/* 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) */ inline bool Voice::updateAmpInc(unsigned int &nextNewAmpInc, std::map::iterator &curSample2AmpInc, qreal &dsp_amp_incr, unsigned int &dsp_i) { if (positionToTurnOff > 0 && dsp_i >= (unsigned int) positionToTurnOff) return false; // if volume is zero skip all phases that do not change that! if (amp == 0.0f) { while (dsp_amp_incr == 0.0f && curSample2AmpInc != Sample2AmpInc.end()) { dsp_i = curSample2AmpInc->first; curSample2AmpInc++; nextNewAmpInc = curSample2AmpInc->first; dsp_amp_incr = curSample2AmpInc->second; } if (curSample2AmpInc == Sample2AmpInc.end()) return false; } if (dsp_i >= nextNewAmpInc) { curSample2AmpInc++; nextNewAmpInc = curSample2AmpInc->first; dsp_amp_incr = curSample2AmpInc->second; } return true; } /* 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; auto curSample2AmpInc = Sample2AmpInc.begin(); qreal dsp_amp_incr = curSample2AmpInc->second; unsigned int nextNewAmpInc = curSample2AmpInc->first; 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] = 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 */ if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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; 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; auto curSample2AmpInc = Sample2AmpInc.begin(); qreal dsp_amp_incr = curSample2AmpInc->second; unsigned int nextNewAmpInc = curSample2AmpInc->first; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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; 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; auto curSample2AmpInc = Sample2AmpInc.begin(); qreal dsp_amp_incr = curSample2AmpInc->second; unsigned int nextNewAmpInc = curSample2AmpInc->first; 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)]; auto val = 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]); dsp_buf[dsp_i] = val; /* increment phase and amplitude */ phase += dsp_phase_incr; dsp_phase_index = phase.index(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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)]; auto val = 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]); dsp_buf[dsp_i] = val; /* increment phase and amplitude */ phase += dsp_phase_incr; dsp_phase_index = phase.index(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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)]; auto val = 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); dsp_buf[dsp_i] = val; /* increment phase and amplitude */ phase += dsp_phase_incr; dsp_phase_index = phase.index(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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)]; auto val = amp * (coeffs[0] * dsp_data[dsp_phase_index-1] + coeffs[1] * dsp_data[dsp_phase_index] + coeffs[2] * end_point1 + coeffs[3] * end_point2); dsp_buf[dsp_i] = val; /* increment phase and amplitude */ phase += dsp_phase_incr; dsp_phase_index = phase.index(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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; auto curSample2AmpInc = Sample2AmpInc.begin(); qreal dsp_amp_incr = curSample2AmpInc->second; unsigned int nextNewAmpInc = curSample2AmpInc->first; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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] = 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(); if (!updateAmpInc(nextNewAmpInc, curSample2AmpInc, dsp_amp_incr, dsp_i)) return dsp_i; 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; return dsp_i; } }