/* 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
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
 * 02111-1307, USA
 */

#include "fluid.h"
#include "voice.h"
#include "sfont.h"

namespace FluidS {

/* 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) */
float Voice::interp_coeff_linear[FLUID_INTERP_MAX][2];

/* 4th order (cubic) interpolation table (4 coefficients centered on 2nd) */
float Voice::interp_coeff[FLUID_INTERP_MAX][4];

/* 7th order interpolation (7 coefficients centered on 3rd) */
float Voice::sinc_table7[FLUID_INTERP_MAX][7];

#define SINC_INTERP_ORDER 7	/* 7th order constant */

//---------------------------------------------------------
//   dsp_float_config
//    Initializes interpolation tables
//---------------------------------------------------------

void Voice::dsp_float_config()
      {
      /* 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 (int i = 0; i < FLUID_INTERP_MAX; i++) {
            double x = (double) i / (double) FLUID_INTERP_MAX;

            interp_coeff[i][0] = (float)(x * (-0.5 + x * (1 - 0.5 * x)));
            interp_coeff[i][1] = (float)(1.0 + x * x * (1.5 * x - 2.5));
            interp_coeff[i][2] = (float)(x * (0.5 + x * (2.0 - 1.5 * x)));
            interp_coeff[i][3] = (float)(0.5 * x * x * (x - 1.0));

            interp_coeff_linear[i][0] = (float)(1.0 - x);
            interp_coeff_linear[i][1] = (float)x;
            }

      /* i: Offset in terms of whole samples */
      for (int i = 0; i < SINC_INTERP_ORDER; i++) {
            /* i2: Offset in terms of fractional samples ('subsamples') */
            for (int i2 = 0; i2 < FLUID_INTERP_MAX; i2++) {
                  /* center on middle of table */
                  double 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) */
                  double v;
                  if (fabs (i_shifted) > 0.000001) {
                        v = (float)sin (i_shifted * M_PI) / (M_PI * i_shifted);
                        /* Hamming window */
                        v *= (float)0.5 * (1.0 + cos (2.0 * M_PI * i_shifted / (float)SINC_INTERP_ORDER));
                        }
                  else
                        v = 1.0;
                  sinc_table7[FLUID_INTERP_MAX - i2 - 1][i] = v;
                  }
            }
      fluid_check_fpe("interpolation table calculation");
      }

//-------------------------------------------------------------------
//   fluid_dsp_float_interpolate_none
//    No interpolation. Just take the sample, which is closest to
//    the playback pointer.  Questionable quality, but very
//    efficient.
//-------------------------------------------------------------------

int Voice::dsp_float_interpolate_none(unsigned n)
      {
      Voice* voice = this;

      Phase dsp_phase = voice->phase;
      Phase dsp_phase_incr; //  end_phase;
      short int *dsp_data = voice->sample->data;
      float *dsp_buf = voice->dsp_buf;
      float dsp_amp = voice->amp;
      float 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 */
      dsp_phase_incr.setFloat(voice->phase_incr);

      /* voice is currently looping? */
      looping = SAMPLEMODE() == FLUID_LOOP_DURING_RELEASE
         || (SAMPLEMODE() == FLUID_LOOP_UNTIL_RELEASE
         && voice->volenv_section < FLUID_VOICE_ENVRELEASE);

      end_index = looping ? voice->loopend - 1 : voice->end;

      while(1) {
            dsp_phase_index = dsp_phase.index_round();      // round to nearest point

            /* interpolate sequence of sample points */
            for ( ; dsp_i < n && dsp_phase_index <= end_index; dsp_i++) {
                  dsp_buf[dsp_i] = dsp_amp * dsp_data[dsp_phase_index];

                  /* increment phase and amplitude */
                  dsp_phase += dsp_phase_incr;
                  dsp_phase_index = dsp_phase.index_round();	/* 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) {
                  dsp_phase -= (voice->loopend - voice->loopstart);
                  voice->has_looped = true;
                  }

            /* break out if filled buffer */
            if (dsp_i >= n)
                  break;
            }

      voice->phase = dsp_phase;
      voice->amp = dsp_amp;
      return dsp_i;
      }

//---------------------------------------------------------
//   dsp_float_interpolate_linear
//    Straight line interpolation.
//    Returns number of samples processed (usually FLUID_BUFSIZE but could be
//    smaller if end of sample occurs).
//---------------------------------------------------------

int Voice::dsp_float_interpolate_linear(unsigned n)
      {
      Voice* voice = this;
      Phase dsp_phase = voice->phase;
      Phase dsp_phase_incr; // end_phase;
      short int *dsp_data = voice->sample->data;
      float *dsp_buf = voice->dsp_buf;
      float dsp_amp = voice->amp;
      float dsp_amp_incr = voice->amp_incr;
      unsigned int dsp_i = 0;
      unsigned int dsp_phase_index;
      unsigned int end_index;
      short int point;
      float *coeffs;
      int looping;

      /* Convert playback "speed" floating point value to phase index/fract */
      dsp_phase_incr.setFloat(voice->phase_incr);

      /* voice is currently looping? */
      looping = SAMPLEMODE() == FLUID_LOOP_DURING_RELEASE
         || (SAMPLEMODE() == FLUID_LOOP_UNTIL_RELEASE
         && voice->volenv_section < FLUID_VOICE_ENVRELEASE);

      /* 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 = dsp_phase.index();

            /* interpolate the sequence of sample points */
            for ( ; dsp_i < n && 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 */
                  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 last point */

            /* interpolate within last point */
            for (; dsp_phase_index <= end_index && dsp_i < n; 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 */
                  dsp_phase += dsp_phase_incr;
                  dsp_phase_index = dsp_phase.index();
                  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) {
                  dsp_phase -= (voice->loopend - voice->loopstart);
                  voice->has_looped = true;
                  }

            /* break out if filled buffer */
            if (dsp_i >= n)
                  break;
            end_index--;	/* set end back to second to last sample point */
            }

      voice->phase = dsp_phase;
      voice->amp = dsp_amp;
      return dsp_i;
      }

//-----------------------------------------------------------------------------
//   dsp_float_interpolate_4th_order
//    4th order (cubic) interpolation.
//    Returns number of samples processed (usually FLUID_BUFSIZE but could be
//    smaller if end of sample occurs).
//-----------------------------------------------------------------------------

int Voice::dsp_float_interpolate_4th_order(unsigned n)
      {
      Phase dsp_phase_incr; // end_phase;
      short int* dsp_data = sample->data;
      float dsp_amp_incr  = amp_incr;
      unsigned int dsp_i  = 0;
      unsigned int dsp_phase_index;
      unsigned int start_index;
      short int start_point, end_point1, end_point2;
      float *coeffs;

      /* Convert playback "speed" floating point value to phase index/fract */
      dsp_phase_incr.setFloat(phase_incr);

      /* voice is currently looping? */
      int looping = SAMPLEMODE() == FLUID_LOOP_DURING_RELEASE
         || (SAMPLEMODE() == FLUID_LOOP_UNTIL_RELEASE
         && volenv_section < FLUID_VOICE_ENVRELEASE);

      /* last index before 4th interpolation point must be specially handled */
      unsigned int end_index = (looping ? loopend - 1 : end) - 2;

      if (has_looped) {	/* set start_index and start point if looped or not */
            start_index = loopstart;
            start_point = dsp_data[loopend - 1];	/* last point in loop (wrap around) */
            }
      else {
            start_index = start;
            start_point = dsp_data[start];	/* just duplicate the point */
            }

      /* get points off the end (loop start if looping, duplicate point if end) */
      if (looping) {
            end_point1 = dsp_data[loopstart];
            end_point2 = dsp_data[loopstart + 1];
            }
      else {
            end_point1 = dsp_data[end];
            end_point2 = end_point1;
            }

      while (1) {
            dsp_phase_index = phase.index();

            /* interpolate first sample point (start or loop start) if needed */
            for ( ; dsp_phase_index == start_index && dsp_i < n; dsp_i++) {
                  coeffs = interp_coeff[fluid_phase_fract_to_tablerow (phase)];
                  dsp_buf[dsp_i] = 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 */
                  phase += dsp_phase_incr;
                  dsp_phase_index = phase.index();
                  amp += dsp_amp_incr;
                  }

            /* interpolate the sequence of sample points */
            for ( ; dsp_i < n && dsp_phase_index <= end_index; dsp_i++) {
                  coeffs = interp_coeff[fluid_phase_fract_to_tablerow (phase)];
                  dsp_buf[dsp_i] = 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 */
                  phase += dsp_phase_incr;
                  dsp_phase_index = phase.index();
                  amp += dsp_amp_incr;
                  }

            /* break out if buffer filled */
            if (dsp_i >= n)
                  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 < n; dsp_i++) {
                  coeffs = interp_coeff[fluid_phase_fract_to_tablerow (phase)];
                  dsp_buf[dsp_i] = 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 */
                  phase += dsp_phase_incr;
                  dsp_phase_index = phase.index();
                  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 < n; dsp_i++) {
                  coeffs = interp_coeff[fluid_phase_fract_to_tablerow (phase)];
                  dsp_buf[dsp_i] = 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 */
                  phase += dsp_phase_incr;
                  dsp_phase_index = phase.index();
                  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) {
                  phase -= (loopend - loopstart);
                  if (!has_looped) {
                        has_looped = true;
                        start_index = loopstart;
                        start_point = dsp_data[loopend - 1];
                        }
                  }
            /* break out if filled buffer */
            if (dsp_i >= n)
                  break;
            end_index -= 2;	/* set end back to third to last sample point */
            }
      return dsp_i;
      }

//-----------------------------------------------------------------------------
//   dsp_float_interpolate_7th_order
//    7th order interpolation.
//    Returns number of samples processed (usually FLUID_BUFSIZE but could be
//    smaller if end of sample occurs).
//-----------------------------------------------------------------------------

int Voice::dsp_float_interpolate_7th_order(unsigned n)
      {
      Voice* voice = this;

      Phase dsp_phase = voice->phase;
      Phase dsp_phase_incr; // end_phase;
      short int *dsp_data = voice->sample->data;
      float *dsp_buf = voice->dsp_buf;
      float dsp_amp = voice->amp;
      float 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];
      float *coeffs;
      int looping;

      /* Convert playback "speed" floating point value to phase index/fract */
      dsp_phase_incr.setFloat(voice->phase_incr);

      /* add 1/2 sample to dsp_phase since 7th order interpolation is centered on
       * the 4th sample point */
      dsp_phase += (Phase)0x80000000;

      /* voice is currently looping? */
      looping = SAMPLEMODE() == FLUID_LOOP_DURING_RELEASE
         || (SAMPLEMODE() == FLUID_LOOP_UNTIL_RELEASE
         && voice->volenv_section < FLUID_VOICE_ENVRELEASE);

      /* 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 = 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;
      }
}

