merged in relevant changes from the v0-40 pd-extended release branch

svn path=/trunk/externals/creb/; revision=10266

1 files changed, 0 insertions, 782 deletions

diff --git a/modules++/blosc.cc b/modules++/blosc.cc
deleted file mode 100644
index f2b13d8..0000000
--- a/modules++/blosc.cc
+++ /dev/null
@@ -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.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); 
-
-	
-    }
-
-}