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