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diff --git a/libs/fluidsynth/src/fluid_rvoice_dsp.c b/libs/fluidsynth/src/fluid_rvoice_dsp.c
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+/* FluidSynth - A Software Synthesizer
+ *
+ * Copyright (C) 2003 Peter Hanappe and others.
+ *
+ * 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
+ */
+
+#include "fluidsynth_priv.h"
+#include "fluid_phase.h"
+#include "fluid_rvoice.h"
+#include "fluid_sys.h"
+
+/* Purpose:
+ *
+ * Interpolates audio data (obtains values between the samples of the original
+ * waveform data).
+ *
+ * Variables loaded from the voice structure (assigned in fluid_voice_write()):
+ * - dsp_data: Pointer to the original waveform data
+ * - dsp_phase: The position in the original waveform data.
+ * This has an integer and a fractional part (between samples).
+ * - dsp_phase_incr: For each output sample, the position in the original
+ * waveform advances by dsp_phase_incr. This also has an integer
+ * part and a fractional part.
+ * If a sample is played at root pitch (no pitch change),
+ * dsp_phase_incr is integer=1 and fractional=0.
+ * - dsp_amp: The current amplitude envelope value.
+ * - dsp_amp_incr: The changing rate of the amplitude envelope.
+ *
+ * A couple of variables are used internally, their results are discarded:
+ * - dsp_i: Index through the output buffer
+ * - dsp_buf: Output buffer of floating point values (FLUID_BUFSIZE in length)
+ */
+
+/* Interpolation (find a value between two samples of the original waveform) */
+
+/* Linear interpolation table (2 coefficients centered on 1st) */
+static fluid_real_t interp_coeff_linear[FLUID_INTERP_MAX][2];
+
+/* 4th order (cubic) interpolation table (4 coefficients centered on 2nd) */
+static fluid_real_t interp_coeff[FLUID_INTERP_MAX][4];
+
+/* 7th order interpolation (7 coefficients centered on 3rd) */
+static fluid_real_t sinc_table7[FLUID_INTERP_MAX][7];
+
+
+#define SINC_INTERP_ORDER 7 /* 7th order constant */
+
+
+/* Initializes interpolation tables */
+void fluid_rvoice_dsp_config (void)
+{
+ int i, i2;
+ double x, v;
+ double i_shifted;
+
+ /* Initialize the coefficients for the interpolation. The math comes
+ * from a mail, posted by Olli Niemitalo to the music-dsp mailing
+ * list (I found it in the music-dsp archives
+ * http://www.smartelectronix.com/musicdsp/). */
+
+ for (i = 0; i < FLUID_INTERP_MAX; i++)
+ {
+ x = (double) i / (double) FLUID_INTERP_MAX;
+
+ interp_coeff[i][0] = (fluid_real_t)(x * (-0.5 + x * (1 - 0.5 * x)));
+ interp_coeff[i][1] = (fluid_real_t)(1.0 + x * x * (1.5 * x - 2.5));
+ interp_coeff[i][2] = (fluid_real_t)(x * (0.5 + x * (2.0 - 1.5 * x)));
+ interp_coeff[i][3] = (fluid_real_t)(0.5 * x * x * (x - 1.0));
+
+ interp_coeff_linear[i][0] = (fluid_real_t)(1.0 - x);
+ interp_coeff_linear[i][1] = (fluid_real_t)x;
+ }
+
+ /* i: Offset in terms of whole samples */
+ for (i = 0; i < SINC_INTERP_ORDER; i++)
+ { /* i2: Offset in terms of fractional samples ('subsamples') */
+ for (i2 = 0; i2 < FLUID_INTERP_MAX; i2++)
+ {
+ /* center on middle of table */
+ i_shifted = (double)i - ((double)SINC_INTERP_ORDER / 2.0)
+ + (double)i2 / (double)FLUID_INTERP_MAX;
+
+ /* sinc(0) cannot be calculated straightforward (limit needed for 0/0) */
+ if (fabs (i_shifted) > 0.000001)
+ {
+ v = (fluid_real_t)sin (i_shifted * M_PI) / (M_PI * i_shifted);
+ /* Hamming window */
+ v *= (fluid_real_t)0.5 * (1.0 + cos (2.0 * M_PI * i_shifted / (fluid_real_t)SINC_INTERP_ORDER));
+ }
+ else v = 1.0;
+
+ sinc_table7[FLUID_INTERP_MAX - i2 - 1][i] = v;
+ }
+ }
+
+#if 0
+ for (i = 0; i < FLUID_INTERP_MAX; i++)
+ {
+ printf ("%d %0.3f %0.3f %0.3f %0.3f %0.3f %0.3f %0.3f\n",
+ i, sinc_table7[0][i], sinc_table7[1][i], sinc_table7[2][i],
+ sinc_table7[3][i], sinc_table7[4][i], sinc_table7[5][i], sinc_table7[6][i]);
+ }
+#endif
+
+ fluid_check_fpe("interpolation table calculation");
+}
+
+/* No interpolation. Just take the sample, which is closest to
+ * the playback pointer. Questionable quality, but very
+ * efficient. */
+int
+fluid_rvoice_dsp_interpolate_none (fluid_rvoice_dsp_t *voice)
+{
+ fluid_phase_t dsp_phase = voice->phase;
+ fluid_phase_t dsp_phase_incr;
+ short int *dsp_data = voice->sample->data;
+ fluid_real_t *dsp_buf = voice->dsp_buf;
+ fluid_real_t dsp_amp = voice->amp;
+ fluid_real_t dsp_amp_incr = voice->amp_incr;
+ unsigned int dsp_i = 0;
+ unsigned int dsp_phase_index;
+ unsigned int end_index;
+ int looping;
+
+ /* Convert playback "speed" floating point value to phase index/fract */
+ fluid_phase_set_float (dsp_phase_incr, voice->phase_incr);
+
+ /* voice is currently looping? */
+ looping = voice->is_looping;
+
+ end_index = looping ? voice->loopend - 1 : voice->end;
+
+ while (1)
+ {
+ dsp_phase_index = fluid_phase_index_round (dsp_phase); /* round to nearest point */
+
+ /* interpolate sequence of sample points */
+ for ( ; dsp_i < FLUID_BUFSIZE && dsp_phase_index <= end_index; dsp_i++)
+ {
+ dsp_buf[dsp_i] = dsp_amp * dsp_data[dsp_phase_index];
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index_round (dsp_phase); /* round to nearest point */
+ dsp_amp += dsp_amp_incr;
+ }
+
+ /* break out if not looping (buffer may not be full) */
+ if (!looping) break;
+
+ /* go back to loop start */
+ if (dsp_phase_index > end_index)
+ {
+ fluid_phase_sub_int (dsp_phase, voice->loopend - voice->loopstart);
+ voice->has_looped = 1;
+ }
+
+ /* break out if filled buffer */
+ if (dsp_i >= FLUID_BUFSIZE) break;
+ }
+
+ voice->phase = dsp_phase;
+ voice->amp = dsp_amp;
+
+ return (dsp_i);
+}
+
+/* Straight line interpolation.
+ * Returns number of samples processed (usually FLUID_BUFSIZE but could be
+ * smaller if end of sample occurs).
+ */
+int
+fluid_rvoice_dsp_interpolate_linear (fluid_rvoice_dsp_t *voice)
+{
+ fluid_phase_t dsp_phase = voice->phase;
+ fluid_phase_t dsp_phase_incr;
+ short int *dsp_data = voice->sample->data;
+ fluid_real_t *dsp_buf = voice->dsp_buf;
+ fluid_real_t dsp_amp = voice->amp;
+ fluid_real_t dsp_amp_incr = voice->amp_incr;
+ unsigned int dsp_i = 0;
+ unsigned int dsp_phase_index;
+ unsigned int end_index;
+ short int point;
+ fluid_real_t *coeffs;
+ int looping;
+
+ /* Convert playback "speed" floating point value to phase index/fract */
+ fluid_phase_set_float (dsp_phase_incr, voice->phase_incr);
+
+ /* voice is currently looping? */
+ looping = voice->is_looping;
+
+ /* last index before 2nd interpolation point must be specially handled */
+ end_index = (looping ? voice->loopend - 1 : voice->end) - 1;
+
+ /* 2nd interpolation point to use at end of loop or sample */
+ if (looping) point = dsp_data[voice->loopstart]; /* loop start */
+ else point = dsp_data[voice->end]; /* duplicate end for samples no longer looping */
+
+ while (1)
+ {
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+
+ /* interpolate the sequence of sample points */
+ for ( ; dsp_i < FLUID_BUFSIZE && dsp_phase_index <= end_index; dsp_i++)
+ {
+ coeffs = interp_coeff_linear[fluid_phase_fract_to_tablerow (dsp_phase)];
+ dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index]
+ + coeffs[1] * dsp_data[dsp_phase_index+1]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ /* break out if buffer filled */
+ if (dsp_i >= FLUID_BUFSIZE) break;
+
+ end_index++; /* we're now interpolating the last point */
+
+ /* interpolate within last point */
+ for (; dsp_phase_index <= end_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = interp_coeff_linear[fluid_phase_fract_to_tablerow (dsp_phase)];
+ dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index]
+ + coeffs[1] * point);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr; /* increment amplitude */
+ }
+
+ if (!looping) break; /* break out if not looping (end of sample) */
+
+ /* go back to loop start (if past */
+ if (dsp_phase_index > end_index)
+ {
+ fluid_phase_sub_int (dsp_phase, voice->loopend - voice->loopstart);
+ voice->has_looped = 1;
+ }
+
+ /* break out if filled buffer */
+ if (dsp_i >= FLUID_BUFSIZE) break;
+
+ end_index--; /* set end back to second to last sample point */
+ }
+
+ voice->phase = dsp_phase;
+ voice->amp = dsp_amp;
+
+ return (dsp_i);
+}
+
+/* 4th order (cubic) interpolation.
+ * Returns number of samples processed (usually FLUID_BUFSIZE but could be
+ * smaller if end of sample occurs).
+ */
+int
+fluid_rvoice_dsp_interpolate_4th_order (fluid_rvoice_dsp_t *voice)
+{
+ fluid_phase_t dsp_phase = voice->phase;
+ fluid_phase_t dsp_phase_incr;
+ short int *dsp_data = voice->sample->data;
+ fluid_real_t *dsp_buf = voice->dsp_buf;
+ fluid_real_t dsp_amp = voice->amp;
+ fluid_real_t dsp_amp_incr = voice->amp_incr;
+ unsigned int dsp_i = 0;
+ unsigned int dsp_phase_index;
+ unsigned int start_index, end_index;
+ short int start_point, end_point1, end_point2;
+ fluid_real_t *coeffs;
+ int looping;
+
+ /* Convert playback "speed" floating point value to phase index/fract */
+ fluid_phase_set_float (dsp_phase_incr, voice->phase_incr);
+
+ /* voice is currently looping? */
+ looping = voice->is_looping;
+
+ /* last index before 4th interpolation point must be specially handled */
+ end_index = (looping ? voice->loopend - 1 : voice->end) - 2;
+
+ if (voice->has_looped) /* set start_index and start point if looped or not */
+ {
+ start_index = voice->loopstart;
+ start_point = dsp_data[voice->loopend - 1]; /* last point in loop (wrap around) */
+ }
+ else
+ {
+ start_index = voice->start;
+ start_point = dsp_data[voice->start]; /* just duplicate the point */
+ }
+
+ /* get points off the end (loop start if looping, duplicate point if end) */
+ if (looping)
+ {
+ end_point1 = dsp_data[voice->loopstart];
+ end_point2 = dsp_data[voice->loopstart + 1];
+ }
+ else
+ {
+ end_point1 = dsp_data[voice->end];
+ end_point2 = end_point1;
+ }
+
+ while (1)
+ {
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+
+ /* interpolate first sample point (start or loop start) if needed */
+ for ( ; dsp_phase_index == start_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = interp_coeff[fluid_phase_fract_to_tablerow (dsp_phase)];
+ dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * start_point
+ + coeffs[1] * dsp_data[dsp_phase_index]
+ + coeffs[2] * dsp_data[dsp_phase_index+1]
+ + coeffs[3] * dsp_data[dsp_phase_index+2]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ /* interpolate the sequence of sample points */
+ for ( ; dsp_i < FLUID_BUFSIZE && dsp_phase_index <= end_index; dsp_i++)
+ {
+ coeffs = interp_coeff[fluid_phase_fract_to_tablerow (dsp_phase)];
+ dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index-1]
+ + coeffs[1] * dsp_data[dsp_phase_index]
+ + coeffs[2] * dsp_data[dsp_phase_index+1]
+ + coeffs[3] * dsp_data[dsp_phase_index+2]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ /* break out if buffer filled */
+ if (dsp_i >= FLUID_BUFSIZE) break;
+
+ end_index++; /* we're now interpolating the 2nd to last point */
+
+ /* interpolate within 2nd to last point */
+ for (; dsp_phase_index <= end_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = interp_coeff[fluid_phase_fract_to_tablerow (dsp_phase)];
+ dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index-1]
+ + coeffs[1] * dsp_data[dsp_phase_index]
+ + coeffs[2] * dsp_data[dsp_phase_index+1]
+ + coeffs[3] * end_point1);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ end_index++; /* we're now interpolating the last point */
+
+ /* interpolate within the last point */
+ for (; dsp_phase_index <= end_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = interp_coeff[fluid_phase_fract_to_tablerow (dsp_phase)];
+ dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index-1]
+ + coeffs[1] * dsp_data[dsp_phase_index]
+ + coeffs[2] * end_point1
+ + coeffs[3] * end_point2);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ if (!looping) break; /* break out if not looping (end of sample) */
+
+ /* go back to loop start */
+ if (dsp_phase_index > end_index)
+ {
+ fluid_phase_sub_int (dsp_phase, voice->loopend - voice->loopstart);
+
+ if (!voice->has_looped)
+ {
+ voice->has_looped = 1;
+ start_index = voice->loopstart;
+ start_point = dsp_data[voice->loopend - 1];
+ }
+ }
+
+ /* break out if filled buffer */
+ if (dsp_i >= FLUID_BUFSIZE) break;
+
+ end_index -= 2; /* set end back to third to last sample point */
+ }
+
+ voice->phase = dsp_phase;
+ voice->amp = dsp_amp;
+
+ return (dsp_i);
+}
+
+/* 7th order interpolation.
+ * Returns number of samples processed (usually FLUID_BUFSIZE but could be
+ * smaller if end of sample occurs).
+ */
+int
+fluid_rvoice_dsp_interpolate_7th_order (fluid_rvoice_dsp_t *voice)
+{
+ fluid_phase_t dsp_phase = voice->phase;
+ fluid_phase_t dsp_phase_incr;
+ short int *dsp_data = voice->sample->data;
+ fluid_real_t *dsp_buf = voice->dsp_buf;
+ fluid_real_t dsp_amp = voice->amp;
+ fluid_real_t dsp_amp_incr = voice->amp_incr;
+ unsigned int dsp_i = 0;
+ unsigned int dsp_phase_index;
+ unsigned int start_index, end_index;
+ short int start_points[3];
+ short int end_points[3];
+ fluid_real_t *coeffs;
+ int looping;
+
+ /* Convert playback "speed" floating point value to phase index/fract */
+ fluid_phase_set_float (dsp_phase_incr, voice->phase_incr);
+
+ /* add 1/2 sample to dsp_phase since 7th order interpolation is centered on
+ * the 4th sample point */
+ fluid_phase_incr (dsp_phase, (fluid_phase_t)0x80000000);
+
+ /* voice is currently looping? */
+ looping = voice->is_looping;
+
+ /* last index before 7th interpolation point must be specially handled */
+ end_index = (looping ? voice->loopend - 1 : voice->end) - 3;
+
+ if (voice->has_looped) /* set start_index and start point if looped or not */
+ {
+ start_index = voice->loopstart;
+ start_points[0] = dsp_data[voice->loopend - 1];
+ start_points[1] = dsp_data[voice->loopend - 2];
+ start_points[2] = dsp_data[voice->loopend - 3];
+ }
+ else
+ {
+ start_index = voice->start;
+ start_points[0] = dsp_data[voice->start]; /* just duplicate the start point */
+ start_points[1] = start_points[0];
+ start_points[2] = start_points[0];
+ }
+
+ /* get the 3 points off the end (loop start if looping, duplicate point if end) */
+ if (looping)
+ {
+ end_points[0] = dsp_data[voice->loopstart];
+ end_points[1] = dsp_data[voice->loopstart + 1];
+ end_points[2] = dsp_data[voice->loopstart + 2];
+ }
+ else
+ {
+ end_points[0] = dsp_data[voice->end];
+ end_points[1] = end_points[0];
+ end_points[2] = end_points[0];
+ }
+
+ while (1)
+ {
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+
+ /* interpolate first sample point (start or loop start) if needed */
+ for ( ; dsp_phase_index == start_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
+
+ dsp_buf[dsp_i] = dsp_amp
+ * (coeffs[0] * (fluid_real_t)start_points[2]
+ + coeffs[1] * (fluid_real_t)start_points[1]
+ + coeffs[2] * (fluid_real_t)start_points[0]
+ + coeffs[3] * (fluid_real_t)dsp_data[dsp_phase_index]
+ + coeffs[4] * (fluid_real_t)dsp_data[dsp_phase_index+1]
+ + coeffs[5] * (fluid_real_t)dsp_data[dsp_phase_index+2]
+ + coeffs[6] * (fluid_real_t)dsp_data[dsp_phase_index+3]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ start_index++;
+
+ /* interpolate 2nd to first sample point (start or loop start) if needed */
+ for ( ; dsp_phase_index == start_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
+
+ dsp_buf[dsp_i] = dsp_amp
+ * (coeffs[0] * (fluid_real_t)start_points[1]
+ + coeffs[1] * (fluid_real_t)start_points[0]
+ + coeffs[2] * (fluid_real_t)dsp_data[dsp_phase_index-1]
+ + coeffs[3] * (fluid_real_t)dsp_data[dsp_phase_index]
+ + coeffs[4] * (fluid_real_t)dsp_data[dsp_phase_index+1]
+ + coeffs[5] * (fluid_real_t)dsp_data[dsp_phase_index+2]
+ + coeffs[6] * (fluid_real_t)dsp_data[dsp_phase_index+3]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ start_index++;
+
+ /* interpolate 3rd to first sample point (start or loop start) if needed */
+ for ( ; dsp_phase_index == start_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
+
+ dsp_buf[dsp_i] = dsp_amp
+ * (coeffs[0] * (fluid_real_t)start_points[0]
+ + coeffs[1] * (fluid_real_t)dsp_data[dsp_phase_index-2]
+ + coeffs[2] * (fluid_real_t)dsp_data[dsp_phase_index-1]
+ + coeffs[3] * (fluid_real_t)dsp_data[dsp_phase_index]
+ + coeffs[4] * (fluid_real_t)dsp_data[dsp_phase_index+1]
+ + coeffs[5] * (fluid_real_t)dsp_data[dsp_phase_index+2]
+ + coeffs[6] * (fluid_real_t)dsp_data[dsp_phase_index+3]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ start_index -= 2; /* set back to original start index */
+
+
+ /* interpolate the sequence of sample points */
+ for ( ; dsp_i < FLUID_BUFSIZE && dsp_phase_index <= end_index; dsp_i++)
+ {
+ coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
+
+ dsp_buf[dsp_i] = dsp_amp
+ * (coeffs[0] * (fluid_real_t)dsp_data[dsp_phase_index-3]
+ + coeffs[1] * (fluid_real_t)dsp_data[dsp_phase_index-2]
+ + coeffs[2] * (fluid_real_t)dsp_data[dsp_phase_index-1]
+ + coeffs[3] * (fluid_real_t)dsp_data[dsp_phase_index]
+ + coeffs[4] * (fluid_real_t)dsp_data[dsp_phase_index+1]
+ + coeffs[5] * (fluid_real_t)dsp_data[dsp_phase_index+2]
+ + coeffs[6] * (fluid_real_t)dsp_data[dsp_phase_index+3]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ /* break out if buffer filled */
+ if (dsp_i >= FLUID_BUFSIZE) break;
+
+ end_index++; /* we're now interpolating the 3rd to last point */
+
+ /* interpolate within 3rd to last point */
+ for (; dsp_phase_index <= end_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
+
+ dsp_buf[dsp_i] = dsp_amp
+ * (coeffs[0] * (fluid_real_t)dsp_data[dsp_phase_index-3]
+ + coeffs[1] * (fluid_real_t)dsp_data[dsp_phase_index-2]
+ + coeffs[2] * (fluid_real_t)dsp_data[dsp_phase_index-1]
+ + coeffs[3] * (fluid_real_t)dsp_data[dsp_phase_index]
+ + coeffs[4] * (fluid_real_t)dsp_data[dsp_phase_index+1]
+ + coeffs[5] * (fluid_real_t)dsp_data[dsp_phase_index+2]
+ + coeffs[6] * (fluid_real_t)end_points[0]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ end_index++; /* we're now interpolating the 2nd to last point */
+
+ /* interpolate within 2nd to last point */
+ for (; dsp_phase_index <= end_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
+
+ dsp_buf[dsp_i] = dsp_amp
+ * (coeffs[0] * (fluid_real_t)dsp_data[dsp_phase_index-3]
+ + coeffs[1] * (fluid_real_t)dsp_data[dsp_phase_index-2]
+ + coeffs[2] * (fluid_real_t)dsp_data[dsp_phase_index-1]
+ + coeffs[3] * (fluid_real_t)dsp_data[dsp_phase_index]
+ + coeffs[4] * (fluid_real_t)dsp_data[dsp_phase_index+1]
+ + coeffs[5] * (fluid_real_t)end_points[0]
+ + coeffs[6] * (fluid_real_t)end_points[1]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ end_index++; /* we're now interpolating the last point */
+
+ /* interpolate within last point */
+ for (; dsp_phase_index <= end_index && dsp_i < FLUID_BUFSIZE; dsp_i++)
+ {
+ coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
+
+ dsp_buf[dsp_i] = dsp_amp
+ * (coeffs[0] * (fluid_real_t)dsp_data[dsp_phase_index-3]
+ + coeffs[1] * (fluid_real_t)dsp_data[dsp_phase_index-2]
+ + coeffs[2] * (fluid_real_t)dsp_data[dsp_phase_index-1]
+ + coeffs[3] * (fluid_real_t)dsp_data[dsp_phase_index]
+ + coeffs[4] * (fluid_real_t)end_points[0]
+ + coeffs[5] * (fluid_real_t)end_points[1]
+ + coeffs[6] * (fluid_real_t)end_points[2]);
+
+ /* increment phase and amplitude */
+ fluid_phase_incr (dsp_phase, dsp_phase_incr);
+ dsp_phase_index = fluid_phase_index (dsp_phase);
+ dsp_amp += dsp_amp_incr;
+ }
+
+ if (!looping) break; /* break out if not looping (end of sample) */
+
+ /* go back to loop start */
+ if (dsp_phase_index > end_index)
+ {
+ fluid_phase_sub_int (dsp_phase, voice->loopend - voice->loopstart);
+
+ if (!voice->has_looped)
+ {
+ voice->has_looped = 1;
+ start_index = voice->loopstart;
+ start_points[0] = dsp_data[voice->loopend - 1];
+ start_points[1] = dsp_data[voice->loopend - 2];
+ start_points[2] = dsp_data[voice->loopend - 3];
+ }
+ }
+
+ /* break out if filled buffer */
+ if (dsp_i >= FLUID_BUFSIZE) break;
+
+ end_index -= 3; /* set end back to 4th to last sample point */
+ }
+
+ /* sub 1/2 sample from dsp_phase since 7th order interpolation is centered on
+ * the 4th sample point (correct back to real value) */
+ fluid_phase_decr (dsp_phase, (fluid_phase_t)0x80000000);
+
+ voice->phase = dsp_phase;
+ voice->amp = dsp_amp;
+
+ return (dsp_i);
+}
generated by cgit v1.2.3 (git 2.39.1) at 2024-05-06 09:51:26 +0000