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//////////////////////////////////////////////////////////////////////////////// |
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/// |
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/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo |
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/// while maintaining the original pitch by using a time domain WSOLA-like method |
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/// with several performance-increasing tweaks. |
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/// |
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/// Anti-alias filter is used to prevent folding of high frequencies when |
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/// transposing the sample rate with interpolation. |
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/// |
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/// Author : Copyright (c) Olli Parviainen |
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/// Author e-mail : oparviai 'at' iki.fi |
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/// SoundTouch WWW: http://www.surina.net/soundtouch |
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/// |
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//////////////////////////////////////////////////////////////////////////////// |
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// |
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// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $ |
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// File revision : $Revision: 4 $ |
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// |
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// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $ |
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// |
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//////////////////////////////////////////////////////////////////////////////// |
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// |
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// License : |
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// |
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// SoundTouch audio processing library |
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// Copyright (c) Olli Parviainen |
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// |
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// This library is free software; you can redistribute it and/or |
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// modify it under the terms of the GNU Lesser General Public |
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// License as published by the Free Software Foundation; either |
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// version 2.1 of the License, or (at your option) any later version. |
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// |
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// This library is distributed in the hope that it will be useful, |
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// but WITHOUT ANY WARRANTY; without even the implied warranty of |
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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// Lesser General Public License for more details. |
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// |
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// You should have received a copy of the GNU Lesser General Public |
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// License along with this library; if not, write to the Free Software |
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// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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// |
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//////////////////////////////////////////////////////////////////////////////// |
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|
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#ifndef AAFilter_H |
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#define AAFilter_H |
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|
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#include "STTypes.h" |
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|
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namespace soundtouch |
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{ |
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|
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class AAFilter |
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{ |
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protected: |
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class FIRFilter *pFIR; |
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|
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/// Low-pass filter cut-off frequency, negative = invalid |
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double cutoffFreq; |
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|
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/// num of filter taps |
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uint length; |
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|
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/// Calculate the FIR coefficients realizing the given cutoff-frequency |
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void calculateCoeffs(); |
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public: |
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AAFilter(uint length); |
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|
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~AAFilter(); |
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|
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/// Sets new anti-alias filter cut-off edge frequency, scaled to sampling |
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/// frequency (nyquist frequency = 0.5). The filter will cut off the |
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/// frequencies than that. |
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void setCutoffFreq(double newCutoffFreq); |
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|
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/// Sets number of FIR filter taps, i.e. ~filter complexity |
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void setLength(uint newLength); |
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|
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uint getLength() const; |
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|
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/// Applies the filter to the given sequence of samples. |
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/// Note : The amount of outputted samples is by value of 'filter length' |
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/// smaller than the amount of input samples. |
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uint evaluate(SAMPLETYPE *dest, |
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const SAMPLETYPE *src, |
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uint numSamples, |
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uint numChannels) const; |
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}; |
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|
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} |
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|
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#endif |
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