michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: /// 
michael@0: /// Sample rate transposer. Changes sample rate by using linear interpolation 
michael@0: /// together with anti-alias filtering (first order interpolation with anti-
michael@0: /// alias filtering should be quite adequate for this application).
michael@0: ///
michael@0: /// Use either of the derived classes of 'RateTransposerInteger' or 
michael@0: /// 'RateTransposerFloat' for corresponding integer/floating point tranposing
michael@0: /// algorithm implementation.
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: 2014-04-06 10:57:21 -0500 (Sun, 06 Apr 2014) $
michael@0: // File revision : $Revision: 4 $
michael@0: //
michael@0: // $Id: RateTransposer.h 195 2014-04-06 15:57:21Z 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 RateTransposer_H
michael@0: #define RateTransposer_H
michael@0: 
michael@0: #include <stddef.h>
michael@0: #include "AAFilter.h"
michael@0: #include "FIFOSamplePipe.h"
michael@0: #include "FIFOSampleBuffer.h"
michael@0: 
michael@0: #include "STTypes.h"
michael@0: 
michael@0: namespace soundtouch
michael@0: {
michael@0: 
michael@0: /// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)
michael@0: class TransposerBase
michael@0: {
michael@0: public:
michael@0:         enum ALGORITHM {
michael@0:         LINEAR = 0,
michael@0:         CUBIC,
michael@0:         SHANNON
michael@0:     };
michael@0: 
michael@0: protected:
michael@0:     virtual void resetRegisters() = 0;
michael@0: 
michael@0:     virtual int transposeMono(SAMPLETYPE *dest, 
michael@0:                         const SAMPLETYPE *src, 
michael@0:                         int &srcSamples)  = 0;
michael@0:     virtual int transposeStereo(SAMPLETYPE *dest, 
michael@0:                         const SAMPLETYPE *src, 
michael@0:                         int &srcSamples) = 0;
michael@0:     virtual int transposeMulti(SAMPLETYPE *dest, 
michael@0:                         const SAMPLETYPE *src, 
michael@0:                         int &srcSamples) = 0;
michael@0: 
michael@0:     static ALGORITHM algorithm;
michael@0: 
michael@0: public:
michael@0:     float rate;
michael@0:     int numChannels;
michael@0: 
michael@0:     TransposerBase();
michael@0:     virtual ~TransposerBase();
michael@0: 
michael@0:     virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
michael@0:     virtual void setRate(float newRate);
michael@0:     virtual void setChannels(int channels);
michael@0: 
michael@0:     // static factory function
michael@0:     static TransposerBase *newInstance();
michael@0: 
michael@0:     // static function to set interpolation algorithm
michael@0:     static void setAlgorithm(ALGORITHM a);
michael@0: };
michael@0: 
michael@0: 
michael@0: /// A common linear samplerate transposer class.
michael@0: ///
michael@0: class RateTransposer : public FIFOProcessor
michael@0: {
michael@0: protected:
michael@0:     /// Anti-alias filter object
michael@0:     AAFilter *pAAFilter;
michael@0:     TransposerBase *pTransposer;
michael@0: 
michael@0:     /// Buffer for collecting samples to feed the anti-alias filter between
michael@0:     /// two batches
michael@0:     FIFOSampleBuffer inputBuffer;
michael@0: 
michael@0:     /// Buffer for keeping samples between transposing & anti-alias filter
michael@0:     FIFOSampleBuffer midBuffer;
michael@0: 
michael@0:     /// Output sample buffer
michael@0:     FIFOSampleBuffer outputBuffer;
michael@0: 
michael@0:     bool bUseAAFilter;
michael@0: 
michael@0: 
michael@0:     /// Transposes sample rate by applying anti-alias filter to prevent folding. 
michael@0:     /// Returns amount of samples returned in the "dest" buffer.
michael@0:     /// The maximum amount of samples that can be returned at a time is set by
michael@0:     /// the 'set_returnBuffer_size' function.
michael@0:     void processSamples(const SAMPLETYPE *src, 
michael@0:                         uint numSamples);
michael@0: 
michael@0: public:
michael@0:     RateTransposer();
michael@0:     virtual ~RateTransposer();
michael@0: 
michael@0:     /// Operator 'new' is overloaded so that it automatically creates a suitable instance 
michael@0:     /// depending on if we're to use integer or floating point arithmetics.
michael@0: //    static void *operator new(size_t s);
michael@0: 
michael@0:     /// Use this function instead of "new" operator to create a new instance of this class. 
michael@0:     /// This function automatically chooses a correct implementation, depending on if 
michael@0:     /// integer ot floating point arithmetics are to be used.
michael@0: //    static RateTransposer *newInstance();
michael@0: 
michael@0:     /// Returns the output buffer object
michael@0:     FIFOSamplePipe *getOutput() { return &outputBuffer; };
michael@0: 
michael@0:     /// Returns the store buffer object
michael@0: //    FIFOSamplePipe *getStore() { return &storeBuffer; };
michael@0: 
michael@0:     /// Return anti-alias filter object
michael@0:     AAFilter *getAAFilter();
michael@0: 
michael@0:     /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
michael@0:     void enableAAFilter(bool newMode);
michael@0: 
michael@0:     /// Returns nonzero if anti-alias filter is enabled.
michael@0:     bool isAAFilterEnabled() const;
michael@0: 
michael@0:     /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 
michael@0:     /// rate, larger faster rates.
michael@0:     virtual void setRate(float newRate);
michael@0: 
michael@0:     /// Sets the number of channels, 1 = mono, 2 = stereo
michael@0:     void setChannels(int channels);
michael@0: 
michael@0:     /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
michael@0:     /// the input of the object.
michael@0:     void putSamples(const SAMPLETYPE *samples, uint numSamples);
michael@0: 
michael@0:     /// Clears all the samples in the object
michael@0:     void clear();
michael@0: 
michael@0:     /// Returns nonzero if there aren't any samples available for outputting.
michael@0:     int isEmpty() const;
michael@0: };
michael@0: 
michael@0: }
michael@0: 
michael@0: #endif