commit 9f909f7ef7d1e6a6d32b5622fb138f5f06142a14
parent d2c9a44b870679f3c2660278424a309bb1917aca
Author: fundamental <mark.d.mccurry@gmail.com>
Date: Wed, 14 Sep 2011 18:11:57 -0400
Yoshimi: converging style in DSP
Diffstat:
9 files changed, 193 insertions(+), 234 deletions(-)
diff --git a/src/DSP/AnalogFilter.cpp b/src/DSP/AnalogFilter.cpp
@@ -22,9 +22,9 @@
*/
-#include <cmath>
-#include <cstdio>
#include <cstring> //memcpy
+#include <cmath>
+
#include "../Misc/Util.h"
#include "AnalogFilter.h"
@@ -32,24 +32,20 @@ AnalogFilter::AnalogFilter(unsigned char Ftype,
float Ffreq,
float Fq,
unsigned char Fstages)
+ :type(Ftype),
+ stages(Fstages),
+ freq(Ffreq),
+ q(Fq),
+ gain(1.0),
+ abovenq(false),
+ oldabovenq(false)
{
- stages = Fstages;
- for(int i = 0; i < 3; ++i) {
- coeff.c[i] = 0.0f;
- coeff.d[i] = 0.0f;
- oldCoeff.c[i] = 0.0f;
- oldCoeff.d[i] = 0.0f;
- }
- type = Ftype;
- freq = Ffreq;
- q = Fq;
- gain = 1.0f;
+ for(int i = 0; i < 3; ++i)
+ coeff.c[i] = coeff.d[i] = oldCoeff.c[i] = oldCoeff.d[i] = 0.0f;
if(stages >= MAX_FILTER_STAGES)
stages = MAX_FILTER_STAGES;
cleanup();
firsttime = false;
- abovenq = false;
- oldabovenq = false;
setfreq_and_q(Ffreq, Fq);
firsttime = true;
coeff.d[0] = 0; //this is not used
@@ -71,10 +67,10 @@ void AnalogFilter::cleanup()
needsinterpolation = false;
}
-void AnalogFilter::computefiltercoefs()
+void AnalogFilter::computefiltercoefs(void)
{
float tmp;
- bool zerocoefs = false; //this is used if the freq is too high
+ bool zerocoefs = false; //this is used if the freq is too high
//do not allow frequencies bigger than samplerate/2
float freq = this->freq;
@@ -93,7 +89,7 @@ void AnalogFilter::computefiltercoefs()
tmpgain = gain;
}
else {
- tmpq = (q > 1.0f ? powf(q, 1.0f / (stages + 1)) : q);
+ tmpq = (q > 1.0f) ? powf(q, 1.0f / (stages + 1)) : q;
tmpgain = powf(gain, 1.0f / (stages + 1));
}
@@ -137,105 +133,85 @@ void AnalogFilter::computefiltercoefs()
break;
case 2: //LPF 2 poles
if(!zerocoefs) {
- alpha = sn / (2 * tmpq);
+ alpha = sn / (2.0f * tmpq);
tmp = 1 + alpha;
- c[0] = (1.0f - cs) / 2.0f / tmp;
c[1] = (1.0f - cs) / tmp;
- c[2] = (1.0f - cs) / 2.0f / tmp;
- d[1] = -2 * cs / tmp * (-1);
- d[2] = (1 - alpha) / tmp * (-1);
+ c[0] = c[2] = c[1] / 2.0f;
+ d[1] = -2.0f * cs / tmp * -1.0f;
+ d[2] = (1.0f - alpha) / tmp * -1.0f;
}
else {
c[0] = 1.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
+ c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
case 3: //HPF 2 poles
if(!zerocoefs) {
- alpha = sn / (2 * tmpq);
+ alpha = sn / (2.0f * tmpq);
tmp = 1 + alpha;
c[0] = (1.0f + cs) / 2.0f / tmp;
c[1] = -(1.0f + cs) / tmp;
c[2] = (1.0f + cs) / 2.0f / tmp;
- d[1] = -2 * cs / tmp * (-1);
- d[2] = (1 - alpha) / tmp * (-1);
- }
- else {
- c[0] = 0.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
+ d[1] = -2.0f * cs / tmp * -1.0f;
+ d[2] = (1.0f - alpha) / tmp * -1.0f;
}
+ else
+ c[0] = c[1] = c[2] = d[1] = d[2] = 0.0f;
order = 2;
break;
case 4: //BPF 2 poles
if(!zerocoefs) {
- alpha = sn / (2 * tmpq);
- tmp = 1 + alpha;
- c[0] = alpha / tmp *sqrt(tmpq + 1);
- c[1] = 0;
- c[2] = -alpha / tmp *sqrt(tmpq + 1);
- d[1] = -2 * cs / tmp * (-1);
- d[2] = (1 - alpha) / tmp * (-1);
- }
- else {
- c[0] = 0.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
+ alpha = sn / (2.0f * tmpq);
+ tmp = 1.0f + alpha;
+ c[0] = alpha / tmp *sqrtf(tmpq + 1.0f);
+ c[1] = 0.0f;
+ c[2] = -alpha / tmp *sqrtf(tmpq + 1.0f);
+ d[1] = -2.0f * cs / tmp * -1.0f;
+ d[2] = (1.0f - alpha) / tmp * -1.0f;
}
+ else
+ c[0] = c[1] = c[2] = d[1] = d[2] = 0.0f;
order = 2;
break;
case 5: //NOTCH 2 poles
if(!zerocoefs) {
- alpha = sn / (2 * sqrt(tmpq));
- tmp = 1 + alpha;
- c[0] = 1 / tmp;
- c[1] = -2 * cs / tmp;
- c[2] = 1 / tmp;
- d[1] = -2 * cs / tmp * (-1);
- d[2] = (1 - alpha) / tmp * (-1);
+ alpha = sn / (2.0f * sqrtf(tmpq));
+ tmp = 1.0f + alpha;
+ c[0] = 1.0f / tmp;
+ c[1] = -2.0f * cs / tmp;
+ c[2] = 1.0f / tmp;
+ d[1] = -2.0f * cs / tmp * -1.0f;
+ d[2] = (1.0f - alpha) / tmp * -1.0f;
}
else {
c[0] = 1.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
+ c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
case 6: //PEAK (2 poles)
if(!zerocoefs) {
tmpq *= 3.0f;
- alpha = sn / (2 * tmpq);
- tmp = 1 + alpha / tmpgain;
+ alpha = sn / (2.0f * tmpq);
+ tmp = 1.0f + alpha / tmpgain;
c[0] = (1.0f + alpha * tmpgain) / tmp;
c[1] = (-2.0f * cs) / tmp;
c[2] = (1.0f - alpha * tmpgain) / tmp;
- d[1] = -2 * cs / tmp * (-1);
- d[2] = (1 - alpha / tmpgain) / tmp * (-1);
+ d[1] = -2.0f * cs / tmp * -1.0f;
+ d[2] = (1.0f - alpha / tmpgain) / tmp * -1.0f;
}
else {
c[0] = 1.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
+ c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
case 7: //Low Shelf - 2 poles
if(!zerocoefs) {
- tmpq = sqrt(tmpq);
- alpha = sn / (2 * tmpq);
- beta = sqrt(tmpgain) / tmpq;
+ tmpq = sqrtf(tmpq);
+ alpha = sn / (2.0f * tmpq);
+ beta = sqrtf(tmpgain) / tmpq;
tmp = (tmpgain + 1.0f) + (tmpgain - 1.0f) * cs + beta * sn;
c[0] = tmpgain
@@ -246,26 +222,22 @@ void AnalogFilter::computefiltercoefs()
c[2] = tmpgain
* ((tmpgain
+ 1.0f) - (tmpgain - 1.0f) * cs - beta * sn) / tmp;
- d[1] = -2.0f
- * ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs) / tmp * (-1);
- d[2] =
- ((tmpgain
- + 1.0f) + (tmpgain - 1.0f) * cs - beta * sn) / tmp * (-1);
+ d[1] = -2.0f * ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs)
+ / tmp * -1.0f;
+ d[2] = ((tmpgain + 1.0f) + (tmpgain - 1.0f) * cs - beta * sn)
+ / tmp * -1.0f;
}
else {
c[0] = tmpgain;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
+ c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
case 8: //High Shelf - 2 poles
if(!zerocoefs) {
- tmpq = sqrt(tmpq);
- alpha = sn / (2 * tmpq);
- beta = sqrt(tmpgain) / tmpq;
+ tmpq = sqrtf(tmpq);
+ alpha = sn / (2.0f * tmpq);
+ beta = sqrtf(tmpgain) / tmpq;
tmp = (tmpgain + 1.0f) - (tmpgain - 1.0f) * cs + beta * sn;
c[0] = tmpgain
@@ -276,18 +248,14 @@ void AnalogFilter::computefiltercoefs()
c[2] = tmpgain
* ((tmpgain
+ 1.0f) + (tmpgain - 1.0f) * cs - beta * sn) / tmp;
- d[1] = 2.0f
- * ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs) / tmp * (-1);
- d[2] =
- ((tmpgain
- + 1.0f) - (tmpgain - 1.0f) * cs - beta * sn) / tmp * (-1);
+ d[1] = 2.0f * ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs)
+ / tmp * -1.0f;
+ d[2] = ((tmpgain + 1.0f) - (tmpgain - 1.0f) * cs - beta * sn)
+ / tmp * -1.0f;
}
else {
c[0] = 1.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
+ c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
@@ -370,7 +338,7 @@ void AnalogFilter::singlefilterout(float *smp, fstage &hist,
hist.x1 = smp[i];
smp[i] = y0;
}
- if(order == 2) //Second order filter
+ if(order == 2) //Second order filter
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float y0 = smp[i] * coeff.c[0] + hist.x1 * coeff.c[1]
+ hist.x2 * coeff.c[2] + hist.y1 * coeff.d[1]
@@ -382,6 +350,7 @@ void AnalogFilter::singlefilterout(float *smp, fstage &hist,
smp[i] = y0;
}
}
+
void AnalogFilter::filterout(float *smp)
{
for(int i = 0; i < stages + 1; ++i)
diff --git a/src/DSP/AnalogFilter.h b/src/DSP/AnalogFilter.h
@@ -34,9 +34,7 @@
class AnalogFilter:public Filter
{
public:
- AnalogFilter(unsigned char Ftype,
- float Ffreq,
- float Fq,
+ AnalogFilter(unsigned char Ftype, float Ffreq, float Fq,
unsigned char Fstages);
~AnalogFilter();
void filterout(float *smp);
@@ -66,15 +64,15 @@ class AnalogFilter:public Filter
//Apply IIR filter to Samples, with coefficients, and past history
void singlefilterout(float *smp, fstage &hist, const Coeff &coeff);
//Update coeff and order
- void computefiltercoefs();
+ void computefiltercoefs(void);
- int type; //The type of the filter (LPF1,HPF1,LPF2,HPF2...)
- int stages; //how many times the filter is applied (0->1,1->2,etc.)
+ int type; //The type of the filter (LPF1,HPF1,LPF2,HPF2...)
+ int stages; //how many times the filter is applied (0->1,1->2,etc.)
float freq; //Frequency given in Hz
float q; //Q factor (resonance or Q factor)
float gain; //the gain of the filter (if are shelf/peak) filters
- int order; //the order of the filter (number of poles)
+ int order; //the order of the filter (number of poles)
bool needsinterpolation, //Interpolation between coeff changes
firsttime; //First Iteration of filter
diff --git a/src/DSP/FormantFilter.cpp b/src/DSP/FormantFilter.cpp
@@ -96,17 +96,16 @@ void FormantFilter::setpos(float input)
slowinput = slowinput
* (1.0f - formantslowness) + input * formantslowness;
- if((fabs(oldinput - input) < 0.001f) && (fabs(slowinput - input) < 0.001f)
- && (fabs(Qfactor - oldQfactor) < 0.001f)) {
-// oldinput=input; daca setez asta, o sa faca probleme la schimbari foarte lente
+ if((fabsf(oldinput - input) < 0.001f) && (fabsf(slowinput - input) < 0.001f)
+ && (fabsf(Qfactor - oldQfactor) < 0.001f)) {
+ // oldinput=input; daca setez asta, o sa faca probleme la schimbari foarte lente
firsttime = 0;
return;
}
else
oldinput = input;
-
- float pos = fmod(input * sequencestretch, 1.0f);
+ float pos = fmodf(input * sequencestretch, 1.0f);
if(pos < 0.0f)
pos += 1.0f;
@@ -115,7 +114,7 @@ void FormantFilter::setpos(float input)
if(p1 < 0)
p1 += sequencesize;
- pos = fmod(pos * sequencesize, 1.0f);
+ pos = fmodf(pos * sequencesize, 1.0f);
if(pos < 0.0f)
pos = 0.0f;
else
@@ -131,14 +130,14 @@ void FormantFilter::setpos(float input)
if(firsttime != 0) {
for(int i = 0; i < numformants; ++i) {
- currentformants[i].freq = formantpar[p1][i].freq
- * (1.0f
- - pos) + formantpar[p2][i].freq * pos;
- currentformants[i].amp = formantpar[p1][i].amp
- * (1.0f
- - pos) + formantpar[p2][i].amp * pos;
- currentformants[i].q = formantpar[p1][i].q
- * (1.0f - pos) + formantpar[p2][i].q * pos;
+ currentformants[i].freq =
+ formantpar[p1][i].freq
+ * (1.0f - pos) + formantpar[p2][i].freq * pos;
+ currentformants[i].amp =
+ formantpar[p1][i].amp
+ * (1.0f - pos) + formantpar[p2][i].amp * pos;
+ currentformants[i].q =
+ formantpar[p1][i].q * (1.0f - pos) + formantpar[p2][i].q * pos;
formant[i]->setfreq_and_q(currentformants[i].freq,
currentformants[i].q * Qfactor);
oldformantamp[i] = currentformants[i].amp;
@@ -147,26 +146,22 @@ void FormantFilter::setpos(float input)
}
else
for(int i = 0; i < numformants; ++i) {
- currentformants[i].freq = currentformants[i].freq
- * (1.0f - formantslowness)
- + (formantpar[p1][i].freq
- * (1.0f
- - pos) + formantpar[p2][i].freq
- * pos) * formantslowness;
-
- currentformants[i].amp = currentformants[i].amp
- * (1.0f - formantslowness)
- + (formantpar[p1][i].amp
- * (1.0f
- - pos) + formantpar[p2][i].amp
- * pos) * formantslowness;
+ currentformants[i].freq =
+ currentformants[i].freq * (1.0f - formantslowness)
+ + (formantpar[p1][i].freq
+ * (1.0f - pos) + formantpar[p2][i].freq * pos)
+ * formantslowness;
+
+ currentformants[i].amp =
+ currentformants[i].amp * (1.0f - formantslowness)
+ + (formantpar[p1][i].amp * (1.0f - pos)
+ + formantpar[p2][i].amp * pos) * formantslowness;
currentformants[i].q = currentformants[i].q
- * (1.0f - formantslowness)
- + (formantpar[p1][i].q
- * (1.0f
- - pos) + formantpar[p2][i].q
- * pos) * formantslowness;
+ * (1.0f - formantslowness)
+ + (formantpar[p1][i].q * (1.0f - pos)
+ + formantpar[p2][i].q * pos) * formantslowness;
+
formant[i]->setfreq_and_q(currentformants[i].freq,
currentformants[i].q * Qfactor);
diff --git a/src/DSP/FormantFilter.h b/src/DSP/FormantFilter.h
@@ -38,9 +38,13 @@ class FormantFilter:public Filter
void setq(float q_);
void setgain(float dBgain);
- void cleanup();
+ void cleanup(void);
+
private:
- class AnalogFilter * formant[FF_MAX_FORMANTS];
+ void setpos(float input);
+
+
+ class AnalogFilter *formant[FF_MAX_FORMANTS];
struct {
float freq, amp, q; //frequency,amplitude,Q
@@ -57,8 +61,6 @@ class FormantFilter:public Filter
float oldinput, slowinput;
float Qfactor, formantslowness, oldQfactor;
float vowelclearness, sequencestretch;
-
- void setpos(float input);
};
#endif
diff --git a/src/DSP/SVFilter.cpp b/src/DSP/SVFilter.cpp
@@ -28,22 +28,19 @@
#include "../Misc/Util.h"
#include "SVFilter.h"
-SVFilter::SVFilter(unsigned char Ftype,
- float Ffreq,
- float Fq,
+SVFilter::SVFilter(unsigned char Ftype, float Ffreq, float Fq,
unsigned char Fstages)
+ :type(Ftype),
+ stages(Fstages),
+ freq(Ffreq),
+ q(Fq),
+ gain(1.0f),
+ needsinterpolation(false),
+ firsttime(true)
{
- assert(Ftype < 4);
- stages = Fstages;
- type = Ftype;
- freq = Ffreq;
- q = Fq;
- gain = 1.0f;
- outgain = 1.0f;
- needsinterpolation = false;
- firsttime = true;
if(stages >= MAX_FILTER_STAGES)
stages = MAX_FILTER_STAGES;
+ outgain = 1.0f;
cleanup();
setfreq_and_q(Ffreq, Fq);
}
@@ -53,24 +50,20 @@ SVFilter::~SVFilter()
void SVFilter::cleanup()
{
- for(int i = 0; i < MAX_FILTER_STAGES + 1; ++i) {
- st[i].low = 0.0f;
- st[i].high = 0.0f;
- st[i].band = 0.0f;
- st[i].notch = 0.0f;
- }
+ for(int i = 0; i < MAX_FILTER_STAGES + 1; ++i)
+ st[i].low = st[i].high = st[i].band = st[i].notch = 0.0f;
oldabovenq = false;
abovenq = false;
}
-void SVFilter::computefiltercoefs()
+void SVFilter::computefiltercoefs(void)
{
par.f = freq / SAMPLE_RATE * 4.0f;
if(par.f > 0.99999f)
par.f = 0.99999f;
- par.q = 1.0f - atanf(sqrt(q)) * 2.0f / PI;
+ par.q = 1.0f - atanf(sqrtf(q)) * 2.0f / PI;
par.q = powf(par.q, 1.0f / (stages + 1));
- par.q_sqrt = sqrt(par.q);
+ par.q_sqrt = sqrtf(par.q);
}
@@ -89,9 +82,9 @@ void SVFilter::setfreq(float frequency)
//if the frequency is changed fast, it needs interpolation
if((rap > 3.0f) || nyquistthresh) { //(now, filter and coeficients backup)
- ipar = par;
if(!firsttime)
needsinterpolation = true;
+ ipar = par;
}
freq = frequency;
computefiltercoefs();
@@ -156,8 +149,7 @@ void SVFilter::singlefilterout(float *smp, fstage &x, parameters &par)
x.high = par.q_sqrt * smp[i] - x.low - par.q * x.band;
x.band = par.f * x.high + x.band;
x.notch = x.high + x.low;
-
- smp[i] = *out;
+ smp[i] = *out;
}
}
diff --git a/src/DSP/SVFilter.h b/src/DSP/SVFilter.h
@@ -25,6 +25,7 @@
#include "../globals.h"
#include "Filter.h"
+
class SVFilter:public Filter
{
public:
@@ -52,19 +53,17 @@ class SVFilter:public Filter
float f, q, q_sqrt;
} par, ipar;
-
void singlefilterout(float *smp, fstage &x, parameters &par);
- void computefiltercoefs();
- int type; //The type of the filter (LPF1,HPF1,LPF2,HPF2...)
- int stages; //how many times the filter is applied (0->1,1->2,etc.)
- float freq; //Frequency given in Hz
- float q; //Q factor (resonance or Q factor)
- float gain; //the gain of the filter (if are shelf/peak) filters
+ void computefiltercoefs(void);
+ int type; // The type of the filter (LPF1,HPF1,LPF2,HPF2...)
+ int stages; // how many times the filter is applied (0->1,1->2,etc.)
+ float freq; // Frequency given in Hz
+ float q; // Q factor (resonance or Q factor)
+ float gain; // the gain of the filter (if are shelf/peak) filters
bool abovenq, //if the frequency is above the nyquist
oldabovenq;
bool needsinterpolation, firsttime;
};
-
#endif
diff --git a/src/DSP/Unison.cpp b/src/DSP/Unison.cpp
@@ -19,36 +19,38 @@
Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
-#include <math.h>
-#include <stdio.h>
+#include <cmath>
+#include <cstring>
+
#include "Unison.h"
-Unison::Unison(int update_period_samples_, float max_delay_sec_) {
- update_period_samples = update_period_samples_;
- max_delay = (int)(max_delay_sec_ * (float)SAMPLE_RATE + 1);
+Unison::Unison(int update_period_samples_, float max_delay_sec_)
+ :unison_size(0),
+ base_freq(1.0f),
+ uv(NULL),
+ update_period_samples(update_period_samples_),
+ update_period_sample_k(0),
+ max_delay((int)(max_delay_sec_ * (float)SAMPLE_RATE + 1)),
+ delay_k(0),
+ first_time(false),
+ delay_buffer(NULL),
+ unison_amplitude_samples(0.0f),
+ unison_bandwidth_cents(10.0f)
+{
if(max_delay < 10)
max_delay = 10;
delay_buffer = new float[max_delay];
- delay_k = 0;
- base_freq = 1.0f;
- unison_bandwidth_cents = 10.0f;
-
- ZERO_float(delay_buffer, max_delay);
-
- uv = NULL;
- update_period_sample_k = 0;
- first_time = 0;
-
- set_size(1);
+ memset(delay_buffer, 0, max_delay * sizeof(float));
+ setSize(1);
}
Unison::~Unison() {
delete [] delay_buffer;
- if(uv)
- delete [] uv;
+ delete [] uv;
}
-void Unison::set_size(int new_size) {
+void Unison::setSize(int new_size)
+{
if(new_size < 1)
new_size = 1;
unison_size = new_size;
@@ -56,28 +58,30 @@ void Unison::set_size(int new_size) {
delete [] uv;
uv = new UnisonVoice[unison_size];
first_time = true;
- update_parameters();
+ updateParameters();
}
-void Unison::set_base_frequency(float freq) {
+void Unison::setBaseFrequency(float freq)
+{
base_freq = freq;
- update_parameters();
+ updateParameters();
}
-void Unison::set_bandwidth(float bandwidth) {
+void Unison::setBandwidth(float bandwidth)
+{
if(bandwidth < 0)
bandwidth = 0.0f;
if(bandwidth > 1200.0f)
bandwidth = 1200.0f;
- printf("bandwidth %g\n", bandwidth);
#warning \
: todo: if bandwidth is too small the audio will be self canceled (because of the sign change of the outputs)
unison_bandwidth_cents = bandwidth;
- update_parameters();
+ updateParameters();
}
-void Unison::update_parameters() {
+void Unison::updateParameters(void)
+{
if(!uv)
return;
float increments_per_second = SAMPLE_RATE
@@ -95,47 +99,44 @@ void Unison::update_parameters() {
}
float max_speed = powf(2.0f, unison_bandwidth_cents / 1200.0f);
- unison_amplitude_samples = 0.125f
- * (max_speed - 1.0f) * SAMPLE_RATE / base_freq;
- printf("unison_amplitude_samples %g\n", unison_amplitude_samples);
+ unison_amplitude_samples = 0.125f * (max_speed - 1.0f)
+ * SAMPLE_RATE / base_freq;
#warning \
todo: test if unison_amplitude_samples is to big and reallocate bigger memory
if(unison_amplitude_samples >= max_delay - 1)
unison_amplitude_samples = max_delay - 2;
- update_unison_data();
+ updateUnisonData();
}
-void Unison::process(int bufsize, float *inbuf, float *outbuf) {
+void Unison::process(int bufsize, float *inbuf, float *outbuf)
+{
if(!uv)
return;
if(!outbuf)
outbuf = inbuf;
- float volume = 1.0f / sqrt(unison_size);
+ float volume = 1.0f / sqrtf(unison_size);
float xpos_step = 1.0f / (float) update_period_samples;
float xpos = (float) update_period_sample_k * xpos_step;
for(int i = 0; i < bufsize; ++i) {
- if((update_period_sample_k++) >= update_period_samples) {
- update_unison_data();
+ if(update_period_sample_k++ >= update_period_samples) {
+ updateUnisonData();
update_period_sample_k = 0;
xpos = 0.0f;
}
xpos += xpos_step;
- float in = inbuf[i], out = 0.0f;
-
+ float in = inbuf[i], out = 0.0f;
float sign = 1.0f;
for(int k = 0; k < unison_size; ++k) {
- float vpos = uv[k].realpos1
- * (1.0f - xpos) + uv[k].realpos2 * xpos; //optimize
- float pos = delay_k + max_delay - vpos - 1.0f; //optimize
+ float vpos = uv[k].realpos1 * (1.0f - xpos) + uv[k].realpos2 * xpos; //optimize
+ float pos = (float)(delay_k + max_delay) - vpos - 1.0f;
int posi;
- float posf;
F2I(pos, posi); //optimize!
if(posi >= max_delay)
posi -= max_delay;
- posf = pos - floor(pos);
+ float posf = pos - floorf(pos);
out +=
((1.0f
- posf) * delay_buffer[posi] + posf
@@ -145,12 +146,12 @@ void Unison::process(int bufsize, float *inbuf, float *outbuf) {
outbuf[i] = out * volume;
// printf("%d %g\n",i,outbuf[i]);
delay_buffer[delay_k] = in;
- if((++delay_k) >= max_delay)
- delay_k = 0;
+ delay_k = (++delay_k < max_delay) ? delay_k : 0;
}
}
-void Unison::update_unison_data() {
+void Unison::updateUnisonData()
+{
if(!uv)
return;
@@ -162,7 +163,7 @@ void Unison::update_unison_data() {
pos = -1.0f;
step = -step;
}
- if(pos >= 1.0f) {
+ else if(pos >= 1.0f) {
pos = 1.0f;
step = -step;
}
@@ -172,9 +173,8 @@ void Unison::update_unison_data() {
#warning \
I have to enlarge (reallocate) the buffer to make place for the whole delay
float newval = 1.0f + 0.5f
- * (vibratto_val
- + 1.0f) * unison_amplitude_samples
- * uv[k].relative_amplitude;
+ * (vibratto_val + 1.0f) * unison_amplitude_samples
+ * uv[k].relative_amplitude;
if(first_time)
uv[k].realpos1 = uv[k].realpos2 = newval;
@@ -186,6 +186,5 @@ void Unison::update_unison_data() {
uv[k].position = pos;
uv[k].step = step;
}
- if(first_time)
- first_time = false;
+ first_time = false;
}
diff --git a/src/DSP/Unison.h b/src/DSP/Unison.h
@@ -21,11 +21,11 @@
#ifndef UNISON_H
#define UNISON_H
-#include <stdlib.h>
+
#include "../Misc/Util.h"
+//how much the unison frequencies varies (always >= 1.0)
#define UNISON_FREQ_SPAN 2.0f
-//how much the unison frequencies varies (always >= 1.0f)
class Unison
{
@@ -33,23 +33,26 @@ class Unison
Unison(int update_period_samples_, float max_delay_sec_);
~Unison();
- void set_size(int new_size);
- void set_base_frequency(float freq);
- void set_bandwidth(float bandwidth_cents);
+ void setSize(int new_size);
+ void setBaseFrequency(float freq);
+ void setBandwidth(float bandwidth_cents);
void process(int bufsize, float *inbuf, float *outbuf = NULL);
private:
- void update_parameters();
- void update_unison_data();
+ void updateParameters(void);
+ void updateUnisonData(void);
int unison_size;
float base_freq;
struct UnisonVoice {
- float step, position; //base LFO
- float realpos1, realpos2; //the position regarding samples
+ float step; //base LFO
+ float position;
+ float realpos1; //the position regarding samples
+ float realpos2;
float relative_amplitude;
- float lin_fpos, lin_ffreq;
+ float lin_fpos;
+ float lin_ffreq;
UnisonVoice() {
position = RND * 1.8f - 0.9f;
realpos1 = 0.0f;
@@ -58,7 +61,9 @@ class Unison
relative_amplitude = 1.0f;
}
} *uv;
- int update_period_samples, update_period_sample_k;
+
+ int update_period_samples;
+ int update_period_sample_k;
int max_delay, delay_k;
bool first_time;
float *delay_buffer;
diff --git a/src/Effects/Reverb.cpp b/src/Effects/Reverb.cpp
@@ -379,8 +379,8 @@ void Reverb::settype(unsigned char Ptype)
bandwidth = NULL;
if(Ptype == 2) { //bandwidth
bandwidth = new Unison(SOUND_BUFFER_SIZE / 4 + 1, 2.0f);
- bandwidth->set_size(50);
- bandwidth->set_base_frequency(1.0f);
+ bandwidth->setSize(50);
+ bandwidth->setBaseFrequency(1.0f);
#warning sa schimb size-ul
}
}
@@ -402,7 +402,7 @@ void Reverb::setbandwidth(unsigned char Pbandwidth) {
this->Pbandwidth = Pbandwidth;
float v = Pbandwidth / 127.0f;
if(bandwidth)
- bandwidth->set_bandwidth(powf(v, 2.0f) * 200.0f);
+ bandwidth->setBandwidth(powf(v, 2.0f) * 200.0f);
}
void Reverb::setpreset(unsigned char npreset)
