cog/Frameworks/libsidplay/sidplay-residfp-code/.svn/pristine/01/019c099bc7e035fccac7c6f4890...

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/*
* This file is part of libsidplayfp, a SID player engine.
*
* Copyright 2011-2013 Leandro Nini <drfiemost@users.sourceforge.net>
* Copyright 2007-2010 Antti Lankila
* Copyright 2004 Dag Lem <resid@nimrod.no>
*
* 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.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
#include "SincResampler.h"
#include <cassert>
#include <cstring>
#include <cmath>
#include <iostream>
#include <sstream>
#include "siddefs-fp.h"
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#ifdef HAVE_MMINTRIN_H
# include <mmintrin.h>
#endif
namespace reSIDfp
{
typedef std::map<std::string, matrix_t> fir_cache_t;
/// Cache for the expensive FIR table computation results.
fir_cache_t FIR_CACHE;
/// Maximum error acceptable in I0 is 1e-6, or ~96 dB.
const double I0E = 1e-6;
const int BITS = 16;
/**
* I0() computes the 0th order modified Bessel function of the first kind.
* This function is originally from resample-1.5/filterkit.c by J. O. Smith.
* It is used to build the Kaiser window for resampling.
*
* @param x evaluate I0 at x
* @return value of I0 at x.
*/
double I0(double x)
{
double sum = 1., u = 1., n = 1.;
const double halfx = x / 2.;
do
{
const double temp = halfx / n;
u *= temp * temp;
sum += u;
n += 1.;
}
while (u >= I0E * sum);
return sum;
}
/**
* Calculate convolution with sample and sinc.
*
* @param a sample buffer input
* @param b sinc
* @param bLength length of the sinc buffer
* @return convolved result
*/
int convolve(const short* a, const short* b, int bLength)
{
#ifdef HAVE_MMINTRIN_H
__m64 acc = _mm_setzero_si64();
const int n = bLength / 4;
for (int i = 0; i < n; i++)
{
const __m64 tmp = _mm_madd_pi16(*(__m64*)a, *(__m64*)b);
acc = _mm_add_pi16(acc, tmp);
a += 4;
b += 4;
}
int out = _mm_cvtsi64_si32(acc) + _mm_cvtsi64_si32(_mm_srli_si64(acc, 32));
_mm_empty();
bLength &= 3;
#else
int out = 0;
#endif
for (int i = 0; i < bLength; i++)
{
out += *a++ * *b++;
}
return (out + (1 << 14)) >> 15;
}
int SincResampler::fir(int subcycle)
{
// find the first of the nearest fir tables close to the phase
int firTableFirst = (subcycle * firRES >> 10);
const int firTableOffset = (subcycle * firRES) & 0x3ff;
// find firN most recent samples, plus one extra in case the FIR wraps.
int sampleStart = sampleIndex - firN + RINGSIZE - 1;
const int v1 = convolve(sample + sampleStart, (*firTable)[firTableFirst], firN);
// Use next FIR table, wrap around to first FIR table using
// previous sample.
if (unlikely(++firTableFirst == firRES))
{
firTableFirst = 0;
++sampleStart;
}
const int v2 = convolve(sample + sampleStart, (*firTable)[firTableFirst], firN);
// Linear interpolation between the sinc tables yields good
// approximation for the exact value.
return v1 + (firTableOffset * (v2 - v1) >> 10);
}
SincResampler::SincResampler(double clockFrequency, double samplingFrequency, double highestAccurateFrequency) :
sampleIndex(0),
cyclesPerSample(static_cast<int>(clockFrequency / samplingFrequency * 1024.)),
sampleOffset(0),
outputValue(0)
{
// 16 bits -> -96dB stopband attenuation.
const double A = -20. * log10(1.0 / (1 << BITS));
// A fraction of the bandwidth is allocated to the transition band, which we double
// because we design the filter to transition halfway at nyquist.
const double dw = (1. - 2.*highestAccurateFrequency / samplingFrequency) * M_PI * 2.;
// For calculation of beta and N see the reference for the kaiserord
// function in the MATLAB Signal Processing Toolbox:
// http://www.mathworks.com/access/helpdesk/help/toolbox/signal/kaiserord.html
const double beta = 0.1102 * (A - 8.7);
const double I0beta = I0(beta);
const double cyclesPerSampleD = clockFrequency / samplingFrequency;
{
// The filter order will maximally be 124 with the current constraints.
// N >= (96.33 - 7.95)/(2 * pi * 2.285 * (maxfreq - passbandfreq) >= 123
// The filter order is equal to the number of zero crossings, i.e.
// it should be an even number (sinc is symmetric about x = 0).
int N = static_cast<int>((A - 7.95) / (2.285 * dw) + 0.5);
N += N & 1;
// The filter length is equal to the filter order + 1.
// The filter length must be an odd number (sinc is symmetric about
// x = 0).
firN = static_cast<int>(N * cyclesPerSampleD) + 1;
firN |= 1;
// Check whether the sample ring buffer would overflow.
assert(firN < RINGSIZE);
// Error is bounded by err < 1.234 / L^2, so L = sqrt(1.234 / (2^-16)) = sqrt(1.234 * 2^16).
firRES = static_cast<int>(ceil(sqrt(1.234 * (1 << BITS)) / cyclesPerSampleD));
// firN*firRES represent the total resolution of the sinc sampling. JOS
// recommends a length of 2^BITS, but we don't quite use that good a filter.
// The filter test program indicates that the filter performs well, though. */
}
std::ostringstream o;
o << firN << "," << firRES << "," << cyclesPerSampleD;
const std::string firKey = o.str();
fir_cache_t::iterator lb = FIR_CACHE.lower_bound(firKey);
// The FIR computation is expensive and we set sampling parameters often, but
// from a very small set of choices. Thus, caching is used to speed initialization.
if (lb != FIR_CACHE.end() && !(FIR_CACHE.key_comp()(firKey, lb->first)))
{
firTable = &(lb->second);
}
else
{
// Allocate memory for FIR tables.
matrix_t tempTable(firRES, firN);
firTable = &(FIR_CACHE.insert(lb, fir_cache_t::value_type(firKey, tempTable))->second);
// The cutoff frequency is midway through the transition band, in effect the same as nyquist.
const double wc = M_PI;
// Calculate the sinc tables.
const double scale = 32768.0 * wc / cyclesPerSampleD / M_PI;
for (int i = 0; i < firRES; i++)
{
const double jPhase = (double) i / firRES + firN / 2;
for (int j = 0; j < firN; j++)
{
const double x = j - jPhase;
const double xt = x / (firN / 2);
const double kaiserXt = fabs(xt) < 1. ? I0(beta * sqrt(1. - xt * xt)) / I0beta : 0.;
const double wt = wc * x / cyclesPerSampleD;
const double sincWt = fabs(wt) >= 1e-8 ? sin(wt) / wt : 1.;
(*firTable)[i][j] = static_cast<short>(scale * sincWt * kaiserXt);
}
}
}
}
bool SincResampler::input(int input)
{
bool ready = false;
sample[sampleIndex] = sample[sampleIndex + RINGSIZE] = input;
sampleIndex = (sampleIndex + 1) & (RINGSIZE - 1);
if (sampleOffset < 1024)
{
outputValue = fir(sampleOffset);
ready = true;
sampleOffset += cyclesPerSample;
}
sampleOffset -= 1024;
return ready;
}
void SincResampler::reset()
{
memset(sample, 0, RINGSIZE * 2 * sizeof(sample[0]));
sampleOffset = 0;
}
} // namespace reSIDfp