[PD-cvs] SF.net SVN: pure-data: [9877] branches/pd-extended/v0-40/externals/creb/ modules++
eighthave at users.sourceforge.net
eighthave at users.sourceforge.net
Sat May 24 00:44:44 CEST 2008
Revision: 9877
http://pure-data.svn.sourceforge.net/pure-data/?rev=9877&view=rev
Author: eighthave
Date: 2008-05-23 15:44:44 -0700 (Fri, 23 May 2008)
Log Message:
-----------
renamed files so that the filenames match the classnames
Added Paths:
-----------
branches/pd-extended/v0-40/externals/creb/modules++/biquadseries~.cc
branches/pd-extended/v0-40/externals/creb/modules++/blosc~.cc
branches/pd-extended/v0-40/externals/creb/modules++/filterortho~.cc
Removed Paths:
-------------
branches/pd-extended/v0-40/externals/creb/modules++/biquadseries.cc
branches/pd-extended/v0-40/externals/creb/modules++/blosc.cc
branches/pd-extended/v0-40/externals/creb/modules++/filterortho.cc
Deleted: branches/pd-extended/v0-40/externals/creb/modules++/biquadseries.cc
===================================================================
--- branches/pd-extended/v0-40/externals/creb/modules++/biquadseries.cc 2008-05-23 22:40:19 UTC (rev 9876)
+++ branches/pd-extended/v0-40/externals/creb/modules++/biquadseries.cc 2008-05-23 22:44:44 UTC (rev 9877)
@@ -1,128 +0,0 @@
-/*
- * biquadseries.cc - second order section filter pd interface
- * 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 "DSPIcomplex.h"
-#include "DSPIfilters.h"
-
-
-
-typedef struct biquadseries_struct
-{
- t_object x_obj;
- t_float x_f;
- DSPIfilterSeries* biquadseries;
-} t_biquadseries;
-
-void biquadseries_bang(t_biquadseries *x)
-{
-
-}
-
-void biquadseries_butterLP(t_biquadseries *x, t_floatarg f)
-{
- x->biquadseries->setButterLP(f / sys_getsr());
-}
-
-void biquadseries_butterHP(t_biquadseries *x, t_floatarg f)
-{
- x->biquadseries->setButterHP(f / sys_getsr());
-}
-
-
-
-static t_int *biquadseries_perform(t_int *w)
-{
-
-
- t_float *in = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
- DSPIfilterSeries* biquadseries = (DSPIfilterSeries *)(w[1]);
- t_int n = (t_int)(w[2]);
- t_int i;
- t_float x;
-
- // dit kan beter
- float smooth = .01;
- //1.0f - pow(.9f,1.0f/(float)(n));
-
- for (i = 0; i < n; i++)
- {
- x = *in++;
- biquadseries->BangSmooth(x, x, smooth);
- *out++ = x;
- }
-
- return (w+5);
-}
-
-static void biquadseries_dsp(t_biquadseries *x, t_signal **sp)
-{
- dsp_add(biquadseries_perform, 4, x->biquadseries, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
-
-}
-void biquadseries_free(void)
-{
-
-}
-
-t_class *biquadseries_class;
-
-void *biquadseries_new(t_floatarg fsections)
-{
- t_biquadseries *x = (t_biquadseries *)pd_new(biquadseries_class);
-
- int sections = (int)fsections;
- if (sections < 1) sections = 1;
- // post("biquadseries~: %d sections", sections);
- x->biquadseries = new DSPIfilterSeries(sections);
-
- // inlet_new(&x->x_obj, &x->x_obj.ob_pd, gensym("float"), gensym("freq"));
- outlet_new(&x->x_obj, gensym("signal"));
-
- biquadseries_butterLP(x, 10000);
-
- return (void *)x;
-}
-
-
-extern "C" {
-
-void biquadseries_tilde_setup(void)
-{
- //post("biquadseries~ v0.1");
-
- biquadseries_class = class_new(gensym("biquadseries~"), (t_newmethod)biquadseries_new,
- (t_method)biquadseries_free, sizeof(t_biquadseries), 0, A_DEFFLOAT, 0);
-
- CLASS_MAINSIGNALIN(biquadseries_class, t_biquadseries, x_f);
-
- class_addmethod(biquadseries_class, (t_method)biquadseries_bang, gensym("bang"), (t_atomtype)0);
-
- class_addmethod(biquadseries_class, (t_method)biquadseries_dsp, gensym("dsp"), (t_atomtype)0);
-
- class_addmethod(biquadseries_class, (t_method)biquadseries_butterLP, gensym("butterLP"), A_FLOAT, A_NULL);
- class_addmethod(biquadseries_class, (t_method)biquadseries_butterHP, gensym("butterHP"), A_FLOAT, A_NULL);
-
-}
-
-}
Copied: branches/pd-extended/v0-40/externals/creb/modules++/biquadseries~.cc (from rev 9861, branches/pd-extended/v0-40/externals/creb/modules++/biquadseries.cc)
===================================================================
--- branches/pd-extended/v0-40/externals/creb/modules++/biquadseries~.cc (rev 0)
+++ branches/pd-extended/v0-40/externals/creb/modules++/biquadseries~.cc 2008-05-23 22:44:44 UTC (rev 9877)
@@ -0,0 +1,128 @@
+/*
+ * biquadseries.cc - second order section filter pd interface
+ * 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 "DSPIcomplex.h"
+#include "DSPIfilters.h"
+
+
+
+typedef struct biquadseries_struct
+{
+ t_object x_obj;
+ t_float x_f;
+ DSPIfilterSeries* biquadseries;
+} t_biquadseries;
+
+void biquadseries_bang(t_biquadseries *x)
+{
+
+}
+
+void biquadseries_butterLP(t_biquadseries *x, t_floatarg f)
+{
+ x->biquadseries->setButterLP(f / sys_getsr());
+}
+
+void biquadseries_butterHP(t_biquadseries *x, t_floatarg f)
+{
+ x->biquadseries->setButterHP(f / sys_getsr());
+}
+
+
+
+static t_int *biquadseries_perform(t_int *w)
+{
+
+
+ t_float *in = (float *)(w[3]);
+ t_float *out = (float *)(w[4]);
+ DSPIfilterSeries* biquadseries = (DSPIfilterSeries *)(w[1]);
+ t_int n = (t_int)(w[2]);
+ t_int i;
+ t_float x;
+
+ // dit kan beter
+ float smooth = .01;
+ //1.0f - pow(.9f,1.0f/(float)(n));
+
+ for (i = 0; i < n; i++)
+ {
+ x = *in++;
+ biquadseries->BangSmooth(x, x, smooth);
+ *out++ = x;
+ }
+
+ return (w+5);
+}
+
+static void biquadseries_dsp(t_biquadseries *x, t_signal **sp)
+{
+ dsp_add(biquadseries_perform, 4, x->biquadseries, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
+
+}
+void biquadseries_free(void)
+{
+
+}
+
+t_class *biquadseries_class;
+
+void *biquadseries_new(t_floatarg fsections)
+{
+ t_biquadseries *x = (t_biquadseries *)pd_new(biquadseries_class);
+
+ int sections = (int)fsections;
+ if (sections < 1) sections = 1;
+ // post("biquadseries~: %d sections", sections);
+ x->biquadseries = new DSPIfilterSeries(sections);
+
+ // inlet_new(&x->x_obj, &x->x_obj.ob_pd, gensym("float"), gensym("freq"));
+ outlet_new(&x->x_obj, gensym("signal"));
+
+ biquadseries_butterLP(x, 10000);
+
+ return (void *)x;
+}
+
+
+extern "C" {
+
+void biquadseries_tilde_setup(void)
+{
+ //post("biquadseries~ v0.1");
+
+ biquadseries_class = class_new(gensym("biquadseries~"), (t_newmethod)biquadseries_new,
+ (t_method)biquadseries_free, sizeof(t_biquadseries), 0, A_DEFFLOAT, 0);
+
+ CLASS_MAINSIGNALIN(biquadseries_class, t_biquadseries, x_f);
+
+ class_addmethod(biquadseries_class, (t_method)biquadseries_bang, gensym("bang"), (t_atomtype)0);
+
+ class_addmethod(biquadseries_class, (t_method)biquadseries_dsp, gensym("dsp"), (t_atomtype)0);
+
+ class_addmethod(biquadseries_class, (t_method)biquadseries_butterLP, gensym("butterLP"), A_FLOAT, A_NULL);
+ class_addmethod(biquadseries_class, (t_method)biquadseries_butterHP, gensym("butterHP"), A_FLOAT, A_NULL);
+
+}
+
+}
Deleted: branches/pd-extended/v0-40/externals/creb/modules++/blosc.cc
===================================================================
--- branches/pd-extended/v0-40/externals/creb/modules++/blosc.cc 2008-05-23 22:40:19 UTC (rev 9876)
+++ branches/pd-extended/v0-40/externals/creb/modules++/blosc.cc 2008-05-23 22:44:44 UTC (rev 9877)
@@ -1,782 +0,0 @@
-/*
- * blosc.c - bandlimited oscillators
- * 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 "DSPIcomplex.h"
-#include "DSPIfilters.h"
-
-/* work around old bug in Apple compilers <hans at at.or.at> */
-//#if MAC_OS_X_VERSION < MAC_OS_X_VERSION_10_4 // arg, this didn't work
-#ifndef isnan
-extern "C" int isnan(double);
-#endif
-#ifndef isinf
-extern "C" int isinf(double);
-#endif
-
-typedef unsigned long long u64;
-typedef unsigned long u32;
-
-
-
-#define LPHASOR (8*sizeof(u32)) // the phasor logsize
-#define VOICES 8 // the number of waveform voices
-#define LLENGTH 6 // the loglength of a fractional delayed basic waveform
-#define LOVERSAMPLE 6 // the log of the oversampling factor (nb of fract delayed waveforms)
-#define LPAD 1 // the log of the time padding factor (to reduce time aliasing)
-#define LTABLE (LLENGTH+LOVERSAMPLE)
-#define N (1<<LTABLE)
-#define M (1<<(LTABLE+LPAD))
-#define S (1<<LOVERSAMPLE)
-#define L (1<<LLENGTH)
-#define LMASK (L-1)
-
-#define WALPHA 0.1f // windowing alpha (0 = cos -> 1 = rect)
-#define CUTOFF 0.8f // fraction of nyquist for impulse cutoff
-#define NBPERIODS ((float)(L) * CUTOFF / 2.0f)
-
-/* sample buffers */
-static float bli[N]; // band limited impulse
-static float bls[N]; // band limited step
-static float blr[N]; // band limited ramp
-
-
-typedef struct bloscctl
-{
- t_int c_index[VOICES]; // array of indices in sample table
- t_float c_frac[VOICES]; // array of fractional indices
- t_float c_vscale[VOICES]; // array of scale factors
- t_int c_next_voice; // next voice to steal (round robin)
- u32 c_phase; // phase of main oscillator
- u32 c_phase2; // phase of secondairy oscillator
- t_float c_state; // state of the square wave
- t_float c_prev_amp; // previous input of comparator
- t_float c_phase_inc_scale;
- t_float c_scale;
- t_float c_scale_update;
- DSPIfilterSeries* c_butter; // the series filter
- t_symbol *c_waveform;
-
-} t_bloscctl;
-
-typedef struct blosc
-{
- t_object x_obj;
- t_float x_f;
- t_bloscctl x_ctl;
-} t_blosc;
-
-
-/* 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)) & ~(S-1);}
-
-
-/* flat table: better for linear interpolation */
-static inline float _play_voice_lint(float *table, t_int *index, float frac, float scale)
-{
- t_int i = *index;
-
- /* perform linear interpolation */
- float f = (((1.0f - frac) * table[i]) + (table[i+1] * frac)) * scale;
-
- /* increment phase index if next 2 elements will still be inside table
- if not there's no increment and the voice will keep playing the same sample */
-
- i += (((i+S+1) >> LTABLE) ^ 1) << LOVERSAMPLE;
-
- *index = i;
- return f;
-}
-
-/* get one sample from the bandlimited discontinuity wavetable playback syth */
-static inline t_float _get_bandlimited_discontinuity(t_bloscctl *ctl, float *table)
-{
- float sum = 0.0f;
- int i;
- /* sum all voices */
- for (i=0; i<VOICES; i++){
- sum += _play_voice_lint(table, ctl->c_index+i, ctl->c_frac[i], ctl->c_vscale[i]);
- }
-
- return sum;
-}
-
-
-/* update waveplayers on zero cross */
-static void _bang_comparator(t_bloscctl *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_bloscctl *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_bloscctl *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 *blosc_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_bloscctl *ctl = (t_bloscctl *)(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 *blosc_perform_saw(t_int *w)
-{
- t_float *freq = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
- t_bloscctl *ctl = (t_bloscctl *)(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 *blosc_perform_pulse(t_int *w)
-{
- t_float *freq = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
- t_bloscctl *ctl = (t_bloscctl *)(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 *blosc_perform_comparator(t_int *w)
-{
- t_float *amp = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
- t_bloscctl *ctl = (t_bloscctl *)(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 blosc_phase(t_blosc *x, t_float f)
-{
- x->x_ctl.c_phase = _float_to_phase(f);
- x->x_ctl.c_phase2 = _float_to_phase(f);
-}
-
-static void blosc_phase1(t_blosc *x, t_float f)
-{
- x->x_ctl.c_phase = _float_to_phase(f);
-}
-
-static void blosc_phase2(t_blosc *x, t_float f)
-{
- x->x_ctl.c_phase2 = _float_to_phase(f);
-}
-
-static void blosc_dsp(t_blosc *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(blosc_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(blosc_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(blosc_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(blosc_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(blosc_perform_saw, 4, &x->x_ctl, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
- }
-
-
-
-}
-static void blosc_free(t_blosc *x)
-{
- delete x->x_ctl.c_butter;
-}
-
-t_class *blosc_class;
-
-static void *blosc_new(t_symbol *s)
-{
- t_blosc *x = (t_blosc *)pd_new(blosc_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 */
- blosc_phase(x, 0);
- blosc_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;
-}
-
-
-
-
-
-
-
-/* CLASS DATA INIT (tables) */
-
-
-/* some vector ops */
-
-/* clear a buffer */
-static inline void _clear(float *array, int size)
-{
- memset(array, 0, sizeof(float)*size);
-}
-
-/* compute complex log */
-static inline void _clog(float *real, float *imag, int size)
-{
- int k;
- for (k=0; k<size; k++){
- float r = real[k];
- float i = imag[k];
- float radius = sqrt(r*r+i*i);
- real[k] = log(radius);
- imag[k] = atan2(i,r);
- }
-}
-
-/* compute complex exp */
-static inline void _cexp(float *real, float *imag, int size)
-{
- int k;
- for (k=0; k<size; k++){
- float r = exp(real[k]);
- float i = imag[k];
- real[k] = r * cos(i);
- imag[k] = r * sin(i);
- }
-}
-
-
-/* compute fft */
-static inline void _fft(float *real, float *imag, int size)
-{
- int i;
- float scale = 1.0f / sqrt((float)size);
- for (i=0; i<size; i++){
- real[i] *= scale;
- imag[i] *= scale;
- if (isnan(real[i])) post("fftpanic %d", i);
- }
- mayer_fft(size, real, imag);
-}
-/* compute ifft */
-static inline void _ifft(float *real, float *imag, int size)
-{
- int i;
- float scale = 1.0f / sqrt((float)size);
- for (i=0; i<size; i++){
- real[i] *= scale;
- imag[i] *= scale;
- if (isnan(real[i])) post("ifftpanic %d", i);
- }
- mayer_ifft(size, real, imag);
-}
-
-/* convert an integer index to a phase: [0 -> pi, -pi -> 0] */
-static inline float _i2theta(int i, int size){
- float p = 2.0f * M_PI * (float)i / (float)size;
- if (p >= M_PI) p -= 2.0f * M_PI;
- return p;
-}
-
-
-/* print matlab array */
-static void _printm(float *array, char *name, int size)
-{
- int i;
- fprintf(stderr, "%s = [", name);
- for (i=0; i<size; i++){
- fprintf(stderr, "%f;", array[i]);
- }
- fprintf(stderr, "];\n");
-}
-
-/* store oversampled waveform as decimated chunks */
-static void _store_decimated(float *dst, float *src, float scale, int size)
-{
- int i;
- for (i=0; i<size; i++){
- int offset = (i % S) * L;
- int index = i / S;
- dst[offset+index] = scale * src[i];
- }
-
-}
-
-/* store waveform as one chunk */
-static void _store(float *dst, float *src, float scale, int size)
-{
- int i;
- for (i=0; i<size; i++){
- dst[i] = scale * src[i];
- }
-
-}
-
-/* create a minimum phase bandlimited impulse */
-static void build_tables(void)
-{
-
- /* table size = M>=N (time padding to reduce time aliasing) */
-
- /* we work in the complex domain to eliminate the need to avoid
- negative spectral components */
-
- float real[M];
- float imag[M];
- float sum,scale;
- int i,j;
-
-
- /* create windowed sinc */
- _clear(imag, M);
- real[0] = 1.0f;
- for (i=1; i<M; i++){
- float tw = _i2theta(i,M);
- float ts = tw * NBPERIODS * (float)(M) / (float)(N);
-
- /* sinc */
- real[i] = sin(ts)/ts;
-
- /* blackman window */
- real[i] *= 0.42f + 0.5f * (cos(tw)) + 0.08f * (cos(2.0f*tw));
-
- //real[i] *= 0.5f * (1.0f + WALPHA) + 0.5f * (1.0f - WALPHA) * (cos(tw));
-
- /* check for nan */
- if (isnan(real[i])) post("sinc NaN panic %d", i);
- if (isinf(real[i])) post("sinc Inf panic %d", i);
-
- }
-
-
- /* compute cepstrum */
- _fft(real, imag, M);
- _clog(real, imag, M);
- _ifft(real, imag, M);
-
-
- /* kill anti-causal part (contribution of non minimum phase zeros) */
- /* should we kill nyquist too ?? */
- for (i=M/2+1; i<M; i++){
- real[i] *= 0.0000f;
- imag[i] *= 0.0000f;
- }
-
-
- /* compute inverse cepstrum */
- _fft(real, imag, M);
- _cexp(real, imag, M);
- _ifft(real, imag, M);
-
-
-
- /* from here on, discard the padded part [N->M-1]
- and work with the first N samples */
-
- /* normalize impulse (integral = 1) */
- sum = 0.0f;
- for (i=0; i<N; i++){sum += real[i];}
- scale = 1.0f / sum;
- for (i=0; i<N; i++){real[i] *= scale;}
-
-
- /* store bli table */
- _store(bli, real, (float)S, N);
- //_printm(bli, "h", N);
-
-
- /* integrate impulse and invert to produce a step function
- from 1->0 */
- sum = 0.0f;
- for (i=0; i<N; i++){
- sum += real[i];
- real[i] = (1.0f - sum);
- }
-
- /* store decimated bls tables */
- _store(bls, real, 1.0f, N);
-
-
-}
-
-extern "C"
-{
- void blosc_tilde_setup(void)
- {
- //post("blosc~ v0.1");
-
- build_tables();
-
- blosc_class = class_new(gensym("blosc~"), (t_newmethod)blosc_new,
- (t_method)blosc_free, sizeof(t_blosc), 0, A_DEFSYMBOL, A_NULL);
- CLASS_MAINSIGNALIN(blosc_class, t_blosc, x_f);
- class_addmethod(blosc_class, (t_method)blosc_dsp, gensym("dsp"), A_NULL);
- class_addmethod(blosc_class, (t_method)blosc_phase, gensym("phase"), A_FLOAT, A_NULL);
- class_addmethod(blosc_class, (t_method)blosc_phase2, gensym("phase2"), A_FLOAT, A_NULL);
-
-
- }
-
-}
Copied: branches/pd-extended/v0-40/externals/creb/modules++/blosc~.cc (from rev 9861, branches/pd-extended/v0-40/externals/creb/modules++/blosc.cc)
===================================================================
--- branches/pd-extended/v0-40/externals/creb/modules++/blosc~.cc (rev 0)
+++ branches/pd-extended/v0-40/externals/creb/modules++/blosc~.cc 2008-05-23 22:44:44 UTC (rev 9877)
@@ -0,0 +1,782 @@
+/*
+ * blosc.c - bandlimited oscillators
+ * 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 "DSPIcomplex.h"
+#include "DSPIfilters.h"
+
+/* work around old bug in Apple compilers <hans at at.or.at> */
+//#if MAC_OS_X_VERSION < MAC_OS_X_VERSION_10_4 // arg, this didn't work
+#ifndef isnan
+extern "C" int isnan(double);
+#endif
+#ifndef isinf
+extern "C" int isinf(double);
+#endif
+
+typedef unsigned long long u64;
+typedef unsigned long u32;
+
+
+
+#define LPHASOR (8*sizeof(u32)) // the phasor logsize
+#define VOICES 8 // the number of waveform voices
+#define LLENGTH 6 // the loglength of a fractional delayed basic waveform
+#define LOVERSAMPLE 6 // the log of the oversampling factor (nb of fract delayed waveforms)
+#define LPAD 1 // the log of the time padding factor (to reduce time aliasing)
+#define LTABLE (LLENGTH+LOVERSAMPLE)
+#define N (1<<LTABLE)
+#define M (1<<(LTABLE+LPAD))
+#define S (1<<LOVERSAMPLE)
+#define L (1<<LLENGTH)
+#define LMASK (L-1)
+
+#define WALPHA 0.1f // windowing alpha (0 = cos -> 1 = rect)
+#define CUTOFF 0.8f // fraction of nyquist for impulse cutoff
+#define NBPERIODS ((float)(L) * CUTOFF / 2.0f)
+
+/* sample buffers */
+static float bli[N]; // band limited impulse
+static float bls[N]; // band limited step
+static float blr[N]; // band limited ramp
+
+
+typedef struct bloscctl
+{
+ t_int c_index[VOICES]; // array of indices in sample table
+ t_float c_frac[VOICES]; // array of fractional indices
+ t_float c_vscale[VOICES]; // array of scale factors
+ t_int c_next_voice; // next voice to steal (round robin)
+ u32 c_phase; // phase of main oscillator
+ u32 c_phase2; // phase of secondairy oscillator
+ t_float c_state; // state of the square wave
+ t_float c_prev_amp; // previous input of comparator
+ t_float c_phase_inc_scale;
+ t_float c_scale;
+ t_float c_scale_update;
+ DSPIfilterSeries* c_butter; // the series filter
+ t_symbol *c_waveform;
+
+} t_bloscctl;
+
+typedef struct blosc
+{
+ t_object x_obj;
+ t_float x_f;
+ t_bloscctl x_ctl;
+} t_blosc;
+
+
+/* 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)) & ~(S-1);}
+
+
+/* flat table: better for linear interpolation */
+static inline float _play_voice_lint(float *table, t_int *index, float frac, float scale)
+{
+ t_int i = *index;
+
+ /* perform linear interpolation */
+ float f = (((1.0f - frac) * table[i]) + (table[i+1] * frac)) * scale;
+
+ /* increment phase index if next 2 elements will still be inside table
+ if not there's no increment and the voice will keep playing the same sample */
+
+ i += (((i+S+1) >> LTABLE) ^ 1) << LOVERSAMPLE;
+
+ *index = i;
+ return f;
+}
+
+/* get one sample from the bandlimited discontinuity wavetable playback syth */
+static inline t_float _get_bandlimited_discontinuity(t_bloscctl *ctl, float *table)
+{
+ float sum = 0.0f;
+ int i;
+ /* sum all voices */
+ for (i=0; i<VOICES; i++){
+ sum += _play_voice_lint(table, ctl->c_index+i, ctl->c_frac[i], ctl->c_vscale[i]);
+ }
+
+ return sum;
+}
+
+
+/* update waveplayers on zero cross */
+static void _bang_comparator(t_bloscctl *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_bloscctl *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_bloscctl *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 *blosc_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_bloscctl *ctl = (t_bloscctl *)(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 *blosc_perform_saw(t_int *w)
+{
+ t_float *freq = (float *)(w[3]);
+ t_float *out = (float *)(w[4]);
+ t_bloscctl *ctl = (t_bloscctl *)(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 *blosc_perform_pulse(t_int *w)
+{
+ t_float *freq = (float *)(w[3]);
+ t_float *out = (float *)(w[4]);
+ t_bloscctl *ctl = (t_bloscctl *)(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 *blosc_perform_comparator(t_int *w)
+{
+ t_float *amp = (float *)(w[3]);
+ t_float *out = (float *)(w[4]);
+ t_bloscctl *ctl = (t_bloscctl *)(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 blosc_phase(t_blosc *x, t_float f)
+{
+ x->x_ctl.c_phase = _float_to_phase(f);
+ x->x_ctl.c_phase2 = _float_to_phase(f);
+}
+
+static void blosc_phase1(t_blosc *x, t_float f)
+{
+ x->x_ctl.c_phase = _float_to_phase(f);
+}
+
+static void blosc_phase2(t_blosc *x, t_float f)
+{
+ x->x_ctl.c_phase2 = _float_to_phase(f);
+}
+
+static void blosc_dsp(t_blosc *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(blosc_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(blosc_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(blosc_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(blosc_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(blosc_perform_saw, 4, &x->x_ctl, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
+ }
+
+
+
+}
+static void blosc_free(t_blosc *x)
+{
+ delete x->x_ctl.c_butter;
+}
+
+t_class *blosc_class;
+
+static void *blosc_new(t_symbol *s)
+{
+ t_blosc *x = (t_blosc *)pd_new(blosc_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 */
+ blosc_phase(x, 0);
+ blosc_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;
+}
+
+
+
+
+
+
+
+/* CLASS DATA INIT (tables) */
+
+
+/* some vector ops */
+
+/* clear a buffer */
+static inline void _clear(float *array, int size)
+{
+ memset(array, 0, sizeof(float)*size);
+}
+
+/* compute complex log */
+static inline void _clog(float *real, float *imag, int size)
+{
+ int k;
+ for (k=0; k<size; k++){
+ float r = real[k];
+ float i = imag[k];
+ float radius = sqrt(r*r+i*i);
+ real[k] = log(radius);
+ imag[k] = atan2(i,r);
+ }
+}
+
+/* compute complex exp */
+static inline void _cexp(float *real, float *imag, int size)
+{
+ int k;
+ for (k=0; k<size; k++){
+ float r = exp(real[k]);
+ float i = imag[k];
+ real[k] = r * cos(i);
+ imag[k] = r * sin(i);
+ }
+}
+
+
+/* compute fft */
+static inline void _fft(float *real, float *imag, int size)
+{
+ int i;
+ float scale = 1.0f / sqrt((float)size);
+ for (i=0; i<size; i++){
+ real[i] *= scale;
+ imag[i] *= scale;
+ if (isnan(real[i])) post("fftpanic %d", i);
+ }
+ mayer_fft(size, real, imag);
+}
+/* compute ifft */
+static inline void _ifft(float *real, float *imag, int size)
+{
+ int i;
+ float scale = 1.0f / sqrt((float)size);
+ for (i=0; i<size; i++){
+ real[i] *= scale;
+ imag[i] *= scale;
+ if (isnan(real[i])) post("ifftpanic %d", i);
+ }
+ mayer_ifft(size, real, imag);
+}
+
+/* convert an integer index to a phase: [0 -> pi, -pi -> 0] */
+static inline float _i2theta(int i, int size){
+ float p = 2.0f * M_PI * (float)i / (float)size;
+ if (p >= M_PI) p -= 2.0f * M_PI;
+ return p;
+}
+
+
+/* print matlab array */
+static void _printm(float *array, char *name, int size)
+{
+ int i;
+ fprintf(stderr, "%s = [", name);
+ for (i=0; i<size; i++){
+ fprintf(stderr, "%f;", array[i]);
+ }
+ fprintf(stderr, "];\n");
+}
+
+/* store oversampled waveform as decimated chunks */
+static void _store_decimated(float *dst, float *src, float scale, int size)
+{
+ int i;
+ for (i=0; i<size; i++){
+ int offset = (i % S) * L;
+ int index = i / S;
+ dst[offset+index] = scale * src[i];
+ }
+
+}
+
+/* store waveform as one chunk */
+static void _store(float *dst, float *src, float scale, int size)
+{
+ int i;
+ for (i=0; i<size; i++){
+ dst[i] = scale * src[i];
+ }
+
+}
+
+/* create a minimum phase bandlimited impulse */
+static void build_tables(void)
+{
+
+ /* table size = M>=N (time padding to reduce time aliasing) */
+
+ /* we work in the complex domain to eliminate the need to avoid
+ negative spectral components */
+
+ float real[M];
+ float imag[M];
+ float sum,scale;
+ int i,j;
+
+
+ /* create windowed sinc */
+ _clear(imag, M);
+ real[0] = 1.0f;
+ for (i=1; i<M; i++){
+ float tw = _i2theta(i,M);
+ float ts = tw * NBPERIODS * (float)(M) / (float)(N);
+
+ /* sinc */
+ real[i] = sin(ts)/ts;
+
+ /* blackman window */
+ real[i] *= 0.42f + 0.5f * (cos(tw)) + 0.08f * (cos(2.0f*tw));
+
+ //real[i] *= 0.5f * (1.0f + WALPHA) + 0.5f * (1.0f - WALPHA) * (cos(tw));
+
+ /* check for nan */
+ if (isnan(real[i])) post("sinc NaN panic %d", i);
+ if (isinf(real[i])) post("sinc Inf panic %d", i);
+
+ }
+
+
+ /* compute cepstrum */
+ _fft(real, imag, M);
+ _clog(real, imag, M);
+ _ifft(real, imag, M);
+
+
+ /* kill anti-causal part (contribution of non minimum phase zeros) */
+ /* should we kill nyquist too ?? */
+ for (i=M/2+1; i<M; i++){
+ real[i] *= 0.0000f;
+ imag[i] *= 0.0000f;
+ }
+
+
+ /* compute inverse cepstrum */
+ _fft(real, imag, M);
+ _cexp(real, imag, M);
+ _ifft(real, imag, M);
+
+
+
+ /* from here on, discard the padded part [N->M-1]
+ and work with the first N samples */
+
+ /* normalize impulse (integral = 1) */
+ sum = 0.0f;
+ for (i=0; i<N; i++){sum += real[i];}
+ scale = 1.0f / sum;
+ for (i=0; i<N; i++){real[i] *= scale;}
+
+
+ /* store bli table */
+ _store(bli, real, (float)S, N);
+ //_printm(bli, "h", N);
+
+
+ /* integrate impulse and invert to produce a step function
+ from 1->0 */
+ sum = 0.0f;
+ for (i=0; i<N; i++){
+ sum += real[i];
+ real[i] = (1.0f - sum);
+ }
+
+ /* store decimated bls tables */
+ _store(bls, real, 1.0f, N);
+
+
+}
+
+extern "C"
+{
+ void blosc_tilde_setup(void)
+ {
+ //post("blosc~ v0.1");
+
+ build_tables();
+
+ blosc_class = class_new(gensym("blosc~"), (t_newmethod)blosc_new,
+ (t_method)blosc_free, sizeof(t_blosc), 0, A_DEFSYMBOL, A_NULL);
+ CLASS_MAINSIGNALIN(blosc_class, t_blosc, x_f);
+ class_addmethod(blosc_class, (t_method)blosc_dsp, gensym("dsp"), A_NULL);
+ class_addmethod(blosc_class, (t_method)blosc_phase, gensym("phase"), A_FLOAT, A_NULL);
+ class_addmethod(blosc_class, (t_method)blosc_phase2, gensym("phase2"), A_FLOAT, A_NULL);
+
+
+ }
+
+}
Deleted: branches/pd-extended/v0-40/externals/creb/modules++/filterortho.cc
===================================================================
--- branches/pd-extended/v0-40/externals/creb/modules++/filterortho.cc 2008-05-23 22:40:19 UTC (rev 9876)
+++ branches/pd-extended/v0-40/externals/creb/modules++/filterortho.cc 2008-05-23 22:44:44 UTC (rev 9877)
@@ -1,133 +0,0 @@
-/*
- * filterortho.cc - orthogonal biquad filter pd interface
- * 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 "DSPIcomplex.h"
-#include "DSPIfilters.h"
-
-
-
-typedef struct filterortho_struct
-{
- t_object x_obj;
- t_float x_f;
- DSPIfilterOrtho *filterortho;
-} t_filterortho;
-
-void filterortho_bang(t_filterortho *x)
-{
-
-}
-
-
-static t_int *filterortho_perform(t_int *w)
-{
-
-
- t_float *in = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
- DSPIfilterOrtho* filterortho = (DSPIfilterOrtho *)(w[1]);
- t_int n = (t_int)(w[2]);
- t_int i;
- t_float x;
-
-
- // dit kan beter
- float smooth = 1.0f - pow(.05f,1.0f/(float)(n));
-
- for (i = 0; i < n; i++)
- {
- x = *in++;
- filterortho->BangSmooth(x, x, smooth);
- *out++ = x;
- }
-
- filterortho->killDenormals();
-
- return (w+5);
-}
-
-static void filterortho_dsp(t_filterortho *x, t_signal **sp)
-{
- dsp_add(filterortho_perform, 4, x->filterortho, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
-
-}
-void filterortho_free(t_filterortho *x)
-{
- delete x->filterortho;
-
-}
-
-t_class *filterortho_class;
-
-
-
-void setLP(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setLP(f / sys_getsr(), Q);}
-void setHP(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setHP(f / sys_getsr(), Q);}
-void setBP(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setBP(f / sys_getsr(), Q);}
-void setBR(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setBR(f / sys_getsr(), Q);}
-void setAP(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setAP(f / sys_getsr(), Q);}
-
-void setLS(t_filterortho *x, t_floatarg f, t_floatarg A) {x->filterortho->setLS(f / sys_getsr(), A);}
-void setHS(t_filterortho *x, t_floatarg f, t_floatarg A) {x->filterortho->setHS(f / sys_getsr(), A);}
-
-void setEQ(t_filterortho *x, t_floatarg f, t_floatarg Q, t_floatarg A) {x->filterortho->setEQ(f / sys_getsr(), Q, A);}
-
-
-void *filterortho_new()
-{
- t_filterortho *x = (t_filterortho *)pd_new(filterortho_class);
- x->filterortho = new DSPIfilterOrtho();
- outlet_new(&x->x_obj, gensym("signal"));
- setLP(x, 10000, 2);
- return (void *)x;
-}
-
-
-
-extern "C" {
-
-void filterortho_tilde_setup(void)
-{
- //post("filterortho~ v0.1");
- filterortho_class = class_new(gensym("filterortho~"), (t_newmethod)filterortho_new,
- (t_method)filterortho_free, sizeof(t_filterortho), 0, A_NULL);
-
- CLASS_MAINSIGNALIN(filterortho_class, t_filterortho, x_f);
-
-
- class_addmethod(filterortho_class, (t_method)filterortho_bang, gensym("bang"), A_NULL);
-
- class_addmethod(filterortho_class, (t_method)filterortho_dsp, gensym("dsp"), A_NULL);
-
- class_addmethod(filterortho_class, (t_method)setLP, gensym("setLP"), A_FLOAT, A_FLOAT, A_NULL);
- class_addmethod(filterortho_class, (t_method)setHP, gensym("setHP"), A_FLOAT, A_FLOAT, A_NULL);
- class_addmethod(filterortho_class, (t_method)setBP, gensym("setBP"), A_FLOAT, A_FLOAT, A_NULL);
- class_addmethod(filterortho_class, (t_method)setBR, gensym("setBR"), A_FLOAT, A_FLOAT, A_NULL);
- class_addmethod(filterortho_class, (t_method)setAP, gensym("setAP"), A_FLOAT, A_FLOAT, A_NULL);
- class_addmethod(filterortho_class, (t_method)setLS, gensym("setLS"), A_FLOAT, A_FLOAT, A_NULL);
- class_addmethod(filterortho_class, (t_method)setHS, gensym("setHS"), A_FLOAT, A_FLOAT, A_NULL);
- class_addmethod(filterortho_class, (t_method)setEQ, gensym("setEQ"), A_FLOAT, A_FLOAT, A_FLOAT, A_NULL);
-
-}
-
-}
Copied: branches/pd-extended/v0-40/externals/creb/modules++/filterortho~.cc (from rev 9861, branches/pd-extended/v0-40/externals/creb/modules++/filterortho.cc)
===================================================================
--- branches/pd-extended/v0-40/externals/creb/modules++/filterortho~.cc (rev 0)
+++ branches/pd-extended/v0-40/externals/creb/modules++/filterortho~.cc 2008-05-23 22:44:44 UTC (rev 9877)
@@ -0,0 +1,133 @@
+/*
+ * filterortho.cc - orthogonal biquad filter pd interface
+ * 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 "DSPIcomplex.h"
+#include "DSPIfilters.h"
+
+
+
+typedef struct filterortho_struct
+{
+ t_object x_obj;
+ t_float x_f;
+ DSPIfilterOrtho *filterortho;
+} t_filterortho;
+
+void filterortho_bang(t_filterortho *x)
+{
+
+}
+
+
+static t_int *filterortho_perform(t_int *w)
+{
+
+
+ t_float *in = (float *)(w[3]);
+ t_float *out = (float *)(w[4]);
+ DSPIfilterOrtho* filterortho = (DSPIfilterOrtho *)(w[1]);
+ t_int n = (t_int)(w[2]);
+ t_int i;
+ t_float x;
+
+
+ // dit kan beter
+ float smooth = 1.0f - pow(.05f,1.0f/(float)(n));
+
+ for (i = 0; i < n; i++)
+ {
+ x = *in++;
+ filterortho->BangSmooth(x, x, smooth);
+ *out++ = x;
+ }
+
+ filterortho->killDenormals();
+
+ return (w+5);
+}
+
+static void filterortho_dsp(t_filterortho *x, t_signal **sp)
+{
+ dsp_add(filterortho_perform, 4, x->filterortho, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
+
+}
+void filterortho_free(t_filterortho *x)
+{
+ delete x->filterortho;
+
+}
+
+t_class *filterortho_class;
+
+
+
+void setLP(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setLP(f / sys_getsr(), Q);}
+void setHP(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setHP(f / sys_getsr(), Q);}
+void setBP(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setBP(f / sys_getsr(), Q);}
+void setBR(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setBR(f / sys_getsr(), Q);}
+void setAP(t_filterortho *x, t_floatarg f, t_floatarg Q) {x->filterortho->setAP(f / sys_getsr(), Q);}
+
+void setLS(t_filterortho *x, t_floatarg f, t_floatarg A) {x->filterortho->setLS(f / sys_getsr(), A);}
+void setHS(t_filterortho *x, t_floatarg f, t_floatarg A) {x->filterortho->setHS(f / sys_getsr(), A);}
+
+void setEQ(t_filterortho *x, t_floatarg f, t_floatarg Q, t_floatarg A) {x->filterortho->setEQ(f / sys_getsr(), Q, A);}
+
+
+void *filterortho_new()
+{
+ t_filterortho *x = (t_filterortho *)pd_new(filterortho_class);
+ x->filterortho = new DSPIfilterOrtho();
+ outlet_new(&x->x_obj, gensym("signal"));
+ setLP(x, 10000, 2);
+ return (void *)x;
+}
+
+
+
+extern "C" {
+
+void filterortho_tilde_setup(void)
+{
+ //post("filterortho~ v0.1");
+ filterortho_class = class_new(gensym("filterortho~"), (t_newmethod)filterortho_new,
+ (t_method)filterortho_free, sizeof(t_filterortho), 0, A_NULL);
+
+ CLASS_MAINSIGNALIN(filterortho_class, t_filterortho, x_f);
+
+
+ class_addmethod(filterortho_class, (t_method)filterortho_bang, gensym("bang"), A_NULL);
+
+ class_addmethod(filterortho_class, (t_method)filterortho_dsp, gensym("dsp"), A_NULL);
+
+ class_addmethod(filterortho_class, (t_method)setLP, gensym("setLP"), A_FLOAT, A_FLOAT, A_NULL);
+ class_addmethod(filterortho_class, (t_method)setHP, gensym("setHP"), A_FLOAT, A_FLOAT, A_NULL);
+ class_addmethod(filterortho_class, (t_method)setBP, gensym("setBP"), A_FLOAT, A_FLOAT, A_NULL);
+ class_addmethod(filterortho_class, (t_method)setBR, gensym("setBR"), A_FLOAT, A_FLOAT, A_NULL);
+ class_addmethod(filterortho_class, (t_method)setAP, gensym("setAP"), A_FLOAT, A_FLOAT, A_NULL);
+ class_addmethod(filterortho_class, (t_method)setLS, gensym("setLS"), A_FLOAT, A_FLOAT, A_NULL);
+ class_addmethod(filterortho_class, (t_method)setHS, gensym("setHS"), A_FLOAT, A_FLOAT, A_NULL);
+ class_addmethod(filterortho_class, (t_method)setEQ, gensym("setEQ"), A_FLOAT, A_FLOAT, A_FLOAT, A_NULL);
+
+}
+
+}
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