c@362: /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
c@375: /*
c@375:     QM DSP Library
c@375: 
c@375:     Centre for Digital Music, Queen Mary, University of London.
c@375:     This file by Chris Cannam.
c@375: 
c@375:     This program is free software; you can redistribute it and/or
c@375:     modify it under the terms of the GNU General Public License as
c@375:     published by the Free Software Foundation; either version 2 of the
c@375:     License, or (at your option) any later version.  See the file
c@375:     COPYING included with this distribution for more information.
c@375: */
c@362: 
c@362: #ifndef RESAMPLER_H
c@362: #define RESAMPLER_H
c@362: 
c@362: #include <vector>
c@362: 
c@375: /**
c@375:  * Resampler resamples a stream from one integer sample rate to
c@375:  * another (arbitrary) rate, using a kaiser-windowed sinc filter.  The
c@375:  * results and performance are pretty similar to libraries such as
c@375:  * libsamplerate, though this implementation does not support
c@375:  * time-varying ratios (the ratio is fixed on construction).
c@375:  *
c@375:  * See also Decimator, which is faster and rougher but supports only
c@375:  * power-of-two downsampling factors.
c@375:  */
c@362: class Resampler
c@362: {
c@362: public:
c@362:     /**
c@362:      * Construct a Resampler to resample from sourceRate to
c@362:      * targetRate.
c@362:      */
c@362:     Resampler(int sourceRate, int targetRate);
c@374: 
c@374:     /**
c@374:      * Construct a Resampler to resample from sourceRate to
c@374:      * targetRate, using the given filter parameters.
c@374:      */
c@374:     Resampler(int sourceRate, int targetRate,
c@374:               double snr, double bandwidth);
c@374: 
c@362:     virtual ~Resampler();
c@362: 
c@362:     /**
c@362:      * Read n input samples from src and write resampled data to
c@362:      * dst. The return value is the number of samples written, which
c@362:      * will be no more than ceil((n * targetRate) / sourceRate). The
c@362:      * caller must ensure the dst buffer has enough space for the
c@362:      * samples returned.
c@362:      */
c@362:     int process(const double *src, double *dst, int n);
c@362: 
c@362:     /**
c@362:      * Return the number of samples of latency at the output due by
c@362:      * the filter. (That is, the output will be delayed by this number
c@362:      * of samples relative to the input.)
c@362:      */
c@362:     int getLatency() const { return m_latency; }
c@362: 
c@363:     /**
c@363:      * Carry out a one-off resample of a single block of n
c@363:      * samples. The output is latency-compensated.
c@363:      */
c@363:     static std::vector<double> resample
c@363:     (int sourceRate, int targetRate, const double *data, int n);
c@363: 
c@362: private:
c@362:     int m_sourceRate;
c@362:     int m_targetRate;
c@362:     int m_gcd;
c@362:     int m_filterLength;
c@362:     int m_bufferLength;
c@362:     int m_latency;
c@381:     double m_peakToPole;
c@362:     
c@362:     struct Phase {
c@362:         int nextPhase;
c@362:         std::vector<double> filter;
c@362:         int drop;
c@362:     };
c@362: 
c@362:     Phase *m_phaseData;
c@362:     int m_phase;
c@364:     std::vector<double> m_buffer;
c@370:     int m_bufferOrigin;
c@362: 
c@374:     void initialise(double, double);
c@366:     double reconstructOne();
c@362: };
c@362: 
c@362: #endif
c@362: