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authorHans-Christoph Steiner <eighthave@users.sourceforge.net>2006-05-25 16:40:19 +0000
committerHans-Christoph Steiner <eighthave@users.sourceforge.net>2006-05-25 16:40:19 +0000
commitd962ed33f6b5fe3040cfbd798f63ab9aa36d1f9e (patch)
treef700234efa8d57c28ce9be56e0db5f6526921fbe /modules/eblosc.c
parent2994949cca515f63fb82fa1d114ce07eb213220d (diff)
renamed files to match their class names
svn path=/trunk/externals/creb/; revision=5127
Diffstat (limited to 'modules/eblosc.c')
-rw-r--r--modules/eblosc.c599
1 files changed, 0 insertions, 599 deletions
diff --git a/modules/eblosc.c b/modules/eblosc.c
deleted file mode 100644
index df1b059..0000000
--- a/modules/eblosc.c
+++ /dev/null
@@ -1,599 +0,0 @@
-/*
- * eblosc.c - bandlimited oscillators with infinite support discontinuities
- * using minimum phase impulse, step & ramp
- * Copyright (c) 2000-2003 by Tom Schouten
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-
-#include "m_pd.h"
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "filters.h"
-
-
-typedef unsigned long long u64;
-typedef unsigned long u32;
-
-
-
-#define LPHASOR (8*sizeof(u32)) // the phasor logsize
-#define VOICES 8 // the number of oscillators
-#define CUTOFF 0.8f // fraction of nyquist for impulse cutoff
-
-
-
-typedef struct ebloscctl
-{
- t_float c_pole[VOICES*2]; // complex poles
- t_float c_gain[VOICES*2]; // complex gains (waveform specific constants)
- t_float c_state[VOICES*2]; // complex state
-
- u32 c_phase; // phase of main oscillator
- u32 c_phase2; // phase of secondairy oscillator
- t_float c_prev_amp; // previous input of comparator
- t_float c_phase_inc_scale;
- t_float c_scale;
- t_float c_scale_update;
- t_symbol *c_waveform;
-
-} t_ebloscctl;
-
-typedef struct eblosc
-{
- t_object x_obj;
- t_float x_f;
- t_ebloscctl x_ctl;
-} t_eblosc;
-
-
-/* phase converters */
-static inline float _phase_to_float(u32 p){return ((float)p) * (1.0f / 4294967296.0f);}
-static inline u32 _float_to_phase(float f){return (u32)(f * 4294967296.0f);}
-
-
-
-/* get one sample from the oscillator bank and perform time tick */
-static inline t_float _osc_tick(t_ebloscctl *ctl)
-{
- float sum = 0.0f;
- int i;
- /* sum all voices */
- for (i=0; i<VOICES*2; i+=2){
- /* rotate state */
- vcmul2(ctl->c_state+i, ctl->c_pole+i);
-
- /* get real part and add to output */
- sum += ctl->c_state[0];
- }
-
- return sum;
-}
-
-/* add shifted impulse */
-static inline void _add_impulse(t_ebloscctl *ctl, t_float t0)
-{
- int i;
- for (i=0; i<VOICES*2; i+=2){
- /* contribution is a_i z_i^t_0 */
-
- float real = 1.0f;
- float imag = 0.0f;
-
- ctl->c_state[0] += real;
- ctl->c_state[1] += imag;
- }
-}
-
-
-/* add step */
-static inline void _add_step(t_ebloscctl *ctl)
-{
- int i;
- for (i=0; i<VOICES*2; i+=2){
- /* contribution is a_i (1 - z_i) */
-
- float real = 1.0f;
- float imag = 0.0f;
-
- ctl->c_state[0] += real;
- ctl->c_state[1] += imag;
- }
-}
-
-
-/* add shifted step */
-static inline void _add_shifted_step(t_ebloscctl *ctl, t_float t0)
-{
- int i;
- for (i=0; i<VOICES*2; i+=2){
- /* contribution is a_i (1 - z_i^t_0) */
-
- float real = 1.0f;
- float imag = 0.0f;
-
- ctl->c_state[0] += real;
- ctl->c_state[1] += imag;
- }
-}
-
-
-#if 0
-/* update waveplayers on zero cross */
-static void _bang_comparator(t_ebloscctl *ctl, float prev, float curr)
-{
-
- /* check for sign change */
- if ((prev * curr) < 0.0f){
-
- int voice;
-
- /* determine the location of the discontinuity (in oversampled coordiates
- using linear interpolation */
-
- float f = (float)S * curr / (curr - prev);
-
- /* get the offset in the oversample table */
-
- u32 table_index = (u32)f;
-
- /* determine the fractional part (in oversampled coordinates)
- for linear interpolation */
-
- float table_frac_index = f - (float)table_index;
-
- /* set state (+ or -) */
-
- ctl->c_state = (curr > 0.0f) ? 0.5f : -0.5f;
-
- /* steal the oldest voice */
-
- voice = ctl->c_next_voice++;
- ctl->c_next_voice &= VOICES-1;
-
- /* initialize the new voice index and interpolation fraction */
-
- ctl->c_index[voice] = table_index;
- ctl->c_frac[voice] = table_frac_index;
- ctl->c_vscale[voice] = -ctl->c_scale * 2.0f * ctl->c_state;
-
- }
-
-}
-
-/* advance phasor and update waveplayers on phase wrap */
-static void _bang_phasor(t_ebloscctl *ctl, float freq)
-{
- u32 phase = ctl->c_phase;
- u32 phase_inc;
- u32 oldphase;
- int voice;
- float scale = ctl->c_scale;
-
- /* get increment */
- float inc = freq * ctl->c_phase_inc_scale;
-
- /* calculate new phase
- the increment (and the phase) should be a multiple of S */
- if (inc < 0.0f) inc = -inc;
- phase_inc = ((u32)inc) & ~(S-1);
- oldphase = phase;
- phase += phase_inc;
-
-
- /* check for phase wrap */
- if (phase < oldphase){
- u32 phase_inc_decimated = phase_inc >> LOVERSAMPLE;
- u32 table_index;
- u32 table_phase;
-
- /* steal the oldest voice if we have a phase wrap */
-
- voice = ctl->c_next_voice++;
- ctl->c_next_voice &= VOICES-1;
-
- /* determine the location of the discontinuity (in oversampled coordinates)
- which is S * (new phase) / (increment) */
-
- table_index = phase / phase_inc_decimated;
-
- /* determine the fractional part (in oversampled coordinates)
- for linear interpolation */
-
- table_phase = phase - (table_index * phase_inc_decimated);
-
- /* use it to initialize the new voice index and interpolation fraction */
-
- ctl->c_index[voice] = table_index;
- ctl->c_frac[voice] = (float)table_phase / (float)phase_inc_decimated;
- ctl->c_vscale[voice] = scale;
- scale = scale * ctl->c_scale_update;
-
- }
-
- /* save state */
- ctl->c_phase = phase;
- ctl->c_scale = scale;
-}
-
-
-/* the 2 oscillator version:
- the second osc can reset the first osc's phase (hence it determines the pitch)
- the first osc determines the waveform */
-
-static void _bang_hardsync_phasor(t_ebloscctl *ctl, float freq, float freq2)
-{
- u32 phase = ctl->c_phase;
- u32 phase2 = ctl->c_phase2;
- u32 phase_inc;
- u32 phase_inc2;
- u32 oldphase;
- u32 oldphase2;
- int voice;
- float scale = ctl->c_scale;
-
-
- /* get increment */
- float inc = freq * ctl->c_phase_inc_scale;
- float inc2 = freq2 * ctl->c_phase_inc_scale;
-
- /* calculate new phases
- the increment (and the phase) should be a multiple of S */
-
- /* save previous phases */
- oldphase = phase;
- oldphase2 = phase2;
-
- /* update second osc */
- if (inc2 < 0.0f) inc2 = -inc2;
- phase_inc2 = ((u32)inc2) & ~(S-1);
- phase2 += phase_inc2;
-
- /* update first osc (freq should be >= freq of sync osc */
- if (inc < 0.0f) inc = -inc;
- phase_inc = ((u32)inc) & ~(S-1);
- if (phase_inc < phase_inc2) phase_inc = phase_inc2;
- phase += phase_inc;
-
-
- /* check for sync discontinuity (osc 2) */
- if (phase2 < oldphase2) {
-
- /* adjust phase depending on the location of the discontinuity in phase2:
- phase/phase_inc == phase2/phase_inc2 */
-
- u64 pi = phase_inc >> LOVERSAMPLE;
- u64 pi2 = phase_inc2 >> LOVERSAMPLE;
- u64 lphase = ((u64)phase2 * pi) / pi2;
- phase = lphase & ~(S-1);
- }
-
-
- /* check for phase discontinuity (osc 1) */
- if (phase < oldphase){
- u32 phase_inc_decimated = phase_inc >> LOVERSAMPLE;
- u32 table_index;
- u32 table_phase;
- float stepsize;
-
- /* steal the oldest voice if we have a phase wrap */
-
- voice = ctl->c_next_voice++;
- ctl->c_next_voice &= VOICES-1;
-
- /* determine the location of the discontinuity (in oversampled coordinates)
- which is S * (new phase) / (increment) */
-
- table_index = phase / phase_inc_decimated;
-
- /* determine the fractional part (in oversampled coordinates)
- for linear interpolation */
-
- table_phase = phase - (table_index * phase_inc_decimated);
-
- /* determine the step size
- as opposed to saw/impulse waveforms, the step is not always equal to one. it is:
- oldphase - phase + phase_inc
- but for the unit step this will overflow to zero, so we
- reduce the bit depth to prevent overflow */
-
- stepsize = _phase_to_float(((oldphase-phase) >> LOVERSAMPLE)
- + phase_inc_decimated) * (float)S;
-
- /* use it to initialize the new voice index and interpolation fraction */
-
- ctl->c_index[voice] = table_index;
- ctl->c_frac[voice] = (float)table_phase / (float)phase_inc_decimated;
- ctl->c_vscale[voice] = scale * stepsize;
- scale = scale * ctl->c_scale_update;
-
- }
-
- /* save state */
- ctl->c_phase = phase;
- ctl->c_phase2 = phase2;
- ctl->c_scale = scale;
-}
-
-
-static t_int *eblosc_perform_hardsync_saw(t_int *w)
-{
- t_float *freq = (float *)(w[3]);
- t_float *freq2 = (float *)(w[4]);
- t_float *out = (float *)(w[5]);
- t_ebloscctl *ctl = (t_ebloscctl *)(w[1]);
- t_int n = (t_int)(w[2]);
- t_int i;
-
- /* set postfilter cutoff */
- ctl->c_butter->setButterHP(0.85f * (*freq / sys_getsr()));
-
- while (n--) {
- float frequency = *freq++;
- float frequency2 = *freq2++;
-
- /* get the bandlimited discontinuity */
- float sample = _get_bandlimited_discontinuity(ctl, bls);
-
- /* add aliased sawtooth wave */
- sample += _phase_to_float(ctl->c_phase) - 0.5f;
-
- /* highpass filter output to remove DC offset and low frequency aliasing */
- ctl->c_butter->BangSmooth(sample, sample, 0.05f);
-
- /* send to output */
- *out++ = sample;
-
- /* advance phasor */
- _bang_hardsync_phasor(ctl, frequency2, frequency);
-
- }
-
- return (w+6);
-}
-
-static t_int *eblosc_perform_saw(t_int *w)
-{
- t_float *freq = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
- t_ebloscctl *ctl = (t_ebloscctl *)(w[1]);
- t_int n = (t_int)(w[2]);
- t_int i;
-
- while (n--) {
- float frequency = *freq++;
-
- /* get the bandlimited discontinuity */
- float sample = _get_bandlimited_discontinuity(ctl, bls);
-
- /* add aliased sawtooth wave */
- sample += _phase_to_float(ctl->c_phase) - 0.5f;
-
- /* send to output */
- *out++ = sample;
-
- /* advance phasor */
- _bang_phasor(ctl, frequency);
-
- }
-
- return (w+5);
-}
-
-
-
-static t_int *eblosc_perform_pulse(t_int *w)
-{
- t_float *freq = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
- t_ebloscctl *ctl = (t_ebloscctl *)(w[1]);
- t_int n = (t_int)(w[2]);
- t_int i;
-
-
- /* set postfilter cutoff */
- ctl->c_butter->setButterHP(0.85f * (*freq / sys_getsr()));
-
- while (n--) {
- float frequency = *freq++;
-
- /* get the bandlimited discontinuity */
- float sample = _get_bandlimited_discontinuity(ctl, bli);
-
- /* highpass filter output to remove DC offset and low frequency aliasing */
- ctl->c_butter->BangSmooth(sample, sample, 0.05f);
-
- /* send to output */
- *out++ = sample;
-
- /* advance phasor */
- _bang_phasor(ctl, frequency);
-
- }
-
- return (w+5);
-}
-
-static t_int *eblosc_perform_comparator(t_int *w)
-{
- t_float *amp = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
- t_ebloscctl *ctl = (t_ebloscctl *)(w[1]);
- t_int n = (t_int)(w[2]);
- t_int i;
- t_float prev_amp = ctl->c_prev_amp;
-
- while (n--) {
- float curr_amp = *amp++;
-
- /* exact zero won't work for zero detection (sic) */
- if (curr_amp == 0.0f) curr_amp = 0.0000001f;
-
- /* get the bandlimited discontinuity */
- float sample = _get_bandlimited_discontinuity(ctl, bls);
-
- /* add the block wave state */
- sample += ctl->c_state;
-
- /* send to output */
- *out++ = sample;
-
- /* advance phasor */
- _bang_comparator(ctl, prev_amp, curr_amp);
-
- prev_amp = curr_amp;
-
- }
-
- ctl->c_prev_amp = prev_amp;
-
- return (w+5);
-}
-
-static void eblosc_phase(t_eblosc *x, t_float f)
-{
- x->x_ctl.c_phase = _float_to_phase(f);
- x->x_ctl.c_phase2 = _float_to_phase(f);
-}
-
-static void eblosc_phase1(t_eblosc *x, t_float f)
-{
- x->x_ctl.c_phase = _float_to_phase(f);
-}
-
-static void eblosc_phase2(t_eblosc *x, t_float f)
-{
- x->x_ctl.c_phase2 = _float_to_phase(f);
-}
-
-static void eblosc_dsp(t_eblosc *x, t_signal **sp)
-{
- int n = sp[0]->s_n;
-
- /* set sampling rate scaling for phasors */
- x->x_ctl.c_phase_inc_scale = 4.0f * (float)(1<<(LPHASOR-2)) / sys_getsr();
-
-
- /* setup & register the correct process routine depending on the waveform */
-
- /* 2 osc */
- if (x->x_ctl.c_waveform == gensym("syncsaw")){
- x->x_ctl.c_scale = 1.0f;
- x->x_ctl.c_scale_update = 1.0f;
- dsp_add(eblosc_perform_hardsync_saw, 5, &x->x_ctl, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec, sp[2]->s_vec);
- }
-
- /* 1 osc */
- else if (x->x_ctl.c_waveform == gensym("pulse")){
- x->x_ctl.c_scale = 1.0f;
- x->x_ctl.c_scale_update = 1.0f;
- dsp_add(eblosc_perform_pulse, 4, &x->x_ctl, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
- }
- else if (x->x_ctl.c_waveform == gensym("pulse2")){
- x->x_ctl.c_phase_inc_scale *= 2;
- x->x_ctl.c_scale = 1.0f;
- x->x_ctl.c_scale_update = -1.0f;
- dsp_add(eblosc_perform_pulse, 4, &x->x_ctl, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
- }
- else if (x->x_ctl.c_waveform == gensym("comparator")){
- x->x_ctl.c_scale = 1.0f;
- x->x_ctl.c_scale_update = 1.0f;
- dsp_add(eblosc_perform_comparator, 4, &x->x_ctl, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
- }
- else{
- x->x_ctl.c_scale = 1.0f;
- x->x_ctl.c_scale_update = 1.0f;
- dsp_add(eblosc_perform_saw, 4, &x->x_ctl, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
- }
-
-
-
-}
-static void eblosc_free(t_eblosc *x)
-{
- delete x->x_ctl.c_butter;
-}
-
-t_class *eblosc_class;
-
-static void *eblosc_new(t_symbol *s)
-{
- t_eblosc *x = (t_eblosc *)pd_new(eblosc_class);
- int i;
-
- /* out 1 */
- outlet_new(&x->x_obj, gensym("signal"));
-
- /* optional signal inlets */
- if (s == gensym("syncsaw")){
- inlet_new(&x->x_obj, &x->x_obj.ob_pd, gensym("signal"), gensym("signal"));
- }
-
- /* optional phase inlet */
- if (s != gensym("comparator")){
- inlet_new(&x->x_obj, &x->x_obj.ob_pd, gensym("float"), gensym("phase"));
- }
-
- /* create the postfilter */
- x->x_ctl.c_butter = new DSPIfilterSeries(3);
-
- /* init oscillators */
- for (i=0; i<VOICES; i++) {
- x->x_ctl.c_index[i] = N-2;
- x->x_ctl.c_frac[i] = 0.0f;
- }
-
- /* init rest of state data */
- eblosc_phase(x, 0);
- eblosc_phase2(x, 0);
- x->x_ctl.c_state = 0.0;
- x->x_ctl.c_prev_amp = 0.0;
- x->x_ctl.c_next_voice = 0;
- x->x_ctl.c_scale = 1.0f;
- x->x_ctl.c_scale_update = 1.0f;
- x->x_ctl.c_waveform = s;
-
- return (void *)x;
-}
-
-
-
-
-
-extern "C"
-{
- void eblosc_tilde_setup(void)
- {
- //post("eblosc~ v0.1");
-
- build_tables();
-
- eblosc_class = class_new(gensym("eblosc~"), (t_newmethod)eblosc_new,
- (t_method)eblosc_free, sizeof(t_eblosc), 0, A_DEFSYMBOL, A_NULL);
- CLASS_MAINSIGNALIN(eblosc_class, t_eblosc, x_f);
- class_addmethod(eblosc_class, (t_method)eblosc_dsp, gensym("dsp"), A_NULL);
- class_addmethod(eblosc_class, (t_method)eblosc_phase, gensym("phase"), A_FLOAT, A_NULL);
- class_addmethod(eblosc_class, (t_method)eblosc_phase2, gensym("phase2"), A_FLOAT, A_NULL);
-
-
- }
-
-}
-
-#endif