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-rw-r--r--modules++/blosc~.cc210
1 files changed, 105 insertions, 105 deletions
diff --git a/modules++/blosc~.cc b/modules++/blosc~.cc
index 8a4dd4b..9028eaf 100644
--- a/modules++/blosc~.cc
+++ b/modules++/blosc~.cc
@@ -46,14 +46,14 @@ typedef unsigned long u32;
#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)
+#define WALPHA 0.1 // windowing alpha (0 = cos -> 1 = rect)
+#define CUTOFF 0.8 // fraction of nyquist for impulse cutoff
+#define NBPERIODS ((t_float)(L) * CUTOFF / 2.0)
/* sample buffers */
-static float bli[N]; // band limited impulse
-static float bls[N]; // band limited step
-static float blr[N]; // band limited ramp
+static t_float bli[N]; // band limited impulse
+static t_float bls[N]; // band limited step
+static t_float blr[N]; // band limited ramp
typedef struct bloscctl
@@ -83,17 +83,17 @@ typedef struct 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);}
+static inline t_float _phase_to_float(u32 p){return ((t_float)p) * (1.0 / 4294967296.0);}
+static inline u32 _float_to_phase(t_float f){return ((u32)(f * 4294967296.0)) & ~(S-1);}
/* flat table: better for linear interpolation */
-static inline float _play_voice_lint(float *table, t_int *index, float frac, float scale)
+static inline t_float _play_voice_lint(t_float *table, t_int *index, t_float frac, t_float scale)
{
int i = *index;
/* perform linear interpolation */
- float f = (((1.0f - frac) * table[i]) + (table[i+1] * frac)) * scale;
+ t_float f = (((1.0 - 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 */
@@ -105,9 +105,9 @@ static inline float _play_voice_lint(float *table, t_int *index, float frac, flo
}
/* get one sample from the bandlimited discontinuity wavetable playback syth */
-static inline t_float _get_bandlimited_discontinuity(t_bloscctl *ctl, float *table)
+static inline t_float _get_bandlimited_discontinuity(t_bloscctl *ctl, t_float *table)
{
- float sum = 0.0f;
+ t_float sum = 0.0;
int i;
/* sum all voices */
for (i=0; i<VOICES; i++){
@@ -119,18 +119,18 @@ static inline t_float _get_bandlimited_discontinuity(t_bloscctl *ctl, float *tab
/* update waveplayers on zero cross */
-static void _bang_comparator(t_bloscctl *ctl, float prev, float curr)
+static void _bang_comparator(t_bloscctl *ctl, t_float prev, t_float curr)
{
/* check for sign change */
- if ((prev * curr) < 0.0f){
+ if ((prev * curr) < 0.0){
int voice;
/* determine the location of the discontinuity (in oversampled coordiates
using linear interpolation */
- float f = (float)S * curr / (curr - prev);
+ t_float f = (t_float)S * curr / (curr - prev);
/* get the offset in the oversample table */
@@ -139,11 +139,11 @@ static void _bang_comparator(t_bloscctl *ctl, float prev, float curr)
/* determine the fractional part (in oversampled coordinates)
for linear interpolation */
- float table_frac_index = f - (float)table_index;
+ t_float table_frac_index = f - (t_float)table_index;
/* set state (+ or -) */
- ctl->c_state = (curr > 0.0f) ? 0.5f : -0.5f;
+ ctl->c_state = (curr > 0.0) ? 0.5 : -0.5;
/* steal the oldest voice */
@@ -154,7 +154,7 @@ static void _bang_comparator(t_bloscctl *ctl, float prev, float curr)
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;
+ ctl->c_vscale[voice] = -ctl->c_scale * 2.0 * ctl->c_state;
}
@@ -162,20 +162,20 @@ static void _bang_comparator(t_bloscctl *ctl, float prev, float curr)
/* advance phasor and update waveplayers on phase wrap */
-static void _bang_phasor(t_bloscctl *ctl, float freq)
+static void _bang_phasor(t_bloscctl *ctl, t_float freq)
{
u32 phase = ctl->c_phase;
u32 phase_inc;
u32 oldphase;
int voice;
- float scale = ctl->c_scale;
+ t_float scale = ctl->c_scale;
/* get increment */
- float inc = freq * ctl->c_phase_inc_scale;
+ t_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;
+ if (inc < 0.0) inc = -inc;
phase_inc = ((u32)inc) & ~(S-1);
oldphase = phase;
phase += phase_inc;
@@ -205,7 +205,7 @@ static void _bang_phasor(t_bloscctl *ctl, float freq)
/* 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_frac[voice] = (t_float)table_phase / (t_float)phase_inc_decimated;
ctl->c_vscale[voice] = scale;
scale = scale * ctl->c_scale_update;
@@ -221,7 +221,7 @@ static void _bang_phasor(t_bloscctl *ctl, float freq)
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)
+static void _bang_hardsync_phasor(t_bloscctl *ctl, t_float freq, t_float freq2)
{
u32 phase = ctl->c_phase;
u32 phase2 = ctl->c_phase2;
@@ -230,12 +230,12 @@ static void _bang_hardsync_phasor(t_bloscctl *ctl, float freq, float freq2)
u32 oldphase;
u32 oldphase2;
int voice;
- float scale = ctl->c_scale;
+ t_float scale = ctl->c_scale;
/* get increment */
- float inc = freq * ctl->c_phase_inc_scale;
- float inc2 = freq2 * ctl->c_phase_inc_scale;
+ t_float inc = freq * ctl->c_phase_inc_scale;
+ t_float inc2 = freq2 * ctl->c_phase_inc_scale;
/* calculate new phases
the increment (and the phase) should be a multiple of S */
@@ -245,12 +245,12 @@ static void _bang_hardsync_phasor(t_bloscctl *ctl, float freq, float freq2)
oldphase2 = phase2;
/* update second osc */
- if (inc2 < 0.0f) inc2 = -inc2;
+ if (inc2 < 0.0) 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;
+ if (inc < 0.0) inc = -inc;
phase_inc = ((u32)inc) & ~(S-1);
if (phase_inc < phase_inc2) phase_inc = phase_inc2;
phase += phase_inc;
@@ -274,7 +274,7 @@ static void _bang_hardsync_phasor(t_bloscctl *ctl, float freq, float freq2)
u32 phase_inc_decimated = phase_inc >> LOVERSAMPLE;
u32 table_index;
u32 table_phase;
- float stepsize;
+ t_float stepsize;
/* steal the oldest voice if we have a phase wrap */
@@ -298,12 +298,12 @@ static void _bang_hardsync_phasor(t_bloscctl *ctl, float freq, float freq2)
reduce the bit depth to prevent overflow */
stepsize = _phase_to_float(((oldphase-phase) >> LOVERSAMPLE)
- + phase_inc_decimated) * (float)S;
+ + phase_inc_decimated) * (t_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_frac[voice] = (t_float)table_phase / (t_float)phase_inc_decimated;
ctl->c_vscale[voice] = scale * stepsize;
scale = scale * ctl->c_scale_update;
@@ -318,28 +318,28 @@ static void _bang_hardsync_phasor(t_bloscctl *ctl, float freq, float freq2)
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_float *freq = (t_float *)(w[3]);
+ t_float *freq2 = (t_float *)(w[4]);
+ t_float *out = (t_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()));
+ ctl->c_butter->setButterHP(0.85 * (*freq / sys_getsr()));
while (n--) {
- float frequency = *freq++;
- float frequency2 = *freq2++;
+ t_float frequency = *freq++;
+ t_float frequency2 = *freq2++;
/* get the bandlimited discontinuity */
- float sample = _get_bandlimited_discontinuity(ctl, bls);
+ t_float sample = _get_bandlimited_discontinuity(ctl, bls);
/* add aliased sawtooth wave */
- sample += _phase_to_float(ctl->c_phase) - 0.5f;
+ sample += _phase_to_float(ctl->c_phase) - 0.5;
/* highpass filter output to remove DC offset and low frequency aliasing */
- ctl->c_butter->BangSmooth(sample, sample, 0.05f);
+ ctl->c_butter->BangSmooth(sample, sample, 0.05);
/* send to output */
*out++ = sample;
@@ -354,20 +354,20 @@ static t_int *blosc_perform_hardsync_saw(t_int *w)
static t_int *blosc_perform_saw(t_int *w)
{
- t_float *freq = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
+ t_float *freq = (t_float *)(w[3]);
+ t_float *out = (t_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++;
+ t_float frequency = *freq++;
/* get the bandlimited discontinuity */
- float sample = _get_bandlimited_discontinuity(ctl, bls);
+ t_float sample = _get_bandlimited_discontinuity(ctl, bls);
/* add aliased sawtooth wave */
- sample += _phase_to_float(ctl->c_phase) - 0.5f;
+ sample += _phase_to_float(ctl->c_phase) - 0.5;
/* send to output */
*out++ = sample;
@@ -384,24 +384,24 @@ static t_int *blosc_perform_saw(t_int *w)
static t_int *blosc_perform_pulse(t_int *w)
{
- t_float *freq = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
+ t_float *freq = (t_float *)(w[3]);
+ t_float *out = (t_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()));
+ ctl->c_butter->setButterHP(0.85 * (*freq / sys_getsr()));
while (n--) {
- float frequency = *freq++;
+ t_float frequency = *freq++;
/* get the bandlimited discontinuity */
- float sample = _get_bandlimited_discontinuity(ctl, bli);
+ t_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);
+ ctl->c_butter->BangSmooth(sample, sample, 0.05);
/* send to output */
*out++ = sample;
@@ -416,21 +416,21 @@ static t_int *blosc_perform_pulse(t_int *w)
static t_int *blosc_perform_comparator(t_int *w)
{
- t_float *amp = (float *)(w[3]);
- t_float *out = (float *)(w[4]);
+ t_float *amp = (t_float *)(w[3]);
+ t_float *out = (t_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++;
+ t_float curr_amp = *amp++;
/* exact zero won't work for zero detection (sic) */
- if (curr_amp == 0.0f) curr_amp = 0.0000001f;
+ if (curr_amp == 0.0) curr_amp = 0.0000001;
/* get the bandlimited discontinuity */
- float sample = _get_bandlimited_discontinuity(ctl, bls);
+ t_float sample = _get_bandlimited_discontinuity(ctl, bls);
/* add the block wave state */
sample += ctl->c_state;
@@ -471,38 +471,38 @@ 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();
+ x->x_ctl.c_phase_inc_scale = 4.0 * (t_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;
+ x->x_ctl.c_scale = 1.0;
+ x->x_ctl.c_scale_update = 1.0;
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;
+ x->x_ctl.c_scale = 1.0;
+ x->x_ctl.c_scale_update = 1.0;
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;
+ x->x_ctl.c_scale = 1.0;
+ x->x_ctl.c_scale_update = -1.0;
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;
+ x->x_ctl.c_scale = 1.0;
+ x->x_ctl.c_scale_update = 1.0;
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;
+ x->x_ctl.c_scale = 1.0;
+ x->x_ctl.c_scale_update = 1.0;
dsp_add(blosc_perform_saw, 4, &x->x_ctl, sp[0]->s_n, sp[0]->s_vec, sp[1]->s_vec);
}
@@ -540,7 +540,7 @@ static void *blosc_new(t_symbol *s)
/* init oscillators */
for (i=0; i<VOICES; i++) {
x->x_ctl.c_index[i] = N-2;
- x->x_ctl.c_frac[i] = 0.0f;
+ x->x_ctl.c_frac[i] = 0.0;
}
/* init rest of state data */
@@ -549,8 +549,8 @@ static void *blosc_new(t_symbol *s)
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_scale = 1.0;
+ x->x_ctl.c_scale_update = 1.0;
x->x_ctl.c_waveform = s;
return (void *)x;
@@ -568,31 +568,31 @@ static void *blosc_new(t_symbol *s)
/* some vector ops */
/* clear a buffer */
-static inline void _clear(float *array, int size)
+static inline void _clear(t_float *array, int size)
{
- memset(array, 0, sizeof(float)*size);
+ memset(array, 0, sizeof(t_float)*size);
}
/* compute complex log */
-static inline void _clog(float *real, float *imag, int size)
+static inline void _clog(t_float *real, t_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);
+ t_float r = real[k];
+ t_float i = imag[k];
+ t_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)
+static inline void _cexp(t_float *real, t_float *imag, int size)
{
int k;
for (k=0; k<size; k++){
- float r = exp(real[k]);
- float i = imag[k];
+ t_float r = exp(real[k]);
+ t_float i = imag[k];
real[k] = r * cos(i);
imag[k] = r * sin(i);
}
@@ -600,10 +600,10 @@ static inline void _cexp(float *real, float *imag, int size)
/* compute fft */
-static inline void _fft(float *real, float *imag, int size)
+static inline void _fft(t_float *real, t_float *imag, int size)
{
int i;
- float scale = 1.0f / sqrt((float)size);
+ t_float scale = 1.0 / sqrt((t_float)size);
for (i=0; i<size; i++){
real[i] *= scale;
imag[i] *= scale;
@@ -612,10 +612,10 @@ static inline void _fft(float *real, float *imag, int size)
mayer_fft(size, real, imag);
}
/* compute ifft */
-static inline void _ifft(float *real, float *imag, int size)
+static inline void _ifft(t_float *real, t_float *imag, int size)
{
int i;
- float scale = 1.0f / sqrt((float)size);
+ t_float scale = 1.0 / sqrt((t_float)size);
for (i=0; i<size; i++){
real[i] *= scale;
imag[i] *= scale;
@@ -625,15 +625,15 @@ static inline void _ifft(float *real, float *imag, int size)
}
/* 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;
+static inline t_float _i2theta(int i, int size){
+ t_float p = 2.0 * M_PI * (t_float)i / (t_float)size;
+ if (p >= M_PI) p -= 2.0 * M_PI;
return p;
}
/* print matlab array */
-static void _printm(float *array, char *name, int size)
+static void _printm(t_float *array, char *name, int size)
{
int i;
fprintf(stderr, "%s = [", name);
@@ -644,7 +644,7 @@ static void _printm(float *array, char *name, int size)
}
/* store oversampled waveform as decimated chunks */
-static void _store_decimated(float *dst, float *src, float scale, int size)
+static void _store_decimated(t_float *dst, t_float *src, t_float scale, int size)
{
int i;
for (i=0; i<size; i++){
@@ -656,7 +656,7 @@ static void _store_decimated(float *dst, float *src, float scale, int size)
}
/* store waveform as one chunk */
-static void _store(float *dst, float *src, float scale, int size)
+static void _store(t_float *dst, t_float *src, t_float scale, int size)
{
int i;
for (i=0; i<size; i++){
@@ -674,24 +674,24 @@ static void build_tables(void)
/* we work in the complex domain to eliminate the need to avoid
negative spectral components */
- float real[M];
- float imag[M];
- float sum,scale;
+ t_float real[M];
+ t_float imag[M];
+ t_float sum,scale;
int i,j;
/* create windowed sinc */
_clear(imag, M);
- real[0] = 1.0f;
+ real[0] = 1.0;
for (i=1; i<M; i++){
- float tw = _i2theta(i,M);
- float ts = tw * NBPERIODS * (float)(M) / (float)(N);
+ t_float tw = _i2theta(i,M);
+ t_float ts = tw * NBPERIODS * (t_float)(M) / (t_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] *= 0.42 + 0.5 * (cos(tw)) + 0.08 * (cos(2.0*tw));
//real[i] *= 0.5f * (1.0f + WALPHA) + 0.5f * (1.0f - WALPHA) * (cos(tw));
@@ -711,8 +711,8 @@ static void build_tables(void)
/* 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;
+ real[i] *= 0.0000;
+ imag[i] *= 0.0000;
}
@@ -727,27 +727,27 @@ static void build_tables(void)
and work with the first N samples */
/* normalize impulse (integral = 1) */
- sum = 0.0f;
+ sum = 0.0;
for (i=0; i<N; i++){sum += real[i];}
- scale = 1.0f / sum;
+ scale = 1.0 / sum;
for (i=0; i<N; i++){real[i] *= scale;}
/* store bli table */
- _store(bli, real, (float)S, N);
+ _store(bli, real, (t_float)S, N);
//_printm(bli, "h", N);
/* integrate impulse and invert to produce a step function
from 1->0 */
- sum = 0.0f;
+ sum = 0.0;
for (i=0; i<N; i++){
sum += real[i];
- real[i] = (1.0f - sum);
+ real[i] = (1.0 - sum);
}
/* store decimated bls tables */
- _store(bls, real, 1.0f, N);
+ _store(bls, real, 1.0, N);
}
generated by cgit v1.2.1 (git 2.18.0) at 2025-02-22 20:24:46 +0000