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d2c156224b
Now there is only one single entry point for both, instead of previously 2 entry and 4 exit points. This also eliminates the explicit shutdown of ZMusic. Timidity++'s two buffers have been put in containers that self-destruct on shutdown and calling dumb_exit is not necessary because the only feature requiring it is not used by any code in the music library.
970 lines
No EOL
23 KiB
C++
970 lines
No EOL
23 KiB
C++
/*
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TiMidity++ -- MIDI to WAVE converter and player
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Copyright (C) 1999-2002 Masanao Izumo <mo@goice.co.jp>
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Copyright (C) 1995 Tuukka Toivonen <tt@cgs.fi>
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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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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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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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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resample.c
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*/
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#include <math.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include "timidity.h"
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#include "common.h"
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#include "instrum.h"
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#include "playmidi.h"
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#include "tables.h"
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#include "resample.h"
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#include "recache.h"
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namespace TimidityPlus
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{
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/* for start/end of samples */
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static float newt_coeffs[58][58];
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static int sample_bounds_min, sample_bounds_max; /* min/max bounds for sample data */
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#define DEFAULT_GAUSS_ORDER 25
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std::vector<float> gauss_table_data;
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static float *gauss_table[(1 << FRACTION_BITS)] = { 0 }; /* don't need doubles */
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static int gauss_n = DEFAULT_GAUSS_ORDER;
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static void initialize_newton_coeffs()
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{
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int i, j, n = 57;
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int sign;
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newt_coeffs[0][0] = 1;
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for (i = 0; i <= n; i++)
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{
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newt_coeffs[i][0] = 1;
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newt_coeffs[i][i] = 1;
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if (i > 1)
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{
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newt_coeffs[i][0] = newt_coeffs[i - 1][0] / i;
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newt_coeffs[i][i] = newt_coeffs[i - 1][0] / i;
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}
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for (j = 1; j < i; j++)
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{
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newt_coeffs[i][j] = newt_coeffs[i - 1][j - 1] + newt_coeffs[i - 1][j];
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if (i > 1)
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newt_coeffs[i][j] /= i;
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}
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}
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for (i = 0; i <= n; i++)
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for (j = 0, sign = pow(-1, i); j <= i; j++, sign *= -1)
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newt_coeffs[i][j] *= sign;
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}
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/* Very fast and accurate table based interpolation. Better speed and higher
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accuracy than Newton. This isn't *quite* true Gauss interpolation; it's
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more a slightly modified Gauss interpolation that I accidently stumbled
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upon. Rather than normalize all x values in the window to be in the range
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[0 to 2*PI], it simply divides them all by 2*PI instead. I don't know why
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this works, but it does. Gauss should only work on periodic data with the
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window spanning exactly one period, so it is no surprise that regular Gauss
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interpolation doesn't work too well on general audio data. But dividing
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the x values by 2*PI magically does. Any other scaling produces degraded
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results or total garbage. If anyone can work out the theory behind why
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this works so well (at first glance, it shouldn't ??), please contact me
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(Eric A. Welsh, ewelsh@ccb.wustl.edu), as I would really like to have some
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mathematical justification for doing this. Despite the lack of any sound
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theoretical basis, this method DOES result in highly accurate interpolation
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(or possibly approximaton, not sure yet if it truly interpolates, but it
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looks like it does). -N 34 is as high as it can go before errors start
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appearing. But even at -N 34, it is more accurate than Newton at -N 57.
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-N 34 has no problem running in realtime on my system, but -N 25 is the
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default, since that is the optimal compromise between speed and accuracy.
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I strongly recommend using Gauss interpolation. It is the highest
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quality interpolation option available, and is much faster than using
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Newton polynomials. */
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static resample_t resample_gauss(sample_t *src, splen_t ofs, resample_rec_t *rec)
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{
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sample_t *sptr;
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int32_t left, right, temp_n;
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left = (ofs >> FRACTION_BITS);
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right = (rec->data_length >> FRACTION_BITS) - left - 1;
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temp_n = (right << 1) - 1;
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if (temp_n > (left << 1) + 1)
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temp_n = (left << 1) + 1;
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if (temp_n < gauss_n) {
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int ii, jj;
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float xd, y;
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if (temp_n <= 0)
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temp_n = 1;
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xd = ofs & FRACTION_MASK;
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xd /= (1L << FRACTION_BITS);
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xd += temp_n >> 1;
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y = 0;
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sptr = src + (ofs >> FRACTION_BITS) - (temp_n >> 1);
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for (ii = temp_n; ii;) {
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for (jj = 0; jj <= ii; jj++)
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y += sptr[jj] * newt_coeffs[ii][jj];
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y *= xd - --ii;
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}
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y += *sptr;
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return ((y > sample_bounds_max) ? sample_bounds_max :
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((y < sample_bounds_min) ? sample_bounds_min : y));
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}
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else {
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float *gptr, *gend;
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float y;
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y = 0;
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sptr = src + left - (gauss_n >> 1);
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gptr = gauss_table[ofs&FRACTION_MASK];
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if (gauss_n == DEFAULT_GAUSS_ORDER) {
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/* expanding the loop for the default case.
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* this will allow intensive optimization when compiled
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* with SSE2 capability.
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*/
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#define do_gauss y += *(sptr++) * *(gptr++);
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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do_gauss;
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y += *sptr * *gptr;
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#undef do_gauss
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}
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else {
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gend = gptr + gauss_n;
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do {
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y += *(sptr++) * *(gptr++);
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} while (gptr <= gend);
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}
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return ((y > sample_bounds_max) ? sample_bounds_max :
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((y < sample_bounds_min) ? sample_bounds_min : y));
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}
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}
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#define RESAMPLATION *dest++ = resample_gauss(src, ofs, &resrc);
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/* exported for recache.c */
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resample_t do_resamplation(sample_t *src, splen_t ofs, resample_rec_t *rec)
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{
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return resample_gauss(src, ofs, rec);
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}
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#define PRECALC_LOOP_COUNT(start, end, incr) (int32_t)(((int64_t)((end) - (start) + (incr) - 1)) / (incr))
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void initialize_gauss_table(int n)
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{
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int m, i, k, n_half = (n >> 1);
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double ck;
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double x, x_inc, xz;
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double z[35], zsin_[34 + 35], *zsin, xzsin[35];
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float *gptr;
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for (i = 0; i <= n; i++)
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z[i] = i / (4 * M_PI);
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zsin = &zsin_[34];
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for (i = -n; i <= n; i++)
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zsin[i] = sin(i / (4 * M_PI));
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x_inc = 1.0 / (1 << FRACTION_BITS);
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gauss_table_data.resize((n + 1) * sizeof(float) * (1 << FRACTION_BITS));
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gptr = gauss_table_data.data();
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for (m = 0, x = 0.0; m < (1 << FRACTION_BITS); m++, x += x_inc)
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{
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xz = (x + n_half) / (4 * M_PI);
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for (i = 0; i <= n; i++)
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xzsin[i] = sin(xz - z[i]);
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gauss_table[m] = gptr;
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for (k = 0; k <= n; k++)
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{
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ck = 1.0;
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for (i = 0; i <= n; i++)
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{
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if (i == k)
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continue;
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ck *= xzsin[i] / zsin[k - i];
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}
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*gptr++ = ck;
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}
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}
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}
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void free_gauss_table(void)
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{
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if (gauss_table[0] != 0)
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free(gauss_table[0]);
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gauss_table[0] = NULL;
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}
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/* initialize the coefficients of the current resampling algorithm */
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void initialize_resampler_coeffs(void)
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{
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// Only needs to be done once.
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static bool done = false;
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if (done) return;
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done = true;
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initialize_newton_coeffs();
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initialize_gauss_table(gauss_n);
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sample_bounds_min = -32768;
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sample_bounds_max = 32767;
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}
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/*************** resampling with fixed increment *****************/
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resample_t *Resampler::rs_plain_c(int v, int32_t *countptr)
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{
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Voice *vp = &player->voice[v];
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resample_t *dest = resample_buffer + resample_buffer_offset;
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sample_t *src = vp->sample->data;
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int32_t ofs, count = *countptr, i, le;
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le = (int32_t)(vp->sample->loop_end >> FRACTION_BITS);
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ofs = (int32_t)(vp->sample_offset >> FRACTION_BITS);
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i = ofs + count;
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if (i > le)
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i = le;
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count = i - ofs;
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for (i = 0; i < count; i++) {
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dest[i] = src[i + ofs];
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}
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ofs += count;
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if (ofs == le)
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{
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vp->timeout = 1;
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*countptr = count;
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}
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vp->sample_offset = ((splen_t)ofs << FRACTION_BITS);
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return resample_buffer + resample_buffer_offset;
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}
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resample_t *Resampler::rs_plain(int v, int32_t *countptr)
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{
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/* Play sample until end, then free the voice. */
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Voice *vp = &player->voice[v];
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resample_t *dest = resample_buffer + resample_buffer_offset;
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sample_t *src = vp->sample->data;
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splen_t
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ofs = vp->sample_offset,
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ls = 0,
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le = vp->sample->data_length;
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resample_rec_t resrc;
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int32_t count = *countptr, incr = vp->sample_increment;
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int32_t i, j;
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if (vp->cache && incr == (1 << FRACTION_BITS))
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return rs_plain_c(v, countptr);
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resrc.loop_start = ls;
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resrc.loop_end = le;
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resrc.data_length = vp->sample->data_length;
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if (incr < 0) incr = -incr; /* In case we're coming out of a bidir loop */
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/* Precalc how many times we should go through the loop.
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NOTE: Assumes that incr > 0 and that ofs <= le */
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i = PRECALC_LOOP_COUNT(ofs, le, incr);
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if (i > count)
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{
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i = count;
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count = 0;
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}
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else count -= i;
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for (j = 0; j < i; j++)
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{
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RESAMPLATION;
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ofs += incr;
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}
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if (ofs >= le)
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{
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vp->timeout = 1;
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*countptr -= count;
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}
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vp->sample_offset = ofs; /* Update offset */
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return resample_buffer + resample_buffer_offset;
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}
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resample_t *Resampler::rs_loop_c(Voice *vp, int32_t count)
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{
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int32_t
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ofs = (int32_t)(vp->sample_offset >> FRACTION_BITS),
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le = (int32_t)(vp->sample->loop_end >> FRACTION_BITS),
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ll = le - (int32_t)(vp->sample->loop_start >> FRACTION_BITS);
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resample_t *dest = resample_buffer + resample_buffer_offset;
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sample_t *src = vp->sample->data;
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int32_t i, j;
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while (count)
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{
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while (ofs >= le)
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ofs -= ll;
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/* Precalc how many times we should go through the loop */
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i = le - ofs;
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if (i > count)
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i = count;
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count -= i;
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for (j = 0; j < i; j++) {
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dest[j] = src[j + ofs];
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}
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dest += i;
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ofs += i;
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}
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vp->sample_offset = ((splen_t)ofs << FRACTION_BITS);
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return resample_buffer + resample_buffer_offset;
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}
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resample_t *Resampler::rs_loop(Voice *vp, int32_t count)
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{
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/* Play sample until end-of-loop, skip back and continue. */
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splen_t
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ofs = vp->sample_offset,
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ls, le, ll;
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resample_rec_t resrc;
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resample_t *dest = resample_buffer + resample_buffer_offset;
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sample_t *src = vp->sample->data;
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int32_t i, j;
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int32_t incr = vp->sample_increment;
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if (vp->cache && incr == (1 << FRACTION_BITS))
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return rs_loop_c(vp, count);
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resrc.loop_start = ls = vp->sample->loop_start;
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resrc.loop_end = le = vp->sample->loop_end;
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ll = le - ls;
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resrc.data_length = vp->sample->data_length;
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while (count)
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{
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while (ofs >= le) { ofs -= ll; }
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/* Precalc how many times we should go through the loop */
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i = PRECALC_LOOP_COUNT(ofs, le, incr);
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if (i > count) {
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i = count;
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count = 0;
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}
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else { count -= i; }
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for (j = 0; j < i; j++) {
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RESAMPLATION;
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ofs += incr;
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}
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}
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vp->sample_offset = ofs; /* Update offset */
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return resample_buffer + resample_buffer_offset;
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}
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resample_t *Resampler::rs_bidir(Voice *vp, int32_t count)
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{
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int32_t
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ofs = vp->sample_offset,
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le = vp->sample->loop_end,
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ls = vp->sample->loop_start;
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resample_t *dest = resample_buffer + resample_buffer_offset;
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sample_t *src = vp->sample->data;
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int32_t incr = vp->sample_increment;
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resample_rec_t resrc;
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int32_t
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le2 = le << 1,
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ls2 = ls << 1;
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int32_t i, j;
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/* Play normally until inside the loop region */
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resrc.loop_start = ls;
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resrc.loop_end = le;
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resrc.data_length = vp->sample->data_length;
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if (incr > 0 && ofs < ls)
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{
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/* NOTE: Assumes that incr > 0, which is NOT always the case
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when doing bidirectional looping. I have yet to see a case
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where both ofs <= ls AND incr < 0, however. */
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i = PRECALC_LOOP_COUNT(ofs, ls, incr);
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if (i > count)
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{
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i = count;
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count = 0;
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}
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else count -= i;
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for (j = 0; j < i; j++)
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{
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RESAMPLATION;
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ofs += incr;
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}
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}
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/* Then do the bidirectional looping */
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while (count)
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{
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/* Precalc how many times we should go through the loop */
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i = PRECALC_LOOP_COUNT(ofs, incr > 0 ? le : ls, incr);
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if (i > count)
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{
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i = count;
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count = 0;
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}
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else count -= i;
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for (j = 0; j < i; j++)
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{
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RESAMPLATION;
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ofs += incr;
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}
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if (ofs >= 0 && ofs >= le)
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{
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/* fold the overshoot back in */
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ofs = le2 - ofs;
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incr *= -1;
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}
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else if (ofs <= 0 || ofs <= ls)
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{
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ofs = ls2 - ofs;
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incr *= -1;
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}
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}
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vp->sample_increment = incr;
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vp->sample_offset = ofs; /* Update offset */
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return resample_buffer + resample_buffer_offset;
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}
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/*********************** vibrato versions ***************************/
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/* We only need to compute one half of the vibrato sine cycle */
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static int vib_phase_to_inc_ptr(int phase)
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{
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if (phase < VIBRATO_SAMPLE_INCREMENTS / 2)
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return VIBRATO_SAMPLE_INCREMENTS / 2 - 1 - phase;
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else if (phase >= 3 * VIBRATO_SAMPLE_INCREMENTS / 2)
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return 5 * VIBRATO_SAMPLE_INCREMENTS / 2 - 1 - phase;
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else
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return phase - VIBRATO_SAMPLE_INCREMENTS / 2;
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}
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int32_t Resampler::update_vibrato(Voice *vp, int sign)
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{
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int32_t depth;
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|
int phase, pb;
|
|
double a;
|
|
int ch = vp->channel;
|
|
|
|
if (vp->vibrato_delay > 0)
|
|
{
|
|
vp->vibrato_delay -= vp->vibrato_control_ratio;
|
|
if (vp->vibrato_delay > 0)
|
|
return vp->sample_increment;
|
|
}
|
|
|
|
if (vp->vibrato_phase++ >= 2 * VIBRATO_SAMPLE_INCREMENTS - 1)
|
|
vp->vibrato_phase = 0;
|
|
phase = vib_phase_to_inc_ptr(vp->vibrato_phase);
|
|
|
|
if (vp->vibrato_sample_increment[phase])
|
|
{
|
|
if (sign)
|
|
return -vp->vibrato_sample_increment[phase];
|
|
else
|
|
return vp->vibrato_sample_increment[phase];
|
|
}
|
|
|
|
/* Need to compute this sample increment. */
|
|
|
|
depth = vp->vibrato_depth;
|
|
depth <<= 7;
|
|
|
|
if (vp->vibrato_sweep && !player->channel[ch].mod.val)
|
|
{
|
|
/* Need to update sweep */
|
|
vp->vibrato_sweep_position += vp->vibrato_sweep;
|
|
if (vp->vibrato_sweep_position >= (1 << SWEEP_SHIFT))
|
|
vp->vibrato_sweep = 0;
|
|
else
|
|
{
|
|
/* Adjust depth */
|
|
depth *= vp->vibrato_sweep_position;
|
|
depth >>= SWEEP_SHIFT;
|
|
}
|
|
}
|
|
|
|
if (vp->sample->inst_type == INST_SF2) {
|
|
pb = (int)((lookup_triangular(vp->vibrato_phase *
|
|
(SINE_CYCLE_LENGTH / (2 * VIBRATO_SAMPLE_INCREMENTS)))
|
|
* (double)(depth)* VIBRATO_AMPLITUDE_TUNING));
|
|
}
|
|
else {
|
|
pb = (int)((lookup_sine(vp->vibrato_phase *
|
|
(SINE_CYCLE_LENGTH / (2 * VIBRATO_SAMPLE_INCREMENTS)))
|
|
* (double)(depth)* VIBRATO_AMPLITUDE_TUNING));
|
|
}
|
|
|
|
a = TIM_FSCALE(((double)(vp->sample->sample_rate) *
|
|
(double)(vp->frequency)) /
|
|
((double)(vp->sample->root_freq) *
|
|
(double)(playback_rate)),
|
|
FRACTION_BITS);
|
|
|
|
if (pb < 0) {
|
|
pb = -pb;
|
|
a /= bend_fine[(pb >> 5) & 0xFF] * bend_coarse[pb >> 13];
|
|
pb = -pb;
|
|
}
|
|
else {
|
|
a *= bend_fine[(pb >> 5) & 0xFF] * bend_coarse[pb >> 13];
|
|
}
|
|
a += 0.5;
|
|
|
|
/* If the sweep's over, we can store the newly computed sample_increment */
|
|
if (!vp->vibrato_sweep || player->channel[ch].mod.val)
|
|
vp->vibrato_sample_increment[phase] = (int32_t)a;
|
|
|
|
if (sign)
|
|
a = -a; /* need to preserve the loop direction */
|
|
|
|
return (int32_t)a;
|
|
}
|
|
|
|
resample_t *Resampler::rs_vib_plain(int v, int32_t *countptr)
|
|
{
|
|
/* Play sample until end, then free the voice. */
|
|
Voice *vp = &player->voice[v];
|
|
resample_t *dest = resample_buffer + resample_buffer_offset;
|
|
sample_t *src = vp->sample->data;
|
|
splen_t
|
|
ls = 0,
|
|
le = vp->sample->data_length,
|
|
ofs = vp->sample_offset;
|
|
resample_rec_t resrc;
|
|
|
|
int32_t count = *countptr, incr = vp->sample_increment;
|
|
int cc = vp->vibrato_control_counter;
|
|
|
|
resrc.loop_start = ls;
|
|
resrc.loop_end = le;
|
|
resrc.data_length = vp->sample->data_length;
|
|
/* This has never been tested */
|
|
|
|
if (incr < 0) incr = -incr; /* In case we're coming out of a bidir loop */
|
|
|
|
while (count--)
|
|
{
|
|
if (!cc--)
|
|
{
|
|
cc = vp->vibrato_control_ratio;
|
|
incr = update_vibrato(vp, 0);
|
|
}
|
|
RESAMPLATION;
|
|
ofs += incr;
|
|
if (ofs >= le)
|
|
{
|
|
vp->timeout = 1;
|
|
*countptr -= count;
|
|
break;
|
|
}
|
|
}
|
|
|
|
vp->vibrato_control_counter = cc;
|
|
vp->sample_increment = incr;
|
|
vp->sample_offset = ofs; /* Update offset */
|
|
return resample_buffer + resample_buffer_offset;
|
|
}
|
|
|
|
resample_t *Resampler::rs_vib_loop(Voice *vp, int32_t count)
|
|
{
|
|
/* Play sample until end-of-loop, skip back and continue. */
|
|
splen_t
|
|
ofs = vp->sample_offset,
|
|
ls = vp->sample->loop_start,
|
|
le = vp->sample->loop_end,
|
|
ll = le - vp->sample->loop_start;
|
|
resample_t *dest = resample_buffer + resample_buffer_offset;
|
|
sample_t *src = vp->sample->data;
|
|
int cc = vp->vibrato_control_counter;
|
|
int32_t incr = vp->sample_increment;
|
|
resample_rec_t resrc;
|
|
int32_t i, j;
|
|
int vibflag = 0;
|
|
|
|
resrc.loop_start = ls;
|
|
resrc.loop_end = le;
|
|
resrc.data_length = vp->sample->data_length;
|
|
|
|
while (count)
|
|
{
|
|
/* Hopefully the loop is longer than an increment */
|
|
while (ofs >= le) { ofs -= ll; }
|
|
/* Precalc how many times to go through the loop, taking
|
|
the vibrato control ratio into account this time. */
|
|
i = PRECALC_LOOP_COUNT(ofs, le, incr);
|
|
if (i > count) {
|
|
i = count;
|
|
}
|
|
if (i > cc) {
|
|
i = cc;
|
|
vibflag = 1;
|
|
}
|
|
else { cc -= i; }
|
|
count -= i;
|
|
if (vibflag) {
|
|
cc = vp->vibrato_control_ratio;
|
|
incr = update_vibrato(vp, 0);
|
|
vibflag = 0;
|
|
}
|
|
for (j = 0; j < i; j++) {
|
|
RESAMPLATION;
|
|
ofs += incr;
|
|
}
|
|
}
|
|
|
|
vp->vibrato_control_counter = cc;
|
|
vp->sample_increment = incr;
|
|
vp->sample_offset = ofs; /* Update offset */
|
|
return resample_buffer + resample_buffer_offset;
|
|
}
|
|
|
|
resample_t *Resampler::rs_vib_bidir(Voice *vp, int32_t count)
|
|
{
|
|
int32_t
|
|
ofs = vp->sample_offset,
|
|
le = vp->sample->loop_end,
|
|
ls = vp->sample->loop_start;
|
|
resample_t *dest = resample_buffer + resample_buffer_offset;
|
|
sample_t *src = vp->sample->data;
|
|
int cc = vp->vibrato_control_counter;
|
|
int32_t incr = vp->sample_increment;
|
|
resample_rec_t resrc;
|
|
|
|
|
|
resrc.loop_start = ls;
|
|
resrc.loop_end = le;
|
|
resrc.data_length = vp->sample->data_length;
|
|
/* Play normally until inside the loop region */
|
|
|
|
if (ofs < ls)
|
|
{
|
|
while (count--)
|
|
{
|
|
if (!cc--)
|
|
{
|
|
cc = vp->vibrato_control_ratio;
|
|
incr = update_vibrato(vp, 0);
|
|
}
|
|
RESAMPLATION;
|
|
ofs += incr;
|
|
if (ofs >= ls)
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* Then do the bidirectional looping */
|
|
|
|
if (count > 0)
|
|
while (count--)
|
|
{
|
|
if (!cc--)
|
|
{
|
|
cc = vp->vibrato_control_ratio;
|
|
incr = update_vibrato(vp, (incr < 0));
|
|
}
|
|
RESAMPLATION;
|
|
ofs += incr;
|
|
if (ofs >= le)
|
|
{
|
|
/* fold the overshoot back in */
|
|
ofs = le - (ofs - le);
|
|
incr = -incr;
|
|
}
|
|
else if (ofs <= ls)
|
|
{
|
|
ofs = ls + (ls - ofs);
|
|
incr = -incr;
|
|
}
|
|
}
|
|
|
|
/* Update changed values */
|
|
vp->vibrato_control_counter = cc;
|
|
vp->sample_increment = incr;
|
|
vp->sample_offset = ofs;
|
|
return resample_buffer + resample_buffer_offset;
|
|
}
|
|
|
|
/*********************** portamento versions ***************************/
|
|
|
|
int Resampler::rs_update_porta(int v)
|
|
{
|
|
Voice *vp = &player->voice[v];
|
|
int32_t d;
|
|
|
|
d = vp->porta_dpb;
|
|
if (vp->porta_pb < 0)
|
|
{
|
|
if (d > -vp->porta_pb)
|
|
d = -vp->porta_pb;
|
|
}
|
|
else
|
|
{
|
|
if (d > vp->porta_pb)
|
|
d = -vp->porta_pb;
|
|
else
|
|
d = -d;
|
|
}
|
|
|
|
vp->porta_pb += d;
|
|
if (vp->porta_pb == 0)
|
|
{
|
|
vp->porta_control_ratio = 0;
|
|
vp->porta_pb = 0;
|
|
}
|
|
player->recompute_freq(v);
|
|
return vp->porta_control_ratio;
|
|
}
|
|
|
|
resample_t *Resampler::porta_resample_voice(int v, int32_t *countptr, int mode)
|
|
{
|
|
Voice *vp = &player->voice[v];
|
|
int32_t n = *countptr, i;
|
|
resample_t *(Resampler::*resampler)(int, int32_t *, int);
|
|
int cc = vp->porta_control_counter;
|
|
int loop;
|
|
|
|
if (vp->vibrato_control_ratio)
|
|
resampler = &Resampler::vib_resample_voice;
|
|
else
|
|
resampler = &Resampler::normal_resample_voice;
|
|
if (mode != 1)
|
|
loop = 1;
|
|
else
|
|
loop = 0;
|
|
|
|
vp->cache = NULL;
|
|
resample_buffer_offset = 0;
|
|
while (resample_buffer_offset < n)
|
|
{
|
|
if (cc == 0)
|
|
{
|
|
if ((cc = rs_update_porta(v)) == 0)
|
|
{
|
|
i = n - resample_buffer_offset;
|
|
(this->*resampler)(v, &i, mode);
|
|
resample_buffer_offset += i;
|
|
break;
|
|
}
|
|
}
|
|
|
|
i = n - resample_buffer_offset;
|
|
if (i > cc)
|
|
i = cc;
|
|
(this->*resampler)(v, &i, mode);
|
|
resample_buffer_offset += i;
|
|
|
|
if (!loop && (i == 0 || vp->status == VOICE_FREE))
|
|
break;
|
|
cc -= i;
|
|
}
|
|
*countptr = resample_buffer_offset;
|
|
resample_buffer_offset = 0;
|
|
vp->porta_control_counter = cc;
|
|
return resample_buffer;
|
|
}
|
|
|
|
/* interface function */
|
|
resample_t *Resampler::vib_resample_voice(int v, int32_t *countptr, int mode)
|
|
{
|
|
Voice *vp = &player->voice[v];
|
|
|
|
vp->cache = NULL;
|
|
if (mode == 0)
|
|
return rs_vib_loop(vp, *countptr);
|
|
if (mode == 1)
|
|
return rs_vib_plain(v, countptr);
|
|
return rs_vib_bidir(vp, *countptr);
|
|
}
|
|
|
|
/* interface function */
|
|
resample_t *Resampler::normal_resample_voice(int v, int32_t *countptr, int mode)
|
|
{
|
|
Voice *vp = &player->voice[v];
|
|
if (mode == 0)
|
|
return rs_loop(vp, *countptr);
|
|
if (mode == 1)
|
|
return rs_plain(v, countptr);
|
|
return rs_bidir(vp, *countptr);
|
|
}
|
|
|
|
/* interface function */
|
|
resample_t *Resampler::resample_voice(int v, int32_t *countptr)
|
|
{
|
|
Voice *vp = &player->voice[v];
|
|
int mode;
|
|
resample_t *result;
|
|
int32_t i;
|
|
|
|
if (vp->sample->sample_rate == playback_rate &&
|
|
vp->sample->root_freq == get_note_freq(vp->sample, vp->sample->note_to_use) &&
|
|
vp->frequency == vp->orig_frequency)
|
|
{
|
|
int32_t ofs;
|
|
|
|
/* Pre-resampled data -- just update the offset and check if
|
|
we're out of data. */
|
|
ofs = (int32_t)(vp->sample_offset >> FRACTION_BITS); /* Kind of silly to use
|
|
FRACTION_BITS here... */
|
|
if (*countptr >= (int32_t)((vp->sample->data_length >> FRACTION_BITS) - ofs))
|
|
{
|
|
/* Note finished. Free the voice. */
|
|
vp->timeout = 1;
|
|
|
|
/* Let the caller know how much data we had left */
|
|
*countptr = (int32_t)(vp->sample->data_length >> FRACTION_BITS) - ofs;
|
|
}
|
|
else
|
|
vp->sample_offset += *countptr << FRACTION_BITS;
|
|
|
|
for (i = 0; i < *countptr; i++) {
|
|
resample_buffer[i] = vp->sample->data[i + ofs];
|
|
}
|
|
return resample_buffer;
|
|
}
|
|
|
|
mode = vp->sample->modes;
|
|
if ((mode & MODES_LOOPING) &&
|
|
((mode & MODES_ENVELOPE) ||
|
|
(vp->status & (VOICE_ON | VOICE_SUSTAINED))))
|
|
{
|
|
if (mode & MODES_PINGPONG)
|
|
{
|
|
vp->cache = NULL;
|
|
mode = 2; /* Bidir loop */
|
|
}
|
|
else
|
|
mode = 0; /* loop */
|
|
}
|
|
else
|
|
mode = 1; /* no loop */
|
|
|
|
if (vp->porta_control_ratio)
|
|
result = porta_resample_voice(v, countptr, mode);
|
|
else if (vp->vibrato_control_ratio)
|
|
result = vib_resample_voice(v, countptr, mode);
|
|
else
|
|
result = normal_resample_voice(v, countptr, mode);
|
|
|
|
return result;
|
|
}
|
|
|
|
|
|
void pre_resample(Sample * sp)
|
|
{
|
|
double a, b;
|
|
splen_t ofs, newlen;
|
|
sample_t *newdata, *dest, *src = (sample_t *)sp->data;
|
|
int32_t i, count, incr, f, x;
|
|
resample_rec_t resrc;
|
|
|
|
f = get_note_freq(sp, sp->note_to_use);
|
|
a = b = ((double)(sp->root_freq) * playback_rate) /
|
|
((double)(sp->sample_rate) * f);
|
|
if ((int64_t)sp->data_length * a >= 0x7fffffffL)
|
|
{
|
|
/* Too large to compute */
|
|
printMessage(CMSG_INFO, VERB_DEBUG, " *** Can't pre-resampling for note %d",
|
|
sp->note_to_use);
|
|
return;
|
|
}
|
|
newlen = (splen_t)(sp->data_length * a);
|
|
count = (newlen >> FRACTION_BITS);
|
|
ofs = incr = (sp->data_length - 1) / (count - 1);
|
|
|
|
if ((double)newlen + incr >= 0x7fffffffL)
|
|
{
|
|
/* Too large to compute */
|
|
printMessage(CMSG_INFO, VERB_DEBUG, " *** Can't pre-resampling for note %d",
|
|
sp->note_to_use);
|
|
return;
|
|
}
|
|
|
|
// [EP] Fix the bad allocation count.
|
|
dest = newdata = (sample_t *)safe_malloc(((int32_t)(newlen >> (FRACTION_BITS - 1)) + 2)*sizeof(sample_t));
|
|
dest[newlen >> FRACTION_BITS] = 0;
|
|
|
|
*dest++ = src[0];
|
|
|
|
resrc.loop_start = 0;
|
|
resrc.loop_end = sp->data_length;
|
|
resrc.data_length = sp->data_length;
|
|
|
|
/* Since we're pre-processing and this doesn't have to be done in
|
|
real-time, we go ahead and do the higher order interpolation. */
|
|
for (i = 1; i < count; i++)
|
|
{
|
|
x = resample_gauss(src, ofs, &resrc);
|
|
*dest++ = (int16_t)((x > 32767) ? 32767 : ((x < -32768) ? -32768 : x));
|
|
ofs += incr;
|
|
}
|
|
|
|
sp->data_length = newlen;
|
|
sp->loop_start = (splen_t)(sp->loop_start * b);
|
|
sp->loop_end = (splen_t)(sp->loop_end * b);
|
|
free(sp->data);
|
|
sp->data = (sample_t *)newdata;
|
|
sp->root_freq = f;
|
|
sp->sample_rate = playback_rate;
|
|
sp->low_freq = freq_table[0];
|
|
sp->high_freq = freq_table[127];
|
|
}
|
|
|
|
} |