🛠️🐜 Antkeeper superbuild with dependencies included
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367 lines
12 KiB
367 lines
12 KiB
/*
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* Sinc interpolator coefficient and delta generator for the OpenAL Soft
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* cross platform audio library.
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*
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* Copyright (C) 2015 by Christopher Fitzgerald.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library 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 GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
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* MA 02110-1301 USA
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*
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* Or visit: http://www.gnu.org/licenses/old-licenses/lgpl-2.0.html
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*
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* --------------------------------------------------------------------------
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*
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* This is a modified version of the bandlimited windowed sinc interpolator
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* algorithm presented here:
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*
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* Smith, J.O. "Windowed Sinc Interpolation", in
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* Physical Audio Signal Processing,
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* https://ccrma.stanford.edu/~jos/pasp/Windowed_Sinc_Interpolation.html,
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* online book,
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* accessed October 2012.
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*/
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#define _UNICODE
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#include <stdio.h>
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#include <math.h>
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#include <string.h>
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#include <stdlib.h>
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#include "../common/win_main_utf8.h"
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#ifndef M_PI
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#define M_PI (3.14159265358979323846)
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#endif
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#if !(defined(_ISOC99_SOURCE) || (defined(_POSIX_C_SOURCE) && _POSIX_C_SOURCE >= 200112L))
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#define log2(x) (log(x) / log(2.0))
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#endif
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/* Same as in alu.h! */
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#define FRACTIONBITS (12)
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#define FRACTIONONE (1<<FRACTIONBITS)
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// The number of distinct scale and phase intervals within the filter table.
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// Must be the same as in alu.h!
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#define BSINC_SCALE_COUNT (16)
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#define BSINC_PHASE_COUNT (16)
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/* 48 points includes the doubling for downsampling, so the maximum number of
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* base sample points is 24, which is 23rd order.
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*/
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#define BSINC_POINTS_MAX (48)
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static double MinDouble(double a, double b)
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{ return (a <= b) ? a : b; }
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static double MaxDouble(double a, double b)
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{ return (a >= b) ? a : b; }
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/* NOTE: This is the normalized (instead of just sin(x)/x) cardinal sine
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* function.
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* 2 f_t sinc(2 f_t x)
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* f_t -- normalized transition frequency (0.5 is nyquist)
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* x -- sample index (-N to N)
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*/
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static double Sinc(const double x)
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{
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if(fabs(x) < 1e-15)
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return 1.0;
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return sin(M_PI * x) / (M_PI * x);
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}
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static double BesselI_0(const double x)
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{
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double term, sum, last_sum, x2, y;
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int i;
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term = 1.0;
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sum = 1.0;
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x2 = x / 2.0;
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i = 1;
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do {
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y = x2 / i;
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i++;
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last_sum = sum;
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term *= y * y;
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sum += term;
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} while(sum != last_sum);
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return sum;
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}
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/* NOTE: k is assumed normalized (-1 to 1)
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* beta is equivalent to 2 alpha
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*/
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static double Kaiser(const double b, const double k)
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{
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if(!(k >= -1.0 && k <= 1.0))
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return 0.0;
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return BesselI_0(b * sqrt(1.0 - k*k)) / BesselI_0(b);
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}
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/* Calculates the (normalized frequency) transition width of the Kaiser window.
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* Rejection is in dB.
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*/
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static double CalcKaiserWidth(const double rejection, const int order)
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{
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double w_t = 2.0 * M_PI;
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if(rejection > 21.0)
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return (rejection - 7.95) / (order * 2.285 * w_t);
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/* This enforces a minimum rejection of just above 21.18dB */
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return 5.79 / (order * w_t);
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}
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static double CalcKaiserBeta(const double rejection)
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{
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if(rejection > 50.0)
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return 0.1102 * (rejection - 8.7);
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else if(rejection >= 21.0)
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return (0.5842 * pow(rejection - 21.0, 0.4)) +
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(0.07886 * (rejection - 21.0));
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return 0.0;
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}
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/* Generates the coefficient, delta, and index tables required by the bsinc resampler */
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static void BsiGenerateTables(FILE *output, const char *tabname, const double rejection, const int order)
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{
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static double filter[BSINC_SCALE_COUNT][BSINC_PHASE_COUNT + 1][BSINC_POINTS_MAX];
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static double scDeltas[BSINC_SCALE_COUNT][BSINC_PHASE_COUNT ][BSINC_POINTS_MAX];
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static double phDeltas[BSINC_SCALE_COUNT][BSINC_PHASE_COUNT + 1][BSINC_POINTS_MAX];
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static double spDeltas[BSINC_SCALE_COUNT][BSINC_PHASE_COUNT ][BSINC_POINTS_MAX];
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static int mt[BSINC_SCALE_COUNT];
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static double at[BSINC_SCALE_COUNT];
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const int num_points_min = order + 1;
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double width, beta, scaleBase, scaleRange;
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int si, pi, i;
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memset(filter, 0, sizeof(filter));
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memset(scDeltas, 0, sizeof(scDeltas));
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memset(phDeltas, 0, sizeof(phDeltas));
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memset(spDeltas, 0, sizeof(spDeltas));
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/* Calculate windowing parameters. The width describes the transition
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band, but it may vary due to the linear interpolation between scales
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of the filter.
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*/
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width = CalcKaiserWidth(rejection, order);
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beta = CalcKaiserBeta(rejection);
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scaleBase = width / 2.0;
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scaleRange = 1.0 - scaleBase;
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// Determine filter scaling.
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for(si = 0; si < BSINC_SCALE_COUNT; si++)
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{
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const double scale = scaleBase + (scaleRange * si / (BSINC_SCALE_COUNT - 1));
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const double a = MinDouble(floor(num_points_min / (2.0 * scale)), num_points_min);
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const int m = 2 * (int)a;
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mt[si] = m;
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at[si] = a;
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}
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/* Calculate the Kaiser-windowed Sinc filter coefficients for each scale
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and phase.
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*/
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for(si = 0; si < BSINC_SCALE_COUNT; si++)
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{
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const int m = mt[si];
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const int o = num_points_min - (m / 2);
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const int l = (m / 2) - 1;
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const double a = at[si];
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const double scale = scaleBase + (scaleRange * si / (BSINC_SCALE_COUNT - 1));
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const double cutoff = (0.5 * scale) - (scaleBase * MaxDouble(0.5, scale));
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for(pi = 0; pi <= BSINC_PHASE_COUNT; pi++)
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{
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const double phase = l + ((double)pi / BSINC_PHASE_COUNT);
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for(i = 0; i < m; i++)
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{
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const double x = i - phase;
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filter[si][pi][o + i] = Kaiser(beta, x / a) * 2.0 * cutoff * Sinc(2.0 * cutoff * x);
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}
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}
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}
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/* Linear interpolation between scales is simplified by pre-calculating
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the delta (b - a) in: x = a + f (b - a)
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Given a difference in points between scales, the destination points
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will be 0, thus: x = a + f (-a)
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*/
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for(si = 0; si < (BSINC_SCALE_COUNT - 1); si++)
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{
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const int m = mt[si];
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const int o = num_points_min - (m / 2);
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for(pi = 0; pi < BSINC_PHASE_COUNT; pi++)
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{
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for(i = 0; i < m; i++)
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scDeltas[si][pi][o + i] = filter[si + 1][pi][o + i] - filter[si][pi][o + i];
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}
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}
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// Linear interpolation between phases is also simplified.
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for(si = 0; si < BSINC_SCALE_COUNT; si++)
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{
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const int m = mt[si];
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const int o = num_points_min - (m / 2);
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for(pi = 0; pi < BSINC_PHASE_COUNT; pi++)
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{
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for(i = 0; i < m; i++)
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phDeltas[si][pi][o + i] = filter[si][pi + 1][o + i] - filter[si][pi][o + i];
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}
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}
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/* This last simplification is done to complete the bilinear equation for
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the combination of scale and phase.
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*/
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for(si = 0; si < (BSINC_SCALE_COUNT - 1); si++)
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{
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const int m = mt[si];
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const int o = num_points_min - (m / 2);
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for(pi = 0; pi < BSINC_PHASE_COUNT; pi++)
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{
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for(i = 0; i < m; i++)
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spDeltas[si][pi][o + i] = phDeltas[si + 1][pi][o + i] - phDeltas[si][pi][o + i];
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}
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}
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// Make sure the number of points is a multiple of 4 (for SIMD).
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for(si = 0; si < BSINC_SCALE_COUNT; si++)
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mt[si] = (mt[si]+3) & ~3;
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// Calculate the table size.
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i = 0;
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for(si = 0; si < BSINC_SCALE_COUNT; si++)
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i += 4 * BSINC_PHASE_COUNT * mt[si];
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fprintf(output,
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"/* This %d%s order filter has a rejection of -%.0fdB, yielding a transition width\n"
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" * of ~%.3f (normalized frequency). Order increases when downsampling to a\n"
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" * limit of one octave, after which the quality of the filter (transition\n"
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" * width) suffers to reduce the CPU cost. The bandlimiting will cut all sound\n"
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" * after downsampling by ~%.2f octaves.\n"
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" */\n"
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"alignas(16) static constexpr float %s_tab[%d] = {\n",
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order, (((order%100)/10) == 1) ? "th" :
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((order%10) == 1) ? "st" :
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((order%10) == 2) ? "nd" :
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((order%10) == 3) ? "rd" : "th",
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rejection, width, log2(1.0/scaleBase), tabname, i);
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for(si = 0; si < BSINC_SCALE_COUNT; si++)
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{
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const int m = mt[si];
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const int o = num_points_min - (m / 2);
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for(pi = 0; pi < BSINC_PHASE_COUNT; pi++)
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{
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fprintf(output, " /* %2d,%2d (%d) */", si, pi, m);
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fprintf(output, "\n ");
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for(i = 0; i < m; i++)
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fprintf(output, " %+14.9ef,", filter[si][pi][o + i]);
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fprintf(output, "\n ");
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for(i = 0; i < m; i++)
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fprintf(output, " %+14.9ef,", scDeltas[si][pi][o + i]);
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fprintf(output, "\n ");
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for(i = 0; i < m; i++)
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fprintf(output, " %+14.9ef,", phDeltas[si][pi][o + i]);
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fprintf(output, "\n ");
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for(i = 0; i < m; i++)
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fprintf(output, " %+14.9ef,", spDeltas[si][pi][o + i]);
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fprintf(output, "\n");
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}
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}
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fprintf(output, "};\nconst BSincTable %s = {\n", tabname);
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/* The scaleBase is calculated from the Kaiser window transition width.
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It represents the absolute limit to the filter before it fully cuts
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the signal. The limit in octaves can be calculated by taking the
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base-2 logarithm of its inverse: log_2(1 / scaleBase)
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*/
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fprintf(output, " /* scaleBase */ %.9ef, /* scaleRange */ %.9ef,\n", scaleBase, 1.0 / scaleRange);
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fprintf(output, " /* m */ {");
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fprintf(output, " %d", mt[0]);
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for(si = 1; si < BSINC_SCALE_COUNT; si++)
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fprintf(output, ", %d", mt[si]);
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fprintf(output, " },\n");
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fprintf(output, " /* filterOffset */ {");
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fprintf(output, " %d", 0);
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i = mt[0]*4*BSINC_PHASE_COUNT;
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for(si = 1; si < BSINC_SCALE_COUNT; si++)
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{
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fprintf(output, ", %d", i);
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i += mt[si]*4*BSINC_PHASE_COUNT;
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}
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fprintf(output, " },\n");
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fprintf(output, " %s_tab\n", tabname);
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fprintf(output, "};\n\n");
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}
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int main(int argc, char *argv[])
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{
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FILE *output;
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GET_UNICODE_ARGS(&argc, &argv);
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if(argc > 2)
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{
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fprintf(stderr, "Usage: %s [output file]\n", argv[0]);
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return 1;
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}
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if(argc == 2)
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{
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output = fopen(argv[1], "wb");
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if(!output)
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{
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fprintf(stderr, "Failed to open %s for writing\n", argv[1]);
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return 1;
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}
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}
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else
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output = stdout;
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fprintf(output, "/* Generated by bsincgen, do not edit! */\n\n"
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"static_assert(BSINC_SCALE_COUNT == %d, \"Unexpected BSINC_SCALE_COUNT value!\");\n"
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"static_assert(BSINC_PHASE_COUNT == %d, \"Unexpected BSINC_PHASE_COUNT value!\");\n"
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"static_assert(FRACTIONONE == %d, \"Unexpected FRACTIONONE value!\");\n\n"
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"struct BSincTable {\n"
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" const float scaleBase, scaleRange;\n"
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" const int m[BSINC_SCALE_COUNT];\n"
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" const int filterOffset[BSINC_SCALE_COUNT];\n"
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" const float *Tab;\n"
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"};\n\n", BSINC_SCALE_COUNT, BSINC_PHASE_COUNT, FRACTIONONE);
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/* A 23rd order filter with a -60dB drop at nyquist. */
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BsiGenerateTables(output, "bsinc24", 60.0, 23);
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/* An 11th order filter with a -40dB drop at nyquist. */
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BsiGenerateTables(output, "bsinc12", 40.0, 11);
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if(output != stdout)
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fclose(output);
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output = NULL;
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return 0;
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}
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