113 lines
3.6 KiB
Plaintext
113 lines
3.6 KiB
Plaintext
/*
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* This file is part of libsidplayfp, a SID player engine.
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*
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* Copyright 2011-2013 Leandro Nini <drfiemost@users.sourceforge.net>
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* Copyright 2007-2010 Antti Lankila
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* Copyright 2004 Dag Lem <resid@nimrod.no>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program 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
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
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*/
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#ifndef SINCRESAMPLER_H
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#define SINCRESAMPLER_H
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#include "Resampler.h"
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#include <string>
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#include <map>
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#include "../array.h"
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namespace reSIDfp
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{
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/**
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* This is the theoretically correct (and computationally intensive) audio sample generation.
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* The samples are generated by resampling to the specified sampling frequency.
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* The work rate is inversely proportional to the percentage of the bandwidth
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* allocated to the filter transition band.
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*
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* This implementation is based on the paper "A Flexible Sampling-Rate Conversion Method",
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* by J. O. Smith and P. Gosset, or rather on the expanded tutorial on the
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* [Digital Audio Resampling Home Page](http://www-ccrma.stanford.edu/~jos/resample/).
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*
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* By building shifted FIR tables with samples according to the sampling frequency,
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* this implementation dramatically reduces the computational effort in the
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* filter convolutions, without any loss of accuracy.
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* The filter convolutions are also vectorizable on current hardware.
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*/
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class SincResampler : public Resampler
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{
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private:
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/// Size of the ring buffer
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static const int RINGSIZE = 2048;
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private:
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/// Table of the fir filter coefficients
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matrix_t* firTable;
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int sampleIndex;
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/// Filter resolution
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int firRES;
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/// Filter length
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int firN;
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const int cyclesPerSample;
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int sampleOffset;
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int outputValue;
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short sample[RINGSIZE * 2];
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private:
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int fir(int subcycle);
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public:
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/**
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* Use a clock freqency of 985248Hz for PAL C64, 1022730Hz for NTSC C64.
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* The default end of passband frequency is pass_freq = 0.9*sample_freq/2
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* for sample frequencies up to ~ 44.1kHz, and 20kHz for higher sample frequencies.
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*
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* For resampling, the ratio between the clock frequency and the sample frequency
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* is limited as follows: 125*clock_freq/sample_freq < 16384
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* E.g. provided a clock frequency of ~ 1MHz, the sample frequency
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* can not be set lower than ~ 8kHz.
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* A lower sample frequency would make the resampling code overfill its 16k sample ring buffer.
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*
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* The end of passband frequency is also limited: pass_freq <= 0.9*sample_freq/2
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*
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* E.g. for a 44.1kHz sampling rate the end of passband frequency is limited
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* to slightly below 20kHz. This constraint ensures that the FIR table is not overfilled.
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*
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* @param clockFrequency System clock frequency at Hz
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* @param samplingFrequency Desired output sampling rate
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* @param highestAccurateFrequency
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*/
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SincResampler(double clockFrequency, double samplingFrequency, double highestAccurateFrequency);
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bool input(int input);
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int output() const { return outputValue; }
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void reset();
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};
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} // namespace reSIDfp
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#endif
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