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