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Diffstat (limited to 'libs/qm-dsp/dsp/rateconversion/Resampler.h')
-rw-r--r-- | libs/qm-dsp/dsp/rateconversion/Resampler.h | 102 |
1 files changed, 0 insertions, 102 deletions
diff --git a/libs/qm-dsp/dsp/rateconversion/Resampler.h b/libs/qm-dsp/dsp/rateconversion/Resampler.h deleted file mode 100644 index 92c0169ba0..0000000000 --- a/libs/qm-dsp/dsp/rateconversion/Resampler.h +++ /dev/null @@ -1,102 +0,0 @@ -/* -*- 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 - |