645 lines
26 KiB
C++
645 lines
26 KiB
C++
/* FluidSynth - A Software Synthesizer
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*
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* Copyright (C) 2003 Peter Hanappe and others.
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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 License
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* as published by the Free Software Foundation; either version 2 of
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* 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, but
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* 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
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
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* 02111-1307, USA
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*/
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#include "fluid.h"
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#include "voice.h"
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#include "sfont.h"
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namespace FluidS {
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/* Purpose:
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*
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* Interpolates audio data (obtains values between the samples of the original
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* waveform data).
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*
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* Variables loaded from the voice structure (assigned in fluid_voice_write()):
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* - dsp_data: Pointer to the original waveform data
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* - dsp_phase: The position in the original waveform data.
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* This has an integer and a fractional part (between samples).
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* - dsp_phase_incr: For each output sample, the position in the original
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* waveform advances by dsp_phase_incr. This also has an integer
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* part and a fractional part.
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* If a sample is played at root pitch (no pitch change),
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* dsp_phase_incr is integer=1 and fractional=0.
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* - dsp_amp: The current amplitude envelope value.
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* - dsp_amp_incr: The changing rate of the amplitude envelope.
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*
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* A couple of variables are used internally, their results are discarded:
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* - dsp_i: Index through the output buffer
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* - dsp_buf: Output buffer of floating point values (FLUID_BUFSIZE in length)
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*/
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/* Interpolation (find a value between two samples of the original waveform) */
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/* Linear interpolation table (2 coefficients centered on 1st) */
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float Voice::interp_coeff_linear[FLUID_INTERP_MAX][2];
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/* 4th order (cubic) interpolation table (4 coefficients centered on 2nd) */
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float Voice::interp_coeff[FLUID_INTERP_MAX][4];
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/* 7th order interpolation (7 coefficients centered on 3rd) */
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float Voice::sinc_table7[FLUID_INTERP_MAX][7];
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#define SINC_INTERP_ORDER 7 /* 7th order constant */
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//---------------------------------------------------------
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// dsp_float_config
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// Initializes interpolation tables
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//---------------------------------------------------------
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void Voice::dsp_float_config()
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{
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/* Initialize the coefficients for the interpolation. The math comes
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* from a mail, posted by Olli Niemitalo to the music-dsp mailing
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* list (I found it in the music-dsp archives
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* http://www.smartelectronix.com/musicdsp/). */
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for (int i = 0; i < FLUID_INTERP_MAX; i++) {
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double x = (double) i / (double) FLUID_INTERP_MAX;
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interp_coeff[i][0] = (float)(x * (-0.5 + x * (1 - 0.5 * x)));
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interp_coeff[i][1] = (float)(1.0 + x * x * (1.5 * x - 2.5));
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interp_coeff[i][2] = (float)(x * (0.5 + x * (2.0 - 1.5 * x)));
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interp_coeff[i][3] = (float)(0.5 * x * x * (x - 1.0));
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interp_coeff_linear[i][0] = (float)(1.0 - x);
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interp_coeff_linear[i][1] = (float)x;
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}
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/* i: Offset in terms of whole samples */
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for (int i = 0; i < SINC_INTERP_ORDER; i++) {
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/* i2: Offset in terms of fractional samples ('subsamples') */
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for (int i2 = 0; i2 < FLUID_INTERP_MAX; i2++) {
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/* center on middle of table */
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double i_shifted = (double)i - ((double)SINC_INTERP_ORDER / 2.0)
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+ (double)i2 / (double)FLUID_INTERP_MAX;
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/* sinc(0) cannot be calculated straightforward (limit needed for 0/0) */
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double v;
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if (fabs (i_shifted) > 0.000001) {
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v = (float)sin (i_shifted * M_PI) / (M_PI * i_shifted);
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/* Hamming window */
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v *= (float)0.5 * (1.0 + cos (2.0 * M_PI * i_shifted / (float)SINC_INTERP_ORDER));
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}
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else
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v = 1.0;
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sinc_table7[FLUID_INTERP_MAX - i2 - 1][i] = v;
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}
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}
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fluid_check_fpe("interpolation table calculation");
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}
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//-------------------------------------------------------------------
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// fluid_dsp_float_interpolate_none
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// No interpolation. Just take the sample, which is closest to
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// the playback pointer. Questionable quality, but very
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// efficient.
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//-------------------------------------------------------------------
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int Voice::dsp_float_interpolate_none(unsigned n)
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{
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Voice* voice = this;
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Phase dsp_phase = voice->phase;
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Phase dsp_phase_incr; // end_phase;
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short int *dsp_data = voice->sample->data;
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float *dsp_buf = voice->dsp_buf;
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float dsp_amp = voice->amp;
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float dsp_amp_incr = voice->amp_incr;
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unsigned int dsp_i = 0;
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unsigned int dsp_phase_index;
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unsigned int end_index;
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int looping;
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/* Convert playback "speed" floating point value to phase index/fract */
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dsp_phase_incr.setFloat(voice->phase_incr);
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/* voice is currently looping? */
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looping = SAMPLEMODE() == FLUID_LOOP_DURING_RELEASE
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|| (SAMPLEMODE() == FLUID_LOOP_UNTIL_RELEASE
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&& voice->volenv_section < FLUID_VOICE_ENVRELEASE);
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end_index = looping ? voice->loopend - 1 : voice->end;
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while(1) {
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dsp_phase_index = dsp_phase.index_round(); // round to nearest point
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/* interpolate sequence of sample points */
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for ( ; dsp_i < n && dsp_phase_index <= end_index; dsp_i++) {
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dsp_buf[dsp_i] = dsp_amp * dsp_data[dsp_phase_index];
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index_round(); /* round to nearest point */
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dsp_amp += dsp_amp_incr;
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}
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/* break out if not looping (buffer may not be full) */
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if (!looping)
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break;
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/* go back to loop start */
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if (dsp_phase_index > end_index) {
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dsp_phase -= (voice->loopend - voice->loopstart);
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voice->has_looped = true;
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}
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/* break out if filled buffer */
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if (dsp_i >= n)
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break;
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}
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voice->phase = dsp_phase;
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voice->amp = dsp_amp;
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return dsp_i;
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}
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//---------------------------------------------------------
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// dsp_float_interpolate_linear
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// Straight line interpolation.
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// Returns number of samples processed (usually FLUID_BUFSIZE but could be
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// smaller if end of sample occurs).
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//---------------------------------------------------------
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int Voice::dsp_float_interpolate_linear(unsigned n)
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{
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Voice* voice = this;
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Phase dsp_phase = voice->phase;
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Phase dsp_phase_incr; // end_phase;
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short int *dsp_data = voice->sample->data;
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float *dsp_buf = voice->dsp_buf;
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float dsp_amp = voice->amp;
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float dsp_amp_incr = voice->amp_incr;
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unsigned int dsp_i = 0;
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unsigned int dsp_phase_index;
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unsigned int end_index;
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short int point;
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float *coeffs;
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int looping;
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/* Convert playback "speed" floating point value to phase index/fract */
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dsp_phase_incr.setFloat(voice->phase_incr);
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/* voice is currently looping? */
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looping = SAMPLEMODE() == FLUID_LOOP_DURING_RELEASE
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|| (SAMPLEMODE() == FLUID_LOOP_UNTIL_RELEASE
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&& voice->volenv_section < FLUID_VOICE_ENVRELEASE);
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/* last index before 2nd interpolation point must be specially handled */
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end_index = (looping ? voice->loopend - 1 : voice->end) - 1;
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/* 2nd interpolation point to use at end of loop or sample */
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if (looping)
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point = dsp_data[voice->loopstart]; /* loop start */
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else
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point = dsp_data[voice->end]; /* duplicate end for samples no longer looping */
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while (1) {
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dsp_phase_index = dsp_phase.index();
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/* interpolate the sequence of sample points */
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for ( ; dsp_i < n && dsp_phase_index <= end_index; dsp_i++) {
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coeffs = interp_coeff_linear[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index]
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+ coeffs[1] * dsp_data[dsp_phase_index+1]);
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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dsp_amp += dsp_amp_incr;
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}
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/* break out if buffer filled */
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if (dsp_i >= n)
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break;
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end_index++; /* we're now interpolating the last point */
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/* interpolate within last point */
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for (; dsp_phase_index <= end_index && dsp_i < n; dsp_i++) {
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coeffs = interp_coeff_linear[fluid_phase_fract_to_tablerow (dsp_phase)];
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dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * dsp_data[dsp_phase_index]
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+ coeffs[1] * point);
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/* increment phase and amplitude */
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dsp_phase += dsp_phase_incr;
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dsp_phase_index = dsp_phase.index();
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dsp_amp += dsp_amp_incr; /* increment amplitude */
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}
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if (!looping)
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break; /* break out if not looping (end of sample) */
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/* go back to loop start (if past */
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if (dsp_phase_index > end_index) {
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dsp_phase -= (voice->loopend - voice->loopstart);
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voice->has_looped = true;
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}
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/* break out if filled buffer */
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if (dsp_i >= n)
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break;
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end_index--; /* set end back to second to last sample point */
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}
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voice->phase = dsp_phase;
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voice->amp = dsp_amp;
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return dsp_i;
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}
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//-----------------------------------------------------------------------------
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// dsp_float_interpolate_4th_order
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// 4th order (cubic) interpolation.
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// Returns number of samples processed (usually FLUID_BUFSIZE but could be
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// smaller if end of sample occurs).
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//-----------------------------------------------------------------------------
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int Voice::dsp_float_interpolate_4th_order(unsigned n)
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{
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Phase dsp_phase_incr; // end_phase;
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short int* dsp_data = sample->data;
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float dsp_amp_incr = amp_incr;
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unsigned int dsp_i = 0;
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unsigned int dsp_phase_index;
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unsigned int start_index;
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short int start_point, end_point1, end_point2;
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float *coeffs;
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/* Convert playback "speed" floating point value to phase index/fract */
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dsp_phase_incr.setFloat(phase_incr);
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/* voice is currently looping? */
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int looping = SAMPLEMODE() == FLUID_LOOP_DURING_RELEASE
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|| (SAMPLEMODE() == FLUID_LOOP_UNTIL_RELEASE
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&& volenv_section < FLUID_VOICE_ENVRELEASE);
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/* last index before 4th interpolation point must be specially handled */
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unsigned int end_index = (looping ? loopend - 1 : end) - 2;
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if (has_looped) { /* set start_index and start point if looped or not */
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start_index = loopstart;
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start_point = dsp_data[loopend - 1]; /* last point in loop (wrap around) */
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}
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else {
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start_index = start;
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start_point = dsp_data[start]; /* just duplicate the point */
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}
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/* get points off the end (loop start if looping, duplicate point if end) */
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if (looping) {
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end_point1 = dsp_data[loopstart];
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end_point2 = dsp_data[loopstart + 1];
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}
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else {
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end_point1 = dsp_data[end];
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end_point2 = end_point1;
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}
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while (1) {
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dsp_phase_index = phase.index();
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/* interpolate first sample point (start or loop start) if needed */
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for ( ; dsp_phase_index == start_index && dsp_i < n; dsp_i++) {
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coeffs = interp_coeff[fluid_phase_fract_to_tablerow (phase)];
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dsp_buf[dsp_i] = amp * (coeffs[0] * start_point
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+ coeffs[1] * dsp_data[dsp_phase_index]
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+ coeffs[2] * dsp_data[dsp_phase_index+1]
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+ coeffs[3] * dsp_data[dsp_phase_index+2]);
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/* increment phase and amplitude */
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phase += dsp_phase_incr;
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dsp_phase_index = phase.index();
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amp += dsp_amp_incr;
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}
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/* interpolate the sequence of sample points */
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for ( ; dsp_i < n && dsp_phase_index <= end_index; dsp_i++) {
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coeffs = interp_coeff[fluid_phase_fract_to_tablerow (phase)];
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dsp_buf[dsp_i] = amp * (coeffs[0] * dsp_data[dsp_phase_index-1]
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+ coeffs[1] * dsp_data[dsp_phase_index]
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+ coeffs[2] * dsp_data[dsp_phase_index+1]
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+ coeffs[3] * dsp_data[dsp_phase_index+2]);
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/* increment phase and amplitude */
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phase += dsp_phase_incr;
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dsp_phase_index = phase.index();
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amp += dsp_amp_incr;
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}
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/* break out if buffer filled */
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if (dsp_i >= n)
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break;
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end_index++; /* we're now interpolating the 2nd to last point */
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/* interpolate within 2nd to last point */
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for (; dsp_phase_index <= end_index && dsp_i < n; dsp_i++) {
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coeffs = interp_coeff[fluid_phase_fract_to_tablerow (phase)];
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dsp_buf[dsp_i] = amp * (coeffs[0] * dsp_data[dsp_phase_index-1]
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+ coeffs[1] * dsp_data[dsp_phase_index]
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+ coeffs[2] * dsp_data[dsp_phase_index+1]
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+ coeffs[3] * end_point1);
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/* increment phase and amplitude */
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phase += dsp_phase_incr;
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dsp_phase_index = phase.index();
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amp += dsp_amp_incr;
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}
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end_index++; /* we're now interpolating the last point */
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/* interpolate within the last point */
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for (; dsp_phase_index <= end_index && dsp_i < n; dsp_i++) {
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coeffs = interp_coeff[fluid_phase_fract_to_tablerow (phase)];
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dsp_buf[dsp_i] = amp * (coeffs[0] * dsp_data[dsp_phase_index-1]
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+ coeffs[1] * dsp_data[dsp_phase_index]
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+ coeffs[2] * end_point1
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+ coeffs[3] * end_point2);
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/* increment phase and amplitude */
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phase += dsp_phase_incr;
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dsp_phase_index = phase.index();
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amp += dsp_amp_incr;
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}
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if (!looping)
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break; /* break out if not looping (end of sample) */
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/* go back to loop start */
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if (dsp_phase_index > end_index) {
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phase -= (loopend - loopstart);
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if (!has_looped) {
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has_looped = true;
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start_index = loopstart;
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start_point = dsp_data[loopend - 1];
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}
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}
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/* break out if filled buffer */
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if (dsp_i >= n)
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break;
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end_index -= 2; /* set end back to third to last sample point */
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}
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return dsp_i;
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}
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//-----------------------------------------------------------------------------
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// dsp_float_interpolate_7th_order
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// 7th order interpolation.
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// Returns number of samples processed (usually FLUID_BUFSIZE but could be
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// smaller if end of sample occurs).
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//-----------------------------------------------------------------------------
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int Voice::dsp_float_interpolate_7th_order(unsigned n)
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{
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Voice* voice = this;
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Phase dsp_phase = voice->phase;
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Phase dsp_phase_incr; // end_phase;
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short int *dsp_data = voice->sample->data;
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float *dsp_buf = voice->dsp_buf;
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float dsp_amp = voice->amp;
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float dsp_amp_incr = voice->amp_incr;
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unsigned int dsp_i = 0;
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unsigned int dsp_phase_index;
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unsigned int start_index, end_index;
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short int start_points[3];
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short int end_points[3];
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float *coeffs;
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int looping;
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/* Convert playback "speed" floating point value to phase index/fract */
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dsp_phase_incr.setFloat(voice->phase_incr);
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/* add 1/2 sample to dsp_phase since 7th order interpolation is centered on
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* the 4th sample point */
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dsp_phase += (Phase)0x80000000;
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/* voice is currently looping? */
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looping = SAMPLEMODE() == FLUID_LOOP_DURING_RELEASE
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|| (SAMPLEMODE() == FLUID_LOOP_UNTIL_RELEASE
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&& voice->volenv_section < FLUID_VOICE_ENVRELEASE);
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/* last index before 7th interpolation point must be specially handled */
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end_index = (looping ? voice->loopend - 1 : voice->end) - 3;
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if (voice->has_looped) { /* set start_index and start point if looped or not */
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start_index = voice->loopstart;
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start_points[0] = dsp_data[voice->loopend - 1];
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start_points[1] = dsp_data[voice->loopend - 2];
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start_points[2] = dsp_data[voice->loopend - 3];
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}
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else {
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start_index = voice->start;
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start_points[0] = dsp_data[voice->start]; /* just duplicate the start point */
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start_points[1] = start_points[0];
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start_points[2] = start_points[0];
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}
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/* get the 3 points off the end (loop start if looping, duplicate point if end) */
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if (looping) {
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end_points[0] = dsp_data[voice->loopstart];
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end_points[1] = dsp_data[voice->loopstart + 1];
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end_points[2] = dsp_data[voice->loopstart + 2];
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}
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else {
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end_points[0] = dsp_data[voice->end];
|
|
end_points[1] = end_points[0];
|
|
end_points[2] = end_points[0];
|
|
}
|
|
|
|
while (1) {
|
|
dsp_phase_index = dsp_phase.index();
|
|
|
|
/* interpolate first sample point (start or loop start) if needed */
|
|
for ( ; dsp_phase_index == start_index && dsp_i < n; dsp_i++) {
|
|
coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
|
|
|
|
dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)start_points[2]
|
|
+ coeffs[1] * (float)start_points[1]
|
|
+ coeffs[2] * (float)start_points[0]
|
|
+ coeffs[3] * (float)dsp_data[dsp_phase_index]
|
|
+ coeffs[4] * (float)dsp_data[dsp_phase_index+1]
|
|
+ coeffs[5] * (float)dsp_data[dsp_phase_index+2]
|
|
+ coeffs[6] * (float)dsp_data[dsp_phase_index+3]);
|
|
|
|
/* increment phase and amplitude */
|
|
dsp_phase += dsp_phase_incr;
|
|
dsp_phase_index = dsp_phase.index();
|
|
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 < n; dsp_i++) {
|
|
coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
|
|
|
|
dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)start_points[1]
|
|
+ coeffs[1] * (float)start_points[0]
|
|
+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
|
|
+ coeffs[3] * (float)dsp_data[dsp_phase_index]
|
|
+ coeffs[4] * (float)dsp_data[dsp_phase_index+1]
|
|
+ coeffs[5] * (float)dsp_data[dsp_phase_index+2]
|
|
+ coeffs[6] * (float)dsp_data[dsp_phase_index+3]);
|
|
|
|
/* increment phase and amplitude */
|
|
dsp_phase += dsp_phase_incr;
|
|
dsp_phase_index = dsp_phase.index();
|
|
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 < n; dsp_i++) {
|
|
coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
|
|
|
|
dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)start_points[0]
|
|
+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
|
|
+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
|
|
+ coeffs[3] * (float)dsp_data[dsp_phase_index]
|
|
+ coeffs[4] * (float)dsp_data[dsp_phase_index+1]
|
|
+ coeffs[5] * (float)dsp_data[dsp_phase_index+2]
|
|
+ coeffs[6] * (float)dsp_data[dsp_phase_index+3]);
|
|
|
|
/* increment phase and amplitude */
|
|
dsp_phase += dsp_phase_incr;
|
|
dsp_phase_index = dsp_phase.index();
|
|
dsp_amp += dsp_amp_incr;
|
|
}
|
|
|
|
start_index -= 2; /* set back to original start index */
|
|
|
|
/* interpolate the sequence of sample points */
|
|
for ( ; dsp_i < n && 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] * (float)dsp_data[dsp_phase_index-3]
|
|
+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
|
|
+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
|
|
+ coeffs[3] * (float)dsp_data[dsp_phase_index]
|
|
+ coeffs[4] * (float)dsp_data[dsp_phase_index+1]
|
|
+ coeffs[5] * (float)dsp_data[dsp_phase_index+2]
|
|
+ coeffs[6] * (float)dsp_data[dsp_phase_index+3]);
|
|
|
|
/* increment phase and amplitude */
|
|
dsp_phase += dsp_phase_incr;
|
|
dsp_phase_index = dsp_phase.index();
|
|
dsp_amp += dsp_amp_incr;
|
|
}
|
|
|
|
/* break out if buffer filled */
|
|
if (dsp_i >= n)
|
|
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 < n; dsp_i++) {
|
|
coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
|
|
|
|
dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
|
|
+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
|
|
+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
|
|
+ coeffs[3] * (float)dsp_data[dsp_phase_index]
|
|
+ coeffs[4] * (float)dsp_data[dsp_phase_index+1]
|
|
+ coeffs[5] * (float)dsp_data[dsp_phase_index+2]
|
|
+ coeffs[6] * (float)end_points[0]);
|
|
|
|
/* increment phase and amplitude */
|
|
dsp_phase += dsp_phase_incr;
|
|
dsp_phase_index = dsp_phase.index();
|
|
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 < n; dsp_i++) {
|
|
coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
|
|
|
|
dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
|
|
+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
|
|
+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
|
|
+ coeffs[3] * (float)dsp_data[dsp_phase_index]
|
|
+ coeffs[4] * (float)dsp_data[dsp_phase_index+1]
|
|
+ coeffs[5] * (float)end_points[0]
|
|
+ coeffs[6] * (float)end_points[1]);
|
|
|
|
/* increment phase and amplitude */
|
|
dsp_phase += dsp_phase_incr;
|
|
dsp_phase_index = dsp_phase.index();
|
|
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 < n; dsp_i++) {
|
|
coeffs = sinc_table7[fluid_phase_fract_to_tablerow (dsp_phase)];
|
|
|
|
dsp_buf[dsp_i] = dsp_amp * (coeffs[0] * (float)dsp_data[dsp_phase_index-3]
|
|
+ coeffs[1] * (float)dsp_data[dsp_phase_index-2]
|
|
+ coeffs[2] * (float)dsp_data[dsp_phase_index-1]
|
|
+ coeffs[3] * (float)dsp_data[dsp_phase_index]
|
|
+ coeffs[4] * (float)end_points[0]
|
|
+ coeffs[5] * (float)end_points[1]
|
|
+ coeffs[6] * (float)end_points[2]);
|
|
|
|
/* increment phase and amplitude */
|
|
dsp_phase += dsp_phase_incr;
|
|
dsp_phase_index = dsp_phase.index();
|
|
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) {
|
|
dsp_phase -= (voice->loopend - voice->loopstart);
|
|
|
|
if (!voice->has_looped) {
|
|
voice->has_looped = true;
|
|
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 >= n)
|
|
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) */
|
|
dsp_phase -= (Phase)0x80000000;
|
|
|
|
voice->phase = dsp_phase;
|
|
voice->amp = dsp_amp;
|
|
|
|
return dsp_i;
|
|
}
|
|
}
|
|
|