is distributed in the hope that it will be useful,
michael@0: //  but WITHOUT ANY WARRANTY; without even the implied warranty of
michael@0: //  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
michael@0: //  Lesser General Public License for more details.
michael@0: //
michael@0: //  You should have received a copy of the GNU Lesser General Public
michael@0: //  License along with this library; if not, write to the Free Software
michael@0: //  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
michael@0: //
michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: 
michael@0: #ifndef AAFilter_H
michael@0: #define AAFilter_H
michael@0: 
michael@0: #include "STTypes.h"
michael@0: #include "FIFOSampleBuffer.h"
michael@0: 
michael@0: namespace soundtouch
michael@0: {
michael@0: 
michael@0: class AAFilter
michael@0: {
michael@0: protected:
michael@0:     class FIRFilter *pFIR;
michael@0: 
michael@0:     /// Low-pass filter cut-off frequency, negative = invalid
michael@0:     double cutoffFreq;
michael@0: 
michael@0:     /// num of filter taps
michael@0:     uint length;
michael@0: 
michael@0:     /// Calculate the FIR coefficients realizing the given cutoff-frequency
michael@0:     void calculateCoeffs();
michael@0: public:
michael@0:     AAFilter(uint length);
michael@0: 
michael@0:     ~AAFilter();
michael@0: 
michael@0:     /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling 
michael@0:     /// frequency (nyquist frequency = 0.5). The filter will cut off the 
michael@0:     /// frequencies than that.
michael@0:     void setCutoffFreq(double newCutoffFreq);
michael@0: 
michael@0:     /// Sets number of FIR filter taps, i.e. ~filter complexity
michael@0:     void setLength(uint newLength);
michael@0: 
michael@0:     uint getLength() const;
michael@0: 
michael@0:     /// Applies the filter to the given sequence of samples. 
michael@0:     /// Note : The amount of outputted samples is by value of 'filter length' 
michael@0:     /// smaller than the amount of input samples.
michael@0:     uint evaluate(SAMPLETYPE *dest, 
michael@0:                   const SAMPLETYPE *src, 
michael@0:                   uint numSamples, 
michael@0:                   uint numChannels) const;
michael@0: 
michael@0:     /// Applies the filter to the given src & dest pipes, so that processed amount of
michael@0:     /// samples get removed from src, and produced amount added to dest 
michael@0:     /// Note : The amount of outputted samples is by value of 'filter length' 
michael@0:     /// smaller than the amount of input samples.
michael@0:     uint evaluate(FIFOSampleBuffer &dest, 
michael@0:                   FIFOSampleBuffer &src) const;
michael@0: 
michael@0: };
michael@0: 
michael@0: }
michael@0: 
michael@0: #endif