commit d2c9a44b870679f3c2660278424a309bb1917aca
parent 604c9ef7ee3f8f2c5a2093a97b75e090bcc494c8
Author: fundamental <[email protected]>
Date: Wed, 14 Sep 2011 16:18:39 -0400
Style: Restyling codebase
- Style shall now be defined in style.cfg and inforced with uncrustify
Diffstat:
149 files changed, 4723 insertions(+), 3411 deletions(-)
diff --git a/ExternalPrograms/Controller/Controller.h b/ExternalPrograms/Controller/Controller.h
@@ -31,4 +31,3 @@ class Controller
};
#endif
-
diff --git a/ExternalPrograms/Spliter/Spliter.h b/ExternalPrograms/Spliter/Spliter.h
@@ -25,4 +25,3 @@ class Spliter
};
#endif
-
diff --git a/src/DSP/AnalogFilter.cpp b/src/DSP/AnalogFilter.cpp
@@ -74,7 +74,7 @@ void AnalogFilter::cleanup()
void AnalogFilter::computefiltercoefs()
{
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;
@@ -103,196 +103,198 @@ void AnalogFilter::computefiltercoefs()
//General Constants
const float omega = 2 * PI * freq / SAMPLE_RATE;
- const float sn = sinf(omega), cs = cosf(omega);
- float alpha, beta;
+ const float sn = sinf(omega), cs = cosf(omega);
+ float alpha, beta;
//most of theese are implementations of
//the "Cookbook formulae for audio EQ" by Robert Bristow-Johnson
//The original location of the Cookbook is:
//http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt
switch(type) {
- case 0: //LPF 1 pole
- if(!zerocoefs)
- tmp = expf(-2.0f * PI * freq / SAMPLE_RATE);
- else
- tmp = 0.0f;
- c[0] = 1.0f - tmp;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = tmp;
- d[2] = 0.0f;
- order = 1;
- break;
- case 1: //HPF 1 pole
- if(!zerocoefs)
- tmp = expf(-2.0f * PI * freq / SAMPLE_RATE);
- else
- tmp = 0.0f;
- c[0] = (1.0f + tmp) / 2.0f;
- c[1] = -(1.0f + tmp) / 2.0f;
- c[2] = 0.0f;
- d[1] = tmp;
- d[2] = 0.0f;
- order = 1;
- break;
- case 2: //LPF 2 poles
- if(!zerocoefs) {
- alpha = sn / (2 * 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] = 1.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
- }
- order = 2;
- break;
- case 3: //HPF 2 poles
- if(!zerocoefs) {
- alpha = sn / (2 * 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;
- }
- 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;
- }
- 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);
- }
- else {
- c[0] = 1.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- 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;
- 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);
- }
- else {
- c[0] = 1.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- 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;
- tmp = (tmpgain + 1.0f) + (tmpgain - 1.0f) * cs + beta * sn;
-
- c[0] = tmpgain
- * ((tmpgain
- + 1.0f) - (tmpgain - 1.0f) * cs + beta * sn) / tmp;
- c[1] = 2.0f * tmpgain
- * ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs) / tmp;
- 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);
- }
- else {
- c[0] = tmpgain;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- 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;
- tmp = (tmpgain + 1.0f) - (tmpgain - 1.0f) * cs + beta * sn;
-
- c[0] = tmpgain
- * ((tmpgain
- + 1.0f) + (tmpgain - 1.0f) * cs + beta * sn) / tmp;
- c[1] = -2.0f * tmpgain
- * ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs) / tmp;
- 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);
- }
- else {
- c[0] = 1.0f;
- c[1] = 0.0f;
- c[2] = 0.0f;
- d[1] = 0.0f;
- d[2] = 0.0f;
- }
- order = 2;
- break;
- default: //wrong type
- type = 0;
- computefiltercoefs();
- break;
+ case 0: //LPF 1 pole
+ if(!zerocoefs)
+ tmp = expf(-2.0f * PI * freq / SAMPLE_RATE);
+ else
+ tmp = 0.0f;
+ c[0] = 1.0f - tmp;
+ c[1] = 0.0f;
+ c[2] = 0.0f;
+ d[1] = tmp;
+ d[2] = 0.0f;
+ order = 1;
+ break;
+ case 1: //HPF 1 pole
+ if(!zerocoefs)
+ tmp = expf(-2.0f * PI * freq / SAMPLE_RATE);
+ else
+ tmp = 0.0f;
+ c[0] = (1.0f + tmp) / 2.0f;
+ c[1] = -(1.0f + tmp) / 2.0f;
+ c[2] = 0.0f;
+ d[1] = tmp;
+ d[2] = 0.0f;
+ order = 1;
+ break;
+ case 2: //LPF 2 poles
+ if(!zerocoefs) {
+ alpha = sn / (2 * 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] = 1.0f;
+ c[1] = 0.0f;
+ c[2] = 0.0f;
+ d[1] = 0.0f;
+ d[2] = 0.0f;
+ }
+ order = 2;
+ break;
+ case 3: //HPF 2 poles
+ if(!zerocoefs) {
+ alpha = sn / (2 * 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;
+ }
+ 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;
+ }
+ 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);
+ }
+ else {
+ c[0] = 1.0f;
+ c[1] = 0.0f;
+ c[2] = 0.0f;
+ d[1] = 0.0f;
+ 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;
+ 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);
+ }
+ else {
+ c[0] = 1.0f;
+ c[1] = 0.0f;
+ c[2] = 0.0f;
+ d[1] = 0.0f;
+ 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;
+ tmp = (tmpgain + 1.0f) + (tmpgain - 1.0f) * cs + beta * sn;
+
+ c[0] = tmpgain
+ * ((tmpgain
+ + 1.0f) - (tmpgain - 1.0f) * cs + beta * sn) / tmp;
+ c[1] = 2.0f * tmpgain
+ * ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs) / tmp;
+ 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);
+ }
+ else {
+ c[0] = tmpgain;
+ c[1] = 0.0f;
+ c[2] = 0.0f;
+ d[1] = 0.0f;
+ 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;
+ tmp = (tmpgain + 1.0f) - (tmpgain - 1.0f) * cs + beta * sn;
+
+ c[0] = tmpgain
+ * ((tmpgain
+ + 1.0f) + (tmpgain - 1.0f) * cs + beta * sn) / tmp;
+ c[1] = -2.0f * tmpgain
+ * ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs) / tmp;
+ 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);
+ }
+ else {
+ c[0] = 1.0f;
+ c[1] = 0.0f;
+ c[2] = 0.0f;
+ d[1] = 0.0f;
+ d[2] = 0.0f;
+ }
+ order = 2;
+ break;
+ default: //wrong type
+ type = 0;
+ computefiltercoefs();
+ break;
}
}
@@ -313,13 +315,13 @@ void AnalogFilter::setfreq(float frequency)
//if the frequency is changed fast, it needs interpolation
if((rap > 3.0f) || nyquistthresh) { //(now, filter and coeficients backup)
- oldCoeff = coeff;
+ oldCoeff = coeff;
for(int i = 0; i < MAX_FILTER_STAGES + 1; ++i)
oldHistory[i] = history[i];
if(!firsttime)
needsinterpolation = true;
}
- freq = frequency;
+ freq = frequency;
computefiltercoefs();
firsttime = false;
}
@@ -360,27 +362,25 @@ void AnalogFilter::setstages(int stages_)
void AnalogFilter::singlefilterout(float *smp, fstage &hist,
const Coeff &coeff)
{
- if(order == 1) { //First order filter
+ if(order == 1) //First order filter
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- float y0 = smp[i]*coeff.c[0] + hist.x1*coeff.c[1]
- + hist.y1*coeff.d[1];
+ float y0 = smp[i] * coeff.c[0] + hist.x1 * coeff.c[1]
+ + hist.y1 * coeff.d[1];
hist.y1 = y0;
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]
- + hist.y2*coeff.d[2];
+ float y0 = smp[i] * coeff.c[0] + hist.x1 * coeff.c[1]
+ + hist.x2 * coeff.c[2] + hist.y1 * coeff.d[1]
+ + hist.y2 * coeff.d[2];
hist.y2 = hist.y1;
hist.y1 = y0;
hist.x2 = hist.x1;
hist.x1 = smp[i];
smp[i] = y0;
}
- }
}
void AnalogFilter::filterout(float *smp)
{
@@ -425,4 +425,3 @@ float AnalogFilter::H(float freq)
h = h / (x * x + y * y);
return powf(h, (stages + 1.0f) / 2.0f);
}
-
diff --git a/src/DSP/AnalogFilter.h b/src/DSP/AnalogFilter.h
@@ -53,13 +53,13 @@ class AnalogFilter:public Filter
private:
struct fstage {
- float x1, x2;//Input History
- float y1, y2;//Output History
+ float x1, x2; //Input History
+ float y1, y2; //Output History
} history[MAX_FILTER_STAGES + 1], oldHistory[MAX_FILTER_STAGES + 1];
struct Coeff {
float c[3], //Feed Forward
- d[3]; //Feed Back
+ d[3]; //Feed Back
} coeff, oldCoeff;
//old coeffs are used for interpolation when paremeters change quickly
@@ -68,21 +68,20 @@ class AnalogFilter:public Filter
//Update coeff and order
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.)
+ 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)
- bool needsinterpolation, //Interpolation between coeff changes
- firsttime; //First Iteration of filter
- bool abovenq, //if the frequency is above the nyquist
- oldabovenq; //if the last time was above nyquist
+ bool needsinterpolation, //Interpolation between coeff changes
+ firsttime; //First Iteration of filter
+ bool abovenq, //if the frequency is above the nyquist
+ oldabovenq; //if the last time was above nyquist
//(used to see if it needs interpolation)
};
#endif
-
diff --git a/src/DSP/FFTwrapper.cpp b/src/DSP/FFTwrapper.cpp
@@ -59,17 +59,17 @@ void FFTwrapper::smps2freqs(const float *smps, fft_t *freqs)
fftw_execute(planfftw);
//Grab data
- memcpy((void*)freqs, (const void*)fft, fftsize*sizeof(double));
+ memcpy((void *)freqs, (const void *)fft, fftsize * sizeof(double));
}
void FFTwrapper::freqs2smps(const fft_t *freqs, float *smps)
{
//Load data
- memcpy( (void*)fft, (const void*)freqs, fftsize*sizeof(double));
+ memcpy((void *)fft, (const void *)freqs, fftsize * sizeof(double));
//clear unused freq channel
- fft[fftsize/2][0] = 0.0f;
- fft[fftsize/2][1] = 0.0f;
+ fft[fftsize / 2][0] = 0.0f;
+ fft[fftsize / 2][1] = 0.0f;
//IDFT
fftw_execute(planfftw_inv);
@@ -83,4 +83,3 @@ void FFT_cleanup()
{
fftw_cleanup();
}
-
diff --git a/src/DSP/FFTwrapper.h b/src/DSP/FFTwrapper.h
@@ -24,7 +24,7 @@
#define FFT_WRAPPER_H
#include <fftw3.h>
#include <complex>
-typedef double fftw_real;
+typedef double fftw_real;
typedef std::complex<fftw_real> fft_t;
/**A wrapper for the FFTW library (Fast Fourier Transforms)*/
@@ -42,12 +42,11 @@ class FFTwrapper
void smps2freqs(const float *smps, fft_t *freqs);
void freqs2smps(const fft_t *freqs, float *smps);
private:
- int fftsize;
+ int fftsize;
fftw_real *time;
fftw_complex *fft;
- fftw_plan planfftw, planfftw_inv;
+ fftw_plan planfftw, planfftw_inv;
};
void FFT_cleanup();
#endif
-
diff --git a/src/DSP/Filter.cpp b/src/DSP/Filter.cpp
@@ -36,22 +36,22 @@ Filter *Filter::generate(FilterParams *pars)
Filter *filter;
switch(pars->Pcategory) {
- case 1:
- filter = new FormantFilter(pars);
- break;
- case 2:
- filter = new SVFilter(Ftype, 1000.0f, pars->getq(), Fstages);
- filter->outgain = dB2rap(pars->getgain());
- if(filter->outgain > 1.0f)
- filter->outgain = sqrt(filter->outgain);
- break;
- default:
- filter = new AnalogFilter(Ftype, 1000.0f, pars->getq(), Fstages);
- if((Ftype >= 6) && (Ftype <= 8))
- filter->setgain(pars->getgain());
- else
+ case 1:
+ filter = new FormantFilter(pars);
+ break;
+ case 2:
+ filter = new SVFilter(Ftype, 1000.0f, pars->getq(), Fstages);
filter->outgain = dB2rap(pars->getgain());
- break;
+ if(filter->outgain > 1.0f)
+ filter->outgain = sqrt(filter->outgain);
+ break;
+ default:
+ filter = new AnalogFilter(Ftype, 1000.0f, pars->getq(), Fstages);
+ if((Ftype >= 6) && (Ftype <= 8))
+ filter->setgain(pars->getgain());
+ else
+ filter->outgain = dB2rap(pars->getgain());
+ break;
}
return filter;
}
@@ -60,4 +60,3 @@ float Filter::getrealfreq(float freqpitch)
{
return powf(2.0f, freqpitch + 9.96578428f); //log2(1000)=9.95748f
}
-
diff --git a/src/DSP/Filter.h b/src/DSP/Filter.h
@@ -29,7 +29,7 @@ class Filter
{
public:
static float getrealfreq(float freqpitch);
- static Filter *generate(class FilterParams *pars);
+ static Filter *generate(class FilterParams * pars);
virtual ~Filter() {}
virtual void filterout(float *smp) = 0;
@@ -43,4 +43,3 @@ class Filter
};
#endif
-
diff --git a/src/DSP/FormantFilter.cpp b/src/DSP/FormantFilter.cpp
@@ -38,9 +38,9 @@ FormantFilter::FormantFilter(FilterParams *pars)
for(int i = 0; i < numformants; ++i) {
formantpar[j][i].freq = pars->getformantfreq(
pars->Pvowels[j].formants[i].freq);
- formantpar[j][i].amp = pars->getformantamp(
+ formantpar[j][i].amp = pars->getformantamp(
pars->Pvowels[j].formants[i].amp);
- formantpar[j][i].q = pars->getformantq(
+ formantpar[j][i].q = pars->getformantq(
pars->Pvowels[j].formants[i].q);
}
@@ -54,19 +54,19 @@ FormantFilter::FormantFilter(FilterParams *pars)
formantslowness = powf(1.0f - (pars->Pformantslowness / 128.0f), 3.0f);
- sequencesize = pars->Psequencesize;
+ sequencesize = pars->Psequencesize;
if(sequencesize == 0)
sequencesize = 1;
for(int k = 0; k < sequencesize; ++k)
sequence[k].nvowel = pars->Psequence[k].nvowel;
- vowelclearness = powf(10.0f, (pars->Pvowelclearness - 32.0f) / 48.0f);
+ vowelclearness = powf(10.0f, (pars->Pvowelclearness - 32.0f) / 48.0f);
sequencestretch = powf(0.1f, (pars->Psequencestretch - 32.0f) / 48.0f);
if(pars->Psequencereversed)
sequencestretch *= -1.0f;
- outgain = dB2rap(pars->getgain());
+ outgain = dB2rap(pars->getgain());
oldinput = -1.0f;
Qfactor = 1.0f;
@@ -123,7 +123,8 @@ void FormantFilter::setpos(float input)
pos = 1.0f;
pos =
(atanf((pos * 2.0f
- - 1.0f) * vowelclearness) / atanf(vowelclearness) + 1.0f) * 0.5f;
+ - 1.0f)
+ * vowelclearness) / atanf(vowelclearness) + 1.0f) * 0.5f;
p1 = sequence[p1].nvowel;
p2 = sequence[p2].nvowel;
@@ -133,18 +134,18 @@ void FormantFilter::setpos(float input)
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].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;
}
firsttime = 0;
}
- else {
+ else
for(int i = 0; i < numformants; ++i) {
currentformants[i].freq = currentformants[i].freq
* (1.0f - formantslowness)
@@ -170,7 +171,6 @@ void FormantFilter::setpos(float input)
formant[i]->setfreq_and_q(currentformants[i].freq,
currentformants[i].q * Qfactor);
}
- }
oldQfactor = Qfactor;
}
@@ -192,7 +192,7 @@ void FormantFilter::setgain(float /*dBgain*/)
inline float log_2(float x)
{
- return logf(x)/logf(2.0f);
+ return logf(x) / logf(2.0f);
}
void FormantFilter::setfreq_and_q(float frequency, float q_)
@@ -233,4 +233,3 @@ void FormantFilter::filterout(float *smp)
}
returnTmpBuffer(inbuffer);
}
-
diff --git a/src/DSP/FormantFilter.h b/src/DSP/FormantFilter.h
@@ -40,7 +40,7 @@ class FormantFilter:public Filter
void cleanup();
private:
- class AnalogFilter *formant[FF_MAX_FORMANTS];
+ class AnalogFilter * formant[FF_MAX_FORMANTS];
struct {
float freq, amp, q; //frequency,amplitude,Q
@@ -53,7 +53,7 @@ class FormantFilter:public Filter
float oldformantamp[FF_MAX_FORMANTS];
- int sequencesize, numformants, firsttime;
+ int sequencesize, numformants, firsttime;
float oldinput, slowinput;
float Qfactor, formantslowness, oldQfactor;
float vowelclearness, sequencestretch;
@@ -62,4 +62,3 @@ class FormantFilter:public Filter
};
#endif
-
diff --git a/src/DSP/SVFilter.cpp b/src/DSP/SVFilter.cpp
@@ -33,13 +33,13 @@ SVFilter::SVFilter(unsigned char Ftype,
float Fq,
unsigned char Fstages)
{
- assert(Ftype<4);
- stages = Fstages;
- type = Ftype;
- freq = Ffreq;
- q = Fq;
- gain = 1.0f;
- outgain = 1.0f;
+ 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)
@@ -65,7 +65,7 @@ void SVFilter::cleanup()
void SVFilter::computefiltercoefs()
{
- par.f = freq / SAMPLE_RATE * 4.0f;
+ 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;
@@ -88,12 +88,12 @@ void SVFilter::setfreq(float frequency)
bool nyquistthresh = (abovenq ^ oldabovenq);
//if the frequency is changed fast, it needs interpolation
- if((rap > 3.0f) || nyquistthresh) {//(now, filter and coeficients backup)
+ if((rap > 3.0f) || nyquistthresh) { //(now, filter and coeficients backup)
ipar = par;
if(!firsttime)
needsinterpolation = true;
}
- freq = frequency;
+ freq = frequency;
computefiltercoefs();
firsttime = false;
}
@@ -135,20 +135,20 @@ void SVFilter::singlefilterout(float *smp, fstage &x, parameters &par)
{
float *out = NULL;
switch(type) {
- case 0:
- out = &x.low;
- break;
- case 1:
- out = &x.high;
- break;
- case 2:
- out = &x.band;
- break;
- case 3:
- out = &x.notch;
- break;
- default:
- errx(1, "Impossible SVFilter type encountered [%d]", type);
+ case 0:
+ out = &x.low;
+ break;
+ case 1:
+ out = &x.high;
+ break;
+ case 2:
+ out = &x.band;
+ break;
+ case 3:
+ out = &x.notch;
+ break;
+ default:
+ errx(1, "Impossible SVFilter type encountered [%d]", type);
}
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
@@ -157,7 +157,7 @@ void SVFilter::singlefilterout(float *smp, fstage &x, parameters &par)
x.band = par.f * x.high + x.band;
x.notch = x.high + x.low;
- smp[i] = *out;
+ smp[i] = *out;
}
}
@@ -184,4 +184,3 @@ void SVFilter::filterout(float *smp)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
smp[i] *= outgain;
}
-
diff --git a/src/DSP/SVFilter.h b/src/DSP/SVFilter.h
@@ -55,8 +55,8 @@ class SVFilter:public Filter
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.)
+ 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
@@ -68,4 +68,3 @@ class SVFilter:public Filter
#endif
-
diff --git a/src/DSP/Unison.cpp b/src/DSP/Unison.cpp
@@ -25,7 +25,7 @@
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);
+ max_delay = (int)(max_delay_sec_ * (float)SAMPLE_RATE + 1);
if(max_delay < 10)
max_delay = 10;
delay_buffer = new float[max_delay];
@@ -81,10 +81,10 @@ void Unison::update_parameters() {
if(!uv)
return;
float increments_per_second = SAMPLE_RATE
- / (float) update_period_samples;
+ / (float) update_period_samples;
// printf("#%g, %g\n",increments_per_second,base_freq);
for(int i = 0; i < unison_size; ++i) {
- float base = powf(UNISON_FREQ_SPAN, RND * 2.0f - 1.0f);
+ float base = powf(UNISON_FREQ_SPAN, RND * 2.0f - 1.0f);
uv[i].relative_amplitude = base;
float period = base / base_freq;
float m = 4.0f / (period * increments_per_second);
@@ -123,14 +123,14 @@ void Unison::process(int bufsize, float *inbuf, float *outbuf) {
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
- int posi;
+ * (1.0f - xpos) + uv[k].realpos2 * xpos; //optimize
+ float pos = delay_k + max_delay - vpos - 1.0f; //optimize
+ int posi;
float posf;
F2I(pos, posi); //optimize!
if(posi >= max_delay)
@@ -172,13 +172,13 @@ 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;
- else{
+ else {
uv[k].realpos1 = uv[k].realpos2;
uv[k].realpos2 = newval;
}
@@ -189,4 +189,3 @@ void Unison::update_unison_data() {
if(first_time)
first_time = false;
}
-
diff --git a/src/DSP/Unison.h b/src/DSP/Unison.h
@@ -43,7 +43,7 @@ class Unison
void update_parameters();
void update_unison_data();
- int unison_size;
+ int unison_size;
float base_freq;
struct UnisonVoice {
float step, position; //base LFO
@@ -51,19 +51,18 @@ class Unison
float relative_amplitude;
float lin_fpos, lin_ffreq;
UnisonVoice() {
- position = RND * 1.8f - 0.9f;
- realpos1 = 0.0f;
- realpos2 = 0.0f;
- step = 0.0f;
- relative_amplitude = 1.0f;
+ position = RND * 1.8f - 0.9f;
+ realpos1 = 0.0f;
+ realpos2 = 0.0f;
+ step = 0.0f;
+ relative_amplitude = 1.0f;
}
} *uv;
- int update_period_samples, update_period_sample_k;
- int max_delay, delay_k;
- bool first_time;
+ int update_period_samples, update_period_sample_k;
+ int max_delay, delay_k;
+ bool first_time;
float *delay_buffer;
float unison_amplitude_samples;
float unison_bandwidth_cents;
};
#endif
-
diff --git a/src/Effects/Alienwah.cpp b/src/Effects/Alienwah.cpp
@@ -66,10 +66,10 @@ void Alienwah::out(const Stereo<float *> &smp)
//left
complex<float> tmp = clfol * x + oldclfol * x1;
- complex<float>out = tmp * oldl[oldk];
+ complex<float> out = tmp * oldl[oldk];
out.real() += (1 - fabs(fb)) * smp.l[i] * pangainL;
- oldl[oldk] = out;
+ oldl[oldk] = out;
float l = out.real() * 10.0f * (fb + 0.1f);
//right
@@ -78,7 +78,7 @@ void Alienwah::out(const Stereo<float *> &smp)
out = tmp * oldr[oldk];
out.real() += (1 - fabs(fb)) * smp.r[i] * pangainR;
- oldr[oldk] = out;
+ oldr[oldk] = out;
float r = out.real() * 10.0f * (fb + 0.1f);
@@ -164,13 +164,13 @@ void Alienwah::setpreset(unsigned char npreset)
const int NUM_PRESETS = 4;
unsigned char presets[NUM_PRESETS][PRESET_SIZE] = {
//AlienWah1
- {127, 64, 70, 0, 0, 62, 60, 105, 25, 0, 64},
+ {127, 64, 70, 0, 0, 62, 60, 105, 25, 0, 64},
//AlienWah2
- {127, 64, 73, 106, 0, 101, 60, 105, 17, 0, 64},
+ {127, 64, 73, 106, 0, 101, 60, 105, 17, 0, 64},
//AlienWah3
- {127, 64, 63, 0, 1, 100, 112, 105, 31, 0, 42},
+ {127, 64, 63, 0, 1, 100, 112, 105, 31, 0, 42},
//AlienWah4
- {93, 64, 25, 0, 1, 66, 101, 11, 47, 0, 86}
+ {93, 64, 25, 0, 1, 66, 101, 11, 47, 0, 86}
};
if(npreset >= NUM_PRESETS)
diff --git a/src/Effects/Alienwah.h b/src/Effects/Alienwah.h
@@ -78,4 +78,3 @@ class Alienwah:public Effect
};
#endif
-
diff --git a/src/Effects/Chorus.cpp b/src/Effects/Chorus.cpp
@@ -79,7 +79,6 @@ void Chorus::out(const Stereo<float *> &input)
dr2 = getdelay(lfor);
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
-
float inl = input.l[i];
float inr = input.r[i];
//LRcross
@@ -110,7 +109,7 @@ void Chorus::out(const Stereo<float *> &input)
mdel = (dr1 * (SOUND_BUFFER_SIZE - i) + dr2 * i) / SOUND_BUFFER_SIZE;
if(++drk >= maxdelay)
drk = 0;
- tmp = drk * 1.0f - mdel + maxdelay * 2.0f; //where should I get the sample from
+ tmp = drk * 1.0f - mdel + maxdelay * 2.0f; //where should I get the sample from
F2I(tmp, dlhi);
dlhi %= maxdelay;
@@ -180,25 +179,25 @@ void Chorus::setpreset(unsigned char npreset)
const int NUM_PRESETS = 10;
unsigned char presets[NUM_PRESETS][PRESET_SIZE] = {
//Chorus1
- {64, 64, 50, 0, 0, 90, 40, 85, 64, 119, 0, 0 },
+ {64, 64, 50, 0, 0, 90, 40, 85, 64, 119, 0, 0 },
//Chorus2
- {64, 64, 45, 0, 0, 98, 56, 90, 64, 19, 0, 0 },
+ {64, 64, 45, 0, 0, 98, 56, 90, 64, 19, 0, 0 },
//Chorus3
- {64, 64, 29, 0, 1, 42, 97, 95, 90, 127, 0, 0 },
+ {64, 64, 29, 0, 1, 42, 97, 95, 90, 127, 0, 0 },
//Celeste1
- {64, 64, 26, 0, 0, 42, 115, 18, 90, 127, 0, 0 },
+ {64, 64, 26, 0, 0, 42, 115, 18, 90, 127, 0, 0 },
//Celeste2
- {64, 64, 29, 117, 0, 50, 115, 9, 31, 127, 0, 1 },
+ {64, 64, 29, 117, 0, 50, 115, 9, 31, 127, 0, 1 },
//Flange1
- {64, 64, 57, 0, 0, 60, 23, 3, 62, 0, 0, 0 },
+ {64, 64, 57, 0, 0, 60, 23, 3, 62, 0, 0, 0 },
//Flange2
- {64, 64, 33, 34, 1, 40, 35, 3, 109, 0, 0, 0 },
+ {64, 64, 33, 34, 1, 40, 35, 3, 109, 0, 0, 0 },
//Flange3
- {64, 64, 53, 34, 1, 94, 35, 3, 54, 0, 0, 1 },
+ {64, 64, 53, 34, 1, 94, 35, 3, 54, 0, 0, 1 },
//Flange4
- {64, 64, 40, 0, 1, 62, 12, 19, 97, 0, 0, 0 },
+ {64, 64, 40, 0, 1, 62, 12, 19, 97, 0, 0, 0 },
//Flange5
- {64, 64, 55, 105, 0, 24, 39, 19, 17, 0, 0, 1 }
+ {64, 64, 55, 105, 0, 24, 39, 19, 17, 0, 0, 1 }
};
if(npreset >= NUM_PRESETS)
@@ -273,4 +272,3 @@ unsigned char Chorus::getpar(int npar) const
default: return 0;
}
}
-
diff --git a/src/Effects/Chorus.h b/src/Effects/Chorus.h
@@ -99,7 +99,7 @@ class Chorus:public Effect
//Internal Values
float depth, delay, fb;
float dl1, dl2, dr1, dr2, lfol, lfor;
- int maxdelay;
+ int maxdelay;
Stereo<Sample> delaySample;
int dlk, drk, dlhi, dlhi2;
float getdelay(float xlfo);
@@ -107,4 +107,3 @@ class Chorus:public Effect
};
#endif
-
diff --git a/src/Effects/Distorsion.cpp b/src/Effects/Distorsion.cpp
@@ -38,13 +38,13 @@ Distorsion::Distorsion(const int &insertion_,
//default values
Pvolume = 50;
- Pdrive = 90;
- Plevel = 64;
- Ptype = 0;
- Pnegate = 0;
- Plpf = 127;
- Phpf = 0;
- Pstereo = 0;
+ Pdrive = 90;
+ Plevel = 64;
+ Ptype = 0;
+ Pnegate = 0;
+ Plpf = 127;
+ Phpf = 0;
+ Pstereo = 0;
Pprefiltering = 0;
setpreset(Ppreset);
@@ -91,23 +91,21 @@ void Distorsion::applyfilters(float *efxoutl, float *efxoutr)
*/
void Distorsion::out(const Stereo<float *> &smp)
{
- int i;
+ int i;
float l, r, lout, rout;
float inputvol = powf(5.0f, (Pdrive - 32.0f) / 127.0f);
if(Pnegate != 0)
inputvol *= -1.0f;
- if(Pstereo != 0) { //Stereo
+ if(Pstereo != 0) //Stereo
for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] = smp.l[i] * inputvol * pangainL;
efxoutr[i] = smp.r[i] * inputvol * pangainR;
}
- }
- else {
+ else
for(i = 0; i < SOUND_BUFFER_SIZE; ++i)
efxoutl[i] = (smp.l[i] * pangainL + smp.r[i] * pangainR) * inputvol;
- }
if(Pprefiltering != 0)
applyfilters(efxoutl, efxoutr);
@@ -180,17 +178,17 @@ void Distorsion::setpreset(unsigned char npreset)
const int NUM_PRESETS = 6;
unsigned char presets[NUM_PRESETS][PRESET_SIZE] = {
//Overdrive 1
- {127, 64, 35, 56, 70, 0, 0, 96, 0, 0, 0 },
+ {127, 64, 35, 56, 70, 0, 0, 96, 0, 0, 0 },
//Overdrive 2
- {127, 64, 35, 29, 75, 1, 0, 127, 0, 0, 0 },
+ {127, 64, 35, 29, 75, 1, 0, 127, 0, 0, 0 },
//A. Exciter 1
- {64, 64, 35, 75, 80, 5, 0, 127, 105, 1, 0 },
+ {64, 64, 35, 75, 80, 5, 0, 127, 105, 1, 0 },
//A. Exciter 2
- {64, 64, 35, 85, 62, 1, 0, 127, 118, 1, 0 },
+ {64, 64, 35, 85, 62, 1, 0, 127, 118, 1, 0 },
//Guitar Amp
- {127, 64, 35, 63, 75, 2, 0, 55, 0, 0, 0 },
+ {127, 64, 35, 63, 75, 2, 0, 55, 0, 0, 0 },
//Quantisize
- {127, 64, 35, 88, 75, 4, 0, 127, 0, 1, 0 }
+ {127, 64, 35, 88, 75, 4, 0, 127, 0, 1, 0 }
};
@@ -267,4 +265,3 @@ unsigned char Distorsion::getpar(int npar) const
default: return 0;
}
}
-
diff --git a/src/Effects/Distorsion.h b/src/Effects/Distorsion.h
@@ -56,9 +56,8 @@ class Distorsion:public Effect
void sethpf(unsigned char Phpf);
//Real Parameters
- class AnalogFilter *lpfl, *lpfr, *hpfl, *hpfr;
+ class AnalogFilter * lpfl, *lpfr, *hpfl, *hpfr;
};
#endif
-
diff --git a/src/Effects/DynamicFilter.cpp b/src/Effects/DynamicFilter.cpp
@@ -151,15 +151,15 @@ void DynamicFilter::setpreset(unsigned char npreset)
const int NUM_PRESETS = 5;
unsigned char presets[NUM_PRESETS][PRESET_SIZE] = {
//WahWah
- {110, 64, 80, 0, 0, 64, 0, 90, 0, 60},
+ {110, 64, 80, 0, 0, 64, 0, 90, 0, 60},
//AutoWah
- {110, 64, 70, 0, 0, 80, 70, 0, 0, 60},
+ {110, 64, 70, 0, 0, 80, 70, 0, 0, 60},
//Sweep
- {100, 64, 30, 0, 0, 50, 80, 0, 0, 60},
+ {100, 64, 30, 0, 0, 50, 80, 0, 0, 60},
//VocalMorph1
- {110, 64, 80, 0, 0, 64, 0, 64, 0, 60},
+ {110, 64, 80, 0, 0, 64, 0, 64, 0, 60},
//VocalMorph1
- {127, 64, 50, 0, 0, 96, 64, 0, 0, 60}
+ {127, 64, 50, 0, 0, 96, 64, 0, 0, 60}
};
if(npreset >= NUM_PRESETS)
@@ -169,86 +169,86 @@ void DynamicFilter::setpreset(unsigned char npreset)
filterpars->defaults();
switch(npreset) {
- case 0:
- filterpars->Pcategory = 0;
- filterpars->Ptype = 2;
- filterpars->Pfreq = 45;
- filterpars->Pq = 64;
- filterpars->Pstages = 1;
- filterpars->Pgain = 64;
- break;
- case 1:
- filterpars->Pcategory = 2;
- filterpars->Ptype = 0;
- filterpars->Pfreq = 72;
- filterpars->Pq = 64;
- filterpars->Pstages = 0;
- filterpars->Pgain = 64;
- break;
- case 2:
- filterpars->Pcategory = 0;
- filterpars->Ptype = 4;
- filterpars->Pfreq = 64;
- filterpars->Pq = 64;
- filterpars->Pstages = 2;
- filterpars->Pgain = 64;
- break;
- case 3:
- filterpars->Pcategory = 1;
- filterpars->Ptype = 0;
- filterpars->Pfreq = 50;
- filterpars->Pq = 70;
- filterpars->Pstages = 1;
- filterpars->Pgain = 64;
-
- filterpars->Psequencesize = 2;
- // "I"
- filterpars->Pvowels[0].formants[0].freq = 34;
- filterpars->Pvowels[0].formants[0].amp = 127;
- filterpars->Pvowels[0].formants[0].q = 64;
- filterpars->Pvowels[0].formants[1].freq = 99;
- filterpars->Pvowels[0].formants[1].amp = 122;
- filterpars->Pvowels[0].formants[1].q = 64;
- filterpars->Pvowels[0].formants[2].freq = 108;
- filterpars->Pvowels[0].formants[2].amp = 112;
- filterpars->Pvowels[0].formants[2].q = 64;
- // "A"
- filterpars->Pvowels[1].formants[0].freq = 61;
- filterpars->Pvowels[1].formants[0].amp = 127;
- filterpars->Pvowels[1].formants[0].q = 64;
- filterpars->Pvowels[1].formants[1].freq = 71;
- filterpars->Pvowels[1].formants[1].amp = 121;
- filterpars->Pvowels[1].formants[1].q = 64;
- filterpars->Pvowels[1].formants[2].freq = 99;
- filterpars->Pvowels[1].formants[2].amp = 117;
- filterpars->Pvowels[1].formants[2].q = 64;
- break;
- case 4:
- filterpars->Pcategory = 1;
- filterpars->Ptype = 0;
- filterpars->Pfreq = 64;
- filterpars->Pq = 70;
- filterpars->Pstages = 1;
- filterpars->Pgain = 64;
-
- filterpars->Psequencesize = 2;
- filterpars->Pnumformants = 2;
- filterpars->Pvowelclearness = 0;
-
- filterpars->Pvowels[0].formants[0].freq = 70;
- filterpars->Pvowels[0].formants[0].amp = 127;
- filterpars->Pvowels[0].formants[0].q = 64;
- filterpars->Pvowels[0].formants[1].freq = 80;
- filterpars->Pvowels[0].formants[1].amp = 122;
- filterpars->Pvowels[0].formants[1].q = 64;
-
- filterpars->Pvowels[1].formants[0].freq = 20;
- filterpars->Pvowels[1].formants[0].amp = 127;
- filterpars->Pvowels[1].formants[0].q = 64;
- filterpars->Pvowels[1].formants[1].freq = 100;
- filterpars->Pvowels[1].formants[1].amp = 121;
- filterpars->Pvowels[1].formants[1].q = 64;
- break;
+ case 0:
+ filterpars->Pcategory = 0;
+ filterpars->Ptype = 2;
+ filterpars->Pfreq = 45;
+ filterpars->Pq = 64;
+ filterpars->Pstages = 1;
+ filterpars->Pgain = 64;
+ break;
+ case 1:
+ filterpars->Pcategory = 2;
+ filterpars->Ptype = 0;
+ filterpars->Pfreq = 72;
+ filterpars->Pq = 64;
+ filterpars->Pstages = 0;
+ filterpars->Pgain = 64;
+ break;
+ case 2:
+ filterpars->Pcategory = 0;
+ filterpars->Ptype = 4;
+ filterpars->Pfreq = 64;
+ filterpars->Pq = 64;
+ filterpars->Pstages = 2;
+ filterpars->Pgain = 64;
+ break;
+ case 3:
+ filterpars->Pcategory = 1;
+ filterpars->Ptype = 0;
+ filterpars->Pfreq = 50;
+ filterpars->Pq = 70;
+ filterpars->Pstages = 1;
+ filterpars->Pgain = 64;
+
+ filterpars->Psequencesize = 2;
+ // "I"
+ filterpars->Pvowels[0].formants[0].freq = 34;
+ filterpars->Pvowels[0].formants[0].amp = 127;
+ filterpars->Pvowels[0].formants[0].q = 64;
+ filterpars->Pvowels[0].formants[1].freq = 99;
+ filterpars->Pvowels[0].formants[1].amp = 122;
+ filterpars->Pvowels[0].formants[1].q = 64;
+ filterpars->Pvowels[0].formants[2].freq = 108;
+ filterpars->Pvowels[0].formants[2].amp = 112;
+ filterpars->Pvowels[0].formants[2].q = 64;
+ // "A"
+ filterpars->Pvowels[1].formants[0].freq = 61;
+ filterpars->Pvowels[1].formants[0].amp = 127;
+ filterpars->Pvowels[1].formants[0].q = 64;
+ filterpars->Pvowels[1].formants[1].freq = 71;
+ filterpars->Pvowels[1].formants[1].amp = 121;
+ filterpars->Pvowels[1].formants[1].q = 64;
+ filterpars->Pvowels[1].formants[2].freq = 99;
+ filterpars->Pvowels[1].formants[2].amp = 117;
+ filterpars->Pvowels[1].formants[2].q = 64;
+ break;
+ case 4:
+ filterpars->Pcategory = 1;
+ filterpars->Ptype = 0;
+ filterpars->Pfreq = 64;
+ filterpars->Pq = 70;
+ filterpars->Pstages = 1;
+ filterpars->Pgain = 64;
+
+ filterpars->Psequencesize = 2;
+ filterpars->Pnumformants = 2;
+ filterpars->Pvowelclearness = 0;
+
+ filterpars->Pvowels[0].formants[0].freq = 70;
+ filterpars->Pvowels[0].formants[0].amp = 127;
+ filterpars->Pvowels[0].formants[0].q = 64;
+ filterpars->Pvowels[0].formants[1].freq = 80;
+ filterpars->Pvowels[0].formants[1].amp = 122;
+ filterpars->Pvowels[0].formants[1].q = 64;
+
+ filterpars->Pvowels[1].formants[0].freq = 20;
+ filterpars->Pvowels[1].formants[0].amp = 127;
+ filterpars->Pvowels[1].formants[0].q = 64;
+ filterpars->Pvowels[1].formants[1].freq = 100;
+ filterpars->Pvowels[1].formants[1].amp = 121;
+ filterpars->Pvowels[1].formants[1].q = 64;
+ break;
}
// for (int i=0;i<5;i++){
@@ -320,4 +320,3 @@ unsigned char DynamicFilter::getpar(int npar) const
default: return 0;
}
}
-
diff --git a/src/Effects/DynamicFilter.h b/src/Effects/DynamicFilter.h
@@ -58,10 +58,9 @@ class DynamicFilter:public Effect
//Internal Values
float depth, ampsns, ampsmooth;
- class Filter *filterl, *filterr;
+ class Filter * filterl, *filterr;
float ms1, ms2, ms3, ms4; //mean squares
};
#endif
-
diff --git a/src/Effects/EQ.cpp b/src/Effects/EQ.cpp
@@ -78,7 +78,7 @@ void EQ::setvolume(unsigned char Pvolume)
{
this->Pvolume = Pvolume;
- outvolume = powf(0.005f, (1.0f - Pvolume / 127.0f)) * 10.0f;
+ outvolume = powf(0.005f, (1.0f - Pvolume / 127.0f)) * 10.0f;
if(insertion == 0)
volume = 1.0f;
else
@@ -109,9 +109,9 @@ void EQ::setpreset(unsigned char npreset)
void EQ::changepar(int npar, unsigned char value)
{
switch(npar) {
- case 0:
- setvolume(value);
- break;
+ case 0:
+ setvolume(value);
+ break;
}
if(npar < 10)
return;
@@ -123,49 +123,49 @@ void EQ::changepar(int npar, unsigned char value)
float tmp;
switch(bp) {
- case 0:
- filter[nb].Ptype = value;
- if(value > 9)
- filter[nb].Ptype = 0; //has to be changed if more filters will be added
- if(filter[nb].Ptype != 0) {
- filter[nb].l->settype(value - 1);
- filter[nb].r->settype(value - 1);
- }
- break;
- case 1:
- filter[nb].Pfreq = value;
- tmp = 600.0f * powf(30.0f, (value - 64.0f) / 64.0f);
- filter[nb].l->setfreq(tmp);
- filter[nb].r->setfreq(tmp);
- break;
- case 2:
- filter[nb].Pgain = value;
- tmp = 30.0f * (value - 64.0f) / 64.0f;
- filter[nb].l->setgain(tmp);
- filter[nb].r->setgain(tmp);
- break;
- case 3:
- filter[nb].Pq = value;
- tmp = powf(30.0f, (value - 64.0f) / 64.0f);
- filter[nb].l->setq(tmp);
- filter[nb].r->setq(tmp);
- break;
- case 4:
- filter[nb].Pstages = value;
- if(value >= MAX_FILTER_STAGES)
- filter[nb].Pstages = MAX_FILTER_STAGES - 1;
- filter[nb].l->setstages(value);
- filter[nb].r->setstages(value);
- break;
+ case 0:
+ filter[nb].Ptype = value;
+ if(value > 9)
+ filter[nb].Ptype = 0; //has to be changed if more filters will be added
+ if(filter[nb].Ptype != 0) {
+ filter[nb].l->settype(value - 1);
+ filter[nb].r->settype(value - 1);
+ }
+ break;
+ case 1:
+ filter[nb].Pfreq = value;
+ tmp = 600.0f * powf(30.0f, (value - 64.0f) / 64.0f);
+ filter[nb].l->setfreq(tmp);
+ filter[nb].r->setfreq(tmp);
+ break;
+ case 2:
+ filter[nb].Pgain = value;
+ tmp = 30.0f * (value - 64.0f) / 64.0f;
+ filter[nb].l->setgain(tmp);
+ filter[nb].r->setgain(tmp);
+ break;
+ case 3:
+ filter[nb].Pq = value;
+ tmp = powf(30.0f, (value - 64.0f) / 64.0f);
+ filter[nb].l->setq(tmp);
+ filter[nb].r->setq(tmp);
+ break;
+ case 4:
+ filter[nb].Pstages = value;
+ if(value >= MAX_FILTER_STAGES)
+ filter[nb].Pstages = MAX_FILTER_STAGES - 1;
+ filter[nb].l->setstages(value);
+ filter[nb].r->setstages(value);
+ break;
}
}
unsigned char EQ::getpar(int npar) const
{
switch(npar) {
- case 0:
- return Pvolume;
- break;
+ case 0:
+ return Pvolume;
+ break;
}
if(npar < 10)
@@ -176,21 +176,21 @@ unsigned char EQ::getpar(int npar) const
return 0;
int bp = npar % 5; //band paramenter
switch(bp) {
- case 0:
- return filter[nb].Ptype;
- break;
- case 1:
- return filter[nb].Pfreq;
- break;
- case 2:
- return filter[nb].Pgain;
- break;
- case 3:
- return filter[nb].Pq;
- break;
- case 4:
- return filter[nb].Pstages;
- break;
+ case 0:
+ return filter[nb].Ptype;
+ break;
+ case 1:
+ return filter[nb].Pfreq;
+ break;
+ case 2:
+ return filter[nb].Pgain;
+ break;
+ case 3:
+ return filter[nb].Pq;
+ break;
+ case 4:
+ return filter[nb].Pstages;
+ break;
}
return 0; //in case of bogus parameter number
@@ -210,4 +210,3 @@ float EQ::getfreqresponse(float freq)
}
return rap2dB(resp * outvolume);
}
-
diff --git a/src/Effects/EQ.h b/src/Effects/EQ.h
@@ -54,4 +54,3 @@ class EQ:public Effect
#endif
-
diff --git a/src/Effects/Echo.cpp b/src/Effects/Echo.cpp
@@ -53,8 +53,8 @@ Echo::~Echo()
*/
void Echo::cleanup()
{
- memset(delay.l,0,MAX_DELAY*SAMPLE_RATE*sizeof(float));
- memset(delay.r,0,MAX_DELAY*SAMPLE_RATE*sizeof(float));
+ memset(delay.l, 0, MAX_DELAY * SAMPLE_RATE * sizeof(float));
+ memset(delay.r, 0, MAX_DELAY * SAMPLE_RATE * sizeof(float));
old = Stereo<float>(0.0f);
}
@@ -75,8 +75,8 @@ void Echo::initdelays()
//number of seconds to delay right chan
float dr = avgDelay + lrdelay;
- ndelta.l = max(1,(int) (dl * SAMPLE_RATE));
- ndelta.r = max(1,(int) (dr * SAMPLE_RATE));
+ ndelta.l = max(1, (int) (dl * SAMPLE_RATE));
+ ndelta.r = max(1, (int) (dr * SAMPLE_RATE));
}
void Echo::out(const Stereo<float *> &input)
@@ -96,20 +96,28 @@ void Echo::out(const Stereo<float *> &input)
rdl = input.r[i] * pangainR - rdl * fb;
//LowPass Filter
- old.l = delay.l[(pos.l+delta.l)%(MAX_DELAY * SAMPLE_RATE)] = ldl * hidamp + old.l * (1.0f - hidamp);
- old.r = delay.r[(pos.r+delta.r)%(MAX_DELAY * SAMPLE_RATE)] = rdl * hidamp + old.r * (1.0f - hidamp);
+ old.l =
+ delay.l[(pos.l
+ + delta.l)
+ % (MAX_DELAY
+ * SAMPLE_RATE)] = ldl * hidamp + old.l * (1.0f - hidamp);
+ old.r =
+ delay.r[(pos.r
+ + delta.r)
+ % (MAX_DELAY
+ * SAMPLE_RATE)] = rdl * hidamp + old.r * (1.0f - hidamp);
//increment
- ++pos.l;// += delta.l;
- ++pos.r;// += delta.r;
+ ++pos.l; // += delta.l;
+ ++pos.r; // += delta.r;
//ensure that pos is still in bounds
pos.l %= MAX_DELAY * SAMPLE_RATE;
pos.r %= MAX_DELAY * SAMPLE_RATE;
//adjust delay if needed
- delta.l = (15*delta.l + ndelta.l)/16;
- delta.r = (15*delta.r + ndelta.r)/16;
+ delta.l = (15 * delta.l + ndelta.l) / 16;
+ delta.r = (15 * delta.r + ndelta.r) / 16;
}
}
@@ -133,8 +141,8 @@ void Echo::setvolume(unsigned char Pvolume)
void Echo::setdelay(unsigned char Pdelay)
{
- this->Pdelay=Pdelay;
- avgDelay=(Pdelay/127.0f*1.5f);//0 .. 1.5f sec
+ this->Pdelay = Pdelay;
+ avgDelay = (Pdelay / 127.0f * 1.5f); //0 .. 1.5f sec
initdelays();
}
@@ -169,23 +177,23 @@ void Echo::setpreset(unsigned char npreset)
const int NUM_PRESETS = 9;
unsigned char presets[NUM_PRESETS][PRESET_SIZE] = {
//Echo 1
- {67, 64, 35, 64, 30, 59, 0 },
+ {67, 64, 35, 64, 30, 59, 0 },
//Echo 2
- {67, 64, 21, 64, 30, 59, 0 },
+ {67, 64, 21, 64, 30, 59, 0 },
//Echo 3
- {67, 75, 60, 64, 30, 59, 10 },
+ {67, 75, 60, 64, 30, 59, 10 },
//Simple Echo
- {67, 60, 44, 64, 30, 0, 0 },
+ {67, 60, 44, 64, 30, 0, 0 },
//Canyon
- {67, 60, 102, 50, 30, 82, 48 },
+ {67, 60, 102, 50, 30, 82, 48 },
//Panning Echo 1
- {67, 64, 44, 17, 0, 82, 24 },
+ {67, 64, 44, 17, 0, 82, 24 },
//Panning Echo 2
- {81, 60, 46, 118, 100, 68, 18 },
+ {81, 60, 46, 118, 100, 68, 18 },
//Panning Echo 3
- {81, 60, 26, 100, 127, 67, 36 },
+ {81, 60, 26, 100, 127, 67, 36 },
//Feedback Echo
- {62, 64, 28, 64, 100, 90, 55 }
+ {62, 64, 28, 64, 100, 90, 55 }
};
diff --git a/src/Effects/Echo.h b/src/Effects/Echo.h
@@ -99,11 +99,11 @@ class Echo:public Effect
void setdryonly();
private:
//Parameters
- char Pvolume; /**<#1 Volume or Dry/Wetness*/
- char Pdelay; /**<#3 Delay of the Echo*/
- char Plrdelay; /**<#4 L/R delay difference*/
- char Pfb; /**<#6Feedback*/
- char Phidamp; /**<#7Dampening of the Echo*/
+ char Pvolume; /**<#1 Volume or Dry/Wetness*/
+ char Pdelay; /**<#3 Delay of the Echo*/
+ char Plrdelay; /**<#4 L/R delay difference*/
+ char Pfb; /**<#6Feedback*/
+ char Phidamp; /**<#7Dampening of the Echo*/
void setvolume(unsigned char Pvolume);
void setdelay(unsigned char Pdelay);
@@ -115,13 +115,13 @@ class Echo:public Effect
float fb, hidamp;
//Left/Right delay lengths
Stereo<int> delayTime;
- float lrdelay;
- float avgDelay;
+ float lrdelay;
+ float avgDelay;
void initdelays();
//2 channel ring buffer
Stereo<float *> delay;
- Stereo<float> old;
+ Stereo<float> old;
//position of reading/writing from delaysample
Stereo<int> pos;
@@ -131,4 +131,3 @@ class Echo:public Effect
};
#endif
-
diff --git a/src/Effects/Effect.cpp b/src/Effects/Effect.cpp
@@ -29,12 +29,12 @@ Effect::Effect(bool insertion_, float *const efxoutl_,
float *const efxoutr_, FilterParams *filterpars_,
const unsigned char &Ppreset_)
:Ppreset(Ppreset_), efxoutl(efxoutl_), efxoutr(efxoutr_),
- filterpars(filterpars_), insertion(insertion_)
+ filterpars(filterpars_), insertion(insertion_)
{}
void Effect::out(float *const smpsl, float *const smpsr)
{
- out(Stereo<float *>(smpsl,smpsr));
+ out(Stereo<float *>(smpsl, smpsr));
}
void Effect::crossover(float &a, float &b, float crossover)
@@ -56,5 +56,5 @@ void Effect::setpanning(char Ppanning_)
void Effect::setlrcross(char Plrcross_)
{
Plrcross = Plrcross_;
- lrcross = (float)Plrcross / 127.0f;
+ lrcross = (float)Plrcross / 127.0f;
}
diff --git a/src/Effects/Effect.h b/src/Effects/Effect.h
@@ -82,9 +82,9 @@ class Effect
return freq;
}
- unsigned char Ppreset; /**<Currently used preset*/
- float *const efxoutl; /**<Effect out Left Channel*/
- float *const efxoutr; /**<Effect out Right Channel*/
+ unsigned char Ppreset; /**<Currently used preset*/
+ float *const efxoutl; /**<Effect out Left Channel*/
+ float *const efxoutr; /**<Effect out Right Channel*/
/**\todo make efxoutl and efxoutr private and replace them with a Stereo<float*>*/
float outvolume;/**<This is the volume of effect and is public because
@@ -107,7 +107,7 @@ class Effect
const bool insertion;/**<If Effect is an insertion effect, insertion=1
*otherwise, it should be insertion=0*/
//panning parameters
- char Ppanning;
+ char Ppanning;
float pangainL;
float pangainR;
char Plrcross; // L/R mix
@@ -115,4 +115,3 @@ class Effect
};
#endif
-
diff --git a/src/Effects/EffectLFO.cpp b/src/Effects/EffectLFO.cpp
@@ -28,9 +28,9 @@
EffectLFO::EffectLFO()
{
- xl = 0.0f;
- xr = 0.0f;
- Pfreq = 40;
+ xl = 0.0f;
+ xr = 0.0f;
+ Pfreq = 40;
Prandomness = 0;
PLFOtype = 0;
Pstereo = 96;
@@ -68,7 +68,7 @@ void EffectLFO::updateparams()
PLFOtype = 1; //this has to be updated if more lfo's are added
lfotype = PLFOtype;
- xr = fmod(xl + (Pstereo - 64.0f) / 127.0f + 1.0f, 1.0f);
+ xr = fmod(xl + (Pstereo - 64.0f) / 127.0f + 1.0f, 1.0f);
}
@@ -79,18 +79,18 @@ float EffectLFO::getlfoshape(float x)
{
float out;
switch(lfotype) {
- case 1: //EffectLFO_TRIANGLE
- if((x > 0.0f) && (x < 0.25f))
- out = 4.0f * x;
- else
- if((x > 0.25f) && (x < 0.75f))
- out = 2 - 4 * x;
- else
- out = 4.0f * x - 4.0f;
- break;
- /**\todo more to be added here; also ::updateparams() need to be updated (to allow more lfotypes)*/
- default:
- out = cosf(x * 2 * PI); //EffectLFO_SINE
+ case 1: //EffectLFO_TRIANGLE
+ if((x > 0.0f) && (x < 0.25f))
+ out = 4.0f * x;
+ else
+ if((x > 0.25f) && (x < 0.75f))
+ out = 2 - 4 * x;
+ else
+ out = 4.0f * x - 4.0f;
+ break;
+ /**\todo more to be added here; also ::updateparams() need to be updated (to allow more lfotypes)*/
+ default:
+ out = cosf(x * 2 * PI); //EffectLFO_SINE
}
return out;
}
@@ -113,10 +113,10 @@ void EffectLFO::effectlfoout(float *outl, float *outr)
}
*outl = (out + 1.0f) * 0.5f;
- out = getlfoshape(xr);
+ out = getlfoshape(xr);
if((lfotype == 0) || (lfotype == 1))
out *= (ampr1 + xr * (ampr2 - ampr1));
- xr += incx;
+ xr += incx;
if(xr > 1.0f) {
xr -= 1.0f;
ampr1 = ampr2;
@@ -124,4 +124,3 @@ void EffectLFO::effectlfoout(float *outl, float *outr)
}
*outr = (out + 1.0f) * 0.5f;
}
-
diff --git a/src/Effects/EffectLFO.h b/src/Effects/EffectLFO.h
@@ -45,9 +45,8 @@ class EffectLFO
float ampl1, ampl2, ampr1, ampr2; //necessary for "randomness"
float lfointensity;
float lfornd;
- char lfotype; /**\todo GET RID OF CHAR (replace with short or enum)*/
+ char lfotype; /**\todo GET RID OF CHAR (replace with short or enum)*/
};
#endif
-
diff --git a/src/Effects/EffectMgr.cpp b/src/Effects/EffectMgr.cpp
@@ -89,34 +89,34 @@ void EffectMgr::changeeffect(int nefx_)
if(efx != NULL)
delete efx;
switch(nefx) { /**\todo replace leaky abstraction*/
- case 1:
- efx = new Reverb(insertion, efxoutl, efxoutr);
- break;
- case 2:
- efx = new Echo(insertion, efxoutl, efxoutr);
- break;
- case 3:
- efx = new Chorus(insertion, efxoutl, efxoutr);
- break;
- case 4:
- efx = new Phaser(insertion, efxoutl, efxoutr);
- break;
- case 5:
- efx = new Alienwah(insertion, efxoutl, efxoutr);
- break;
- case 6:
- efx = new Distorsion(insertion, efxoutl, efxoutr);
- break;
- case 7:
- efx = new EQ(insertion, efxoutl, efxoutr);
- break;
- case 8:
- efx = new DynamicFilter(insertion, efxoutl, efxoutr);
- break;
- //put more effect here
- default:
- efx = NULL;
- break; //no effect (thru)
+ case 1:
+ efx = new Reverb(insertion, efxoutl, efxoutr);
+ break;
+ case 2:
+ efx = new Echo(insertion, efxoutl, efxoutr);
+ break;
+ case 3:
+ efx = new Chorus(insertion, efxoutl, efxoutr);
+ break;
+ case 4:
+ efx = new Phaser(insertion, efxoutl, efxoutr);
+ break;
+ case 5:
+ efx = new Alienwah(insertion, efxoutl, efxoutr);
+ break;
+ case 6:
+ efx = new Distorsion(insertion, efxoutl, efxoutr);
+ break;
+ case 7:
+ efx = new EQ(insertion, efxoutl, efxoutr);
+ break;
+ case 8:
+ efx = new DynamicFilter(insertion, efxoutl, efxoutr);
+ break;
+ //put more effect here
+ default:
+ efx = NULL;
+ break; //no effect (thru)
}
if(efx != NULL)
@@ -254,29 +254,26 @@ void EffectMgr::out(float *smpsl, float *smpsr)
if((nefx == 1) || (nefx == 2))
v2 *= v2; //for Reverb and Echo, the wet function is not liniar
- if(dryonly) { //this is used for instrument effect only
+ if(dryonly) //this is used for instrument effect only
for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
smpsl[i] *= v1;
smpsr[i] *= v1;
efxoutl[i] *= v2;
efxoutr[i] *= v2;
}
- }
- else { //normal instrument/insertion effect
+ else //normal instrument/insertion effect
for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
smpsl[i] = smpsl[i] * v1 + efxoutl[i] * v2;
smpsr[i] = smpsr[i] * v1 + efxoutr[i] * v2;
}
- }
}
- else { //System effect
+ else //System effect
for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] *= 2.0f * volume;
efxoutr[i] *= 2.0f * volume;
smpsl[i] = efxoutl[i];
smpsr[i] = efxoutr[i];
}
- }
}
/*
@@ -354,14 +351,12 @@ void EffectMgr::getfromXML(XMLwrapper *xml)
seteffectpar_nolock(n, xml->getpar127("par", par));
xml->exitbranch();
}
- if(filterpars != NULL) {
+ if(filterpars != NULL)
if(xml->enterbranch("FILTER")) {
filterpars->getfromXML(xml);
xml->exitbranch();
}
- }
xml->exitbranch();
}
cleanup();
}
-
diff --git a/src/Effects/EffectMgr.h b/src/Effects/EffectMgr.h
@@ -85,7 +85,7 @@ class EffectMgr:public Presets
void seteffectpar_nolock(int npar, unsigned char value);
unsigned char geteffectpar(int npar);
const bool insertion; /**<1 if the effect is connected as insertion effect*/
- float *efxoutl, *efxoutr;
+ float *efxoutl, *efxoutr;
/**used by UI
* \todo needs to be decoupled*/
@@ -101,4 +101,3 @@ class EffectMgr:public Presets
};
#endif
-
diff --git a/src/Effects/Phaser.cpp b/src/Effects/Phaser.cpp
@@ -41,7 +41,7 @@ using namespace std;
Phaser::Phaser(const int &insertion_, float *efxoutl_, float *efxoutr_)
:Effect(insertion_, efxoutl_, efxoutr_, NULL, 0), old(NULL), xn1(NULL),
- yn1(NULL), diff(0.0f), oldgain(0.0f), fb(0.0f)
+ yn1(NULL), diff(0.0f), oldgain(0.0f), fb(0.0f)
{
analog_setup();
setpreset(Ppreset);
@@ -51,28 +51,28 @@ Phaser::Phaser(const int &insertion_, float *efxoutl_, float *efxoutr_)
void Phaser::analog_setup()
{
//model mismatch between JFET devices
- offset[0] = -0.2509303f;
- offset[1] = 0.9408924f;
- offset[2] = 0.998f;
- offset[3] = -0.3486182f;
- offset[4] = -0.2762545f;
- offset[5] = -0.5215785f;
- offset[6] = 0.2509303f;
- offset[7] = -0.9408924f;
- offset[8] = -0.998f;
- offset[9] = 0.3486182f;
+ offset[0] = -0.2509303f;
+ offset[1] = 0.9408924f;
+ offset[2] = 0.998f;
+ offset[3] = -0.3486182f;
+ offset[4] = -0.2762545f;
+ offset[5] = -0.5215785f;
+ offset[6] = 0.2509303f;
+ offset[7] = -0.9408924f;
+ offset[8] = -0.998f;
+ offset[9] = 0.3486182f;
offset[10] = 0.2762545f;
offset[11] = 0.5215785f;
barber = 0; //Deactivate barber pole phasing by default
- mis = 1.0f;
- Rmin = 625.0f;// 2N5457 typical on resistance at Vgs = 0
- Rmax = 22000.0f;// Resistor parallel to FET
- Rmx = Rmin/Rmax;
- Rconst = 1.0f + Rmx; // Handle parallel resistor relationship
- C = 0.00000005f; // 50 nF
- CFs = (float) 2.0f*(float)SAMPLE_RATE*C;
+ mis = 1.0f;
+ Rmin = 625.0f; // 2N5457 typical on resistance at Vgs = 0
+ Rmax = 22000.0f; // Resistor parallel to FET
+ Rmx = Rmin / Rmax;
+ Rconst = 1.0f + Rmx; // Handle parallel resistor relationship
+ C = 0.00000005f; // 50 nF
+ CFs = (float) 2.0f * (float)SAMPLE_RATE * C;
invperiod = 1.0f / ((float) SOUND_BUFFER_SIZE);
}
@@ -101,11 +101,12 @@ void Phaser::out(const Stereo<float *> &input)
void Phaser::AnalogPhase(const Stereo<float *> &input)
{
- Stereo<float> gain(0.0f), lfoVal(0.0f), mod(0.0f), g(0.0f), b(0.0f), hpf(0.0f);
+ Stereo<float> gain(0.0f), lfoVal(0.0f), mod(0.0f), g(0.0f), b(0.0f), hpf(
+ 0.0f);
lfo.effectlfoout(&lfoVal.l, &lfoVal.r);
- mod.l = lfoVal.l*width + (depth - 0.5f);
- mod.r = lfoVal.r*width + (depth - 0.5f);
+ mod.l = lfoVal.l * width + (depth - 0.5f);
+ mod.r = lfoVal.r * width + (depth - 0.5f);
mod.l = limit(mod.l, ZERO_, ONE_);
mod.r = limit(mod.r, ZERO_, ONE_);
@@ -128,13 +129,13 @@ void Phaser::AnalogPhase(const Stereo<float *> &input)
g = oldgain;
oldgain = mod;
- for (int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- g.l += diff.l;// Linear interpolation between LFO samples
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
+ g.l += diff.l; // Linear interpolation between LFO samples
g.r += diff.r;
Stereo<float> xn(input.l[i] * pangainL, input.r[i] * pangainR);
- if (barber) {
+ if(barber) {
g.l = fmodf((g.l + 0.25f), ONE_);
g.r = fmodf((g.r + 0.25f), ONE_);
}
@@ -156,29 +157,29 @@ void Phaser::AnalogPhase(const Stereo<float *> &input)
}
float Phaser::applyPhase(float x, float g, float fb,
- float &hpf, float *yn1, float *xn1)
+ float &hpf, float *yn1, float *xn1)
{
for(int j = 0; j < Pstages; ++j) { //Phasing routine
- mis = 1.0f + offsetpct*offset[j];
+ mis = 1.0f + offsetpct * offset[j];
//This is symmetrical.
//FET is not, so this deviates slightly, however sym dist. is
//better sounding than a real FET.
- float d = (1.0f + 2.0f*(0.25f + g)*hpf*hpf*distortion) * mis;
- Rconst = 1.0f + mis*Rmx;
+ float d = (1.0f + 2.0f * (0.25f + g) * hpf * hpf * distortion) * mis;
+ Rconst = 1.0f + mis * Rmx;
// This is 1/R. R is being modulated to control filter fc.
- float b = (Rconst - g)/ (d*Rmin);
- float gain = (CFs - b)/(CFs + b);
+ float b = (Rconst - g) / (d * Rmin);
+ float gain = (CFs - b) / (CFs + b);
yn1[j] = gain * (x + yn1[j]) - xn1[j];
//high pass filter:
//Distortion depends on the high-pass part of the AP stage.
- hpf = yn1[j] + (1.0f-gain)*xn1[j];
+ hpf = yn1[j] + (1.0f - gain) * xn1[j];
xn1[j] = x;
- x = yn1[j];
- if (j==1)
+ x = yn1[j];
+ if(j == 1)
x += fb; //Insert feedback after first phase stage
}
return x;
@@ -188,8 +189,12 @@ void Phaser::normalPhase(const Stereo<float *> &input)
Stereo<float> gain(0.0f), lfoVal(0.0f);
lfo.effectlfoout(&lfoVal.l, &lfoVal.r);
- gain.l = (expf(lfoVal.l * PHASER_LFO_SHAPE) - 1) / (expf(PHASER_LFO_SHAPE) - 1.0f);
- gain.r = (expf(lfoVal.r * PHASER_LFO_SHAPE) - 1) / (expf(PHASER_LFO_SHAPE) - 1.0f);
+ gain.l =
+ (expf(lfoVal.l
+ * PHASER_LFO_SHAPE) - 1) / (expf(PHASER_LFO_SHAPE) - 1.0f);
+ gain.r =
+ (expf(lfoVal.r
+ * PHASER_LFO_SHAPE) - 1) / (expf(PHASER_LFO_SHAPE) - 1.0f);
gain.l = 1.0f - phase * (1.0f - depth) - (1.0f - phase) * gain.l * depth;
gain.r = 1.0f - phase * (1.0f - depth) - (1.0f - phase) * gain.r * depth;
@@ -198,14 +203,14 @@ void Phaser::normalPhase(const Stereo<float *> &input)
gain.r = limit(gain.r, ZERO_, ONE_);
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- float x = (float) i / SOUND_BUFFER_SIZE;
- float x1 = 1.0f - x;
+ float x = (float) i / SOUND_BUFFER_SIZE;
+ float x1 = 1.0f - x;
//TODO think about making panning an external feature
Stereo<float> xn(input.l[i] * pangainL + fb.l,
- input.r[i] * pangainR + fb.r);
+ input.r[i] * pangainR + fb.r);
Stereo<float> g(gain.l * x + oldgain.l * x1,
- gain.r * x + oldgain.r * x1);
+ gain.r * x + oldgain.r * x1);
xn.l = applyPhase(xn.l, g.l, old.l);
xn.r = applyPhase(xn.r, g.r, old.r);
@@ -232,7 +237,7 @@ float Phaser::applyPhase(float x, float g, float *old)
for(int j = 0; j < Pstages * 2; ++j) { //Phasing routine
float tmp = old[j];
old[j] = g * tmp + x;
- x = tmp - g *old[j];
+ x = tmp - g * old[j];
}
return x;
}
@@ -267,7 +272,7 @@ void Phaser::setwidth(unsigned char Pwidth)
void Phaser::setfb(unsigned char Pfb)
{
this->Pfb = Pfb;
- feedback = (float) (Pfb - 64) / 64.2f;
+ feedback = (float) (Pfb - 64) / 64.2f;
}
void Phaser::setvolume(unsigned char Pvolume)
@@ -289,7 +294,7 @@ void Phaser::setdistortion(unsigned char Pdistortion)
void Phaser::setoffset(unsigned char Poffset)
{
this->Poffset = Poffset;
- offsetpct = (float)Poffset / 127.0f;
+ offsetpct = (float)Poffset / 127.0f;
}
void Phaser::setstages(unsigned char Pstages)
@@ -307,13 +312,13 @@ void Phaser::setstages(unsigned char Pstages)
this->Pstages = min(MAX_PHASER_STAGES, (int)Pstages);
old = Stereo<float *>(new float[Pstages * 2],
- new float[Pstages * 2]);
+ new float[Pstages * 2]);
xn1 = Stereo<float *>(new float[Pstages],
- new float[Pstages]);
+ new float[Pstages]);
yn1 = Stereo<float *>(new float[Pstages],
- new float[Pstages]);
+ new float[Pstages]);
cleanup();
}
@@ -333,25 +338,38 @@ void Phaser::setdepth(unsigned char Pdepth)
void Phaser::setpreset(unsigned char npreset)
{
- const int PRESET_SIZE = 15;
- const int NUM_PRESETS = 12;
+ const int PRESET_SIZE = 15;
+ const int NUM_PRESETS = 12;
unsigned char presets[NUM_PRESETS][PRESET_SIZE] = {
//Phaser
//0 1 2 3 4 5 6 7 8 9 10 11 12 13 14
- {64, 64, 36, 0, 0, 64, 110, 64, 1, 0, 0, 20, 0, 0, 0},
- {64, 64, 35, 0, 0, 88, 40, 64, 3, 0, 0, 20, 0, 0, 0},
- {64, 64, 31, 0, 0, 66, 68, 107, 2, 0, 0, 20, 0, 0, 0},
- {39, 64, 22, 0, 0, 66, 67, 10, 5, 0, 1, 20, 0, 0, 0},
- {64, 64, 20, 0, 1, 110, 67, 78, 10, 0, 0, 20, 0, 0, 0},
- {64, 64, 53, 100, 0, 58, 37, 78, 3, 0, 0, 20, 0, 0, 0},
+ {64, 64, 36, 0, 0, 64, 110, 64, 1, 0, 0, 20,
+ 0, 0,
+ 0 },
+ {64, 64, 35, 0, 0, 88, 40, 64, 3, 0, 0, 20, 0, 0,
+ 0 },
+ {64, 64, 31, 0, 0, 66, 68, 107, 2, 0, 0, 20, 0, 0,
+ 0 },
+ {39, 64, 22, 0, 0, 66, 67, 10, 5, 0, 1, 20, 0, 0,
+ 0 },
+ {64, 64, 20, 0, 1, 110, 67, 78, 10, 0, 0, 20, 0, 0,
+ 0 },
+ {64, 64, 53, 100, 0, 58, 37, 78, 3, 0, 0, 20, 0, 0,
+ 0 },
//APhaser
//0 1 2 3 4 5 6 7 8 9 10 11 12 13 14
- {64, 64, 14, 0, 1, 64, 64, 40, 4, 10, 0, 110, 1, 20, 1},
- {64, 64, 14, 5, 1, 64, 70, 40, 6, 10, 0, 110, 1, 20, 1},
- {64, 64, 9, 0, 0, 64, 60, 40, 8, 10, 0, 40, 0, 20, 1},
- {64, 64, 14, 10, 0, 64, 45, 80, 7, 10, 1, 110, 1, 20, 1},
- {25, 64, 127, 10, 0, 64, 25, 16, 8, 100, 0, 25, 0, 20, 1},
- {64, 64, 1, 10, 1, 64, 70, 40, 12, 10, 0, 110, 1, 20, 1}
+ {64, 64, 14, 0, 1, 64, 64, 40, 4, 10, 0, 110,1, 20,
+ 1 },
+ {64, 64, 14, 5, 1, 64, 70, 40, 6, 10, 0, 110,1, 20,
+ 1 },
+ {64, 64, 9, 0, 0, 64, 60, 40, 8, 10, 0, 40, 0, 20,
+ 1 },
+ {64, 64, 14, 10, 0, 64, 45, 80, 7, 10, 1, 110,1, 20,
+ 1 },
+ {25, 64, 127, 10, 0, 64, 25, 16, 8, 100, 0, 25, 0, 20,
+ 1 },
+ {64, 64, 1, 10, 1, 64, 70, 40, 12, 10, 0, 110,1, 20,
+ 1 }
};
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
@@ -401,7 +419,7 @@ void Phaser::changepar(int npar, unsigned char value)
setoffset(value);
break;
case 10:
- Poutsub = min((int)value,1);
+ Poutsub = min((int)value, 1);
break;
case 11:
setphase(value);
@@ -432,10 +450,10 @@ unsigned char Phaser::getpar(int npar) const
case 7: return Pfb;
case 8: return Pstages;
case 9: return Plrcross;
- return Poffset; //same
+ return Poffset; //same
case 10: return Poutsub;
case 11: return Pphase;
- return Pwidth; //same
+ return Pwidth; //same
case 12: return Phyper;
case 13: return Pdistortion;
case 14: return Panalog;
diff --git a/src/Effects/Phaser.h b/src/Effects/Phaser.h
@@ -45,7 +45,7 @@ class Phaser:public Effect
private:
//Phaser parameters
- EffectLFO lfo; //Phaser modulator
+ EffectLFO lfo; //Phaser modulator
unsigned char Pvolume; //Used to set wet/dry mix
unsigned char Pdistortion; //Model distortion added by FET element
unsigned char Pdepth; //Depth of phaser sweep
@@ -69,14 +69,14 @@ class Phaser:public Effect
void setphase(unsigned char Pphase);
//Internal Variables
- bool barber; //Barber pole phasing flag
+ bool barber; //Barber pole phasing flag
float distortion, width, offsetpct;
float feedback, depth, phase;
Stereo<float *> old, xn1, yn1;
- Stereo<float> diff, oldgain, fb;
+ Stereo<float> diff, oldgain, fb;
float invperiod;
float offset[12];
-
+
float mis;
float Rmin; // 3N5457 typical on resistance at Vgs = 0
float Rmax; // Resistor parallel to FET
@@ -89,11 +89,10 @@ class Phaser:public Effect
void AnalogPhase(const Stereo<float *> &input);
//analog case
float applyPhase(float x, float g, float fb,
- float &hpf, float *yn1, float *xn1);
+ float &hpf, float *yn1, float *xn1);
void normalPhase(const Stereo<float *> &input);
float applyPhase(float x, float g, float *old);
};
#endif
-
diff --git a/src/Effects/Reverb.cpp b/src/Effects/Reverb.cpp
@@ -125,12 +125,12 @@ void Reverb::processmono(int ch, float *output, float *inputbuf)
for(int j = REV_COMBS * ch; j < REV_COMBS * (ch + 1); ++j) {
int &ck = combk[j];
const int comblength = comblen[j];
- float &lpcombj = lpcomb[j];
+ float &lpcombj = lpcomb[j];
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float fbout = comb[j][ck] * combfb[j];
- fbout = fbout * (1.0f - lohifb) + lpcombj * lohifb;
- lpcombj = fbout;
+ fbout = fbout * (1.0f - lohifb) + lpcombj * lohifb;
+ lpcombj = fbout;
comb[j][ck] = inputbuf[i] + fbout;
output[i] += fbout;
@@ -165,7 +165,7 @@ void Reverb::out(const Stereo<float *> &smp)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
inputbuf[i] = (smp.l[i] + smp.r[i]) / 2.0f;
- if(idelay != NULL) {
+ if(idelay != NULL)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
//Initial delay r
float tmp = inputbuf[i] + idelay[idelayk] * idelayfb;
@@ -175,7 +175,6 @@ void Reverb::out(const Stereo<float *> &smp)
if(idelayk >= idelaylen)
idelayk = 0;
}
- }
if(bandwidth)
bandwidth->process(SOUND_BUFFER_SIZE, inputbuf);
@@ -221,7 +220,7 @@ void Reverb::setvolume(unsigned char Pvolume)
void Reverb::settime(unsigned char Ptime)
{
- int i;
+ int i;
float t;
this->Ptime = Ptime;
t = powf(60.0f, (float)Ptime / 127.0f) - 0.97f;
@@ -229,7 +228,7 @@ void Reverb::settime(unsigned char Ptime)
for(i = 0; i < REV_COMBS * 2; ++i)
combfb[i] =
-expf((float)comblen[i] / (float)SAMPLE_RATE * logf(0.001f) / t);
- //the feedback is negative because it removes the DC
+ //the feedback is negative because it removes the DC
}
void Reverb::setlohidamp(unsigned char Plohidamp)
@@ -260,7 +259,7 @@ void Reverb::setidelay(unsigned char Pidelay)
if(idelay != NULL)
delete [] idelay;
- idelay = NULL;
+ idelay = NULL;
idelaylen = (int) (SAMPLE_RATE * delay / 1000);
if(idelaylen > 1) {
@@ -353,7 +352,7 @@ void Reverb::settype(unsigned char Ptype)
lpcomb[i] = 0;
if(comb[i] != NULL)
delete [] comb[i];
- comb[i] = new float[comblen[i]];
+ comb[i] = new float[comblen[i]];
}
for(int i = 0; i < REV_APS * 2; ++i) {
@@ -361,17 +360,17 @@ void Reverb::settype(unsigned char Ptype)
tmp = 500 + (int)(RND * 500);
else
tmp = aptunings[Ptype][i % REV_APS];
- tmp *= roomsize;
+ tmp *= roomsize;
if(i > REV_APS)
tmp += 23.0f;
- tmp *= SAMPLE_RATE / 44100.0f; //adjust the combs according to the samplerate
+ tmp *= SAMPLE_RATE / 44100.0f; //adjust the combs according to the samplerate
if(tmp < 10)
tmp = 10;
aplen[i] = (int) tmp;
apk[i] = 0;
if(ap[i] != NULL)
delete [] ap[i];
- ap[i] = new float[aplen[i]];
+ ap[i] = new float[aplen[i]];
}
settime(Ptime);
cleanup();
@@ -412,31 +411,31 @@ void Reverb::setpreset(unsigned char npreset)
const int NUM_PRESETS = 13;
unsigned char presets[NUM_PRESETS][PRESET_SIZE] = {
//Cathedral1
- {80, 64, 63, 24, 0, 0, 0, 85, 5, 83, 1, 64, 20 },
+ {80, 64, 63, 24, 0, 0, 0, 85, 5, 83, 1, 64, 20 },
//Cathedral2
- {80, 64, 69, 35, 0, 0, 0, 127, 0, 71, 0, 64, 20 },
+ {80, 64, 69, 35, 0, 0, 0, 127, 0, 71, 0, 64, 20 },
//Cathedral3
- {80, 64, 69, 24, 0, 0, 0, 127, 75, 78, 1, 85, 20 },
+ {80, 64, 69, 24, 0, 0, 0, 127, 75, 78, 1, 85, 20 },
//Hall1
- {90, 64, 51, 10, 0, 0, 0, 127, 21, 78, 1, 64, 20 },
+ {90, 64, 51, 10, 0, 0, 0, 127, 21, 78, 1, 64, 20 },
//Hall2
- {90, 64, 53, 20, 0, 0, 0, 127, 75, 71, 1, 64, 20 },
+ {90, 64, 53, 20, 0, 0, 0, 127, 75, 71, 1, 64, 20 },
//Room1
- {100, 64, 33, 0, 0, 0, 0, 127, 0, 106, 0, 30, 20 },
+ {100, 64, 33, 0, 0, 0, 0, 127, 0, 106, 0, 30, 20 },
//Room2
- {100, 64, 21, 26, 0, 0, 0, 62, 0, 77, 1, 45, 20 },
+ {100, 64, 21, 26, 0, 0, 0, 62, 0, 77, 1, 45, 20 },
//Basement
- {110, 64, 14, 0, 0, 0, 0, 127, 5, 71, 0, 25, 20 },
+ {110, 64, 14, 0, 0, 0, 0, 127, 5, 71, 0, 25, 20 },
//Tunnel
- {85, 80, 84, 20, 42, 0, 0, 51, 0, 78, 1, 105, 20 },
+ {85, 80, 84, 20, 42, 0, 0, 51, 0, 78, 1, 105, 20 },
//Echoed1
- {95, 64, 26, 60, 71, 0, 0, 114, 0, 64, 1, 64, 20 },
+ {95, 64, 26, 60, 71, 0, 0, 114, 0, 64, 1, 64, 20 },
//Echoed2
- {90, 64, 40, 88, 71, 0, 0, 114, 0, 88, 1, 64, 20 },
+ {90, 64, 40, 88, 71, 0, 0, 114, 0, 88, 1, 64, 20 },
//VeryLong1
- {90, 64, 93, 15, 0, 0, 0, 114, 0, 77, 0, 95, 20 },
+ {90, 64, 93, 15, 0, 0, 0, 114, 0, 77, 0, 95, 20 },
//VeryLong2
- {90, 64, 111, 30, 0, 0, 0, 114, 90, 74, 1, 80, 20 }
+ {90, 64, 111, 30, 0, 0, 0, 114, 90, 74, 1, 80, 20 }
};
if(npreset >= NUM_PRESETS)
diff --git a/src/Effects/Reverb.h b/src/Effects/Reverb.h
@@ -99,19 +99,19 @@ class Reverb:public Effect
float erbalance;
//Parameters
- int lohidamptype; /**<0=disable,1=highdamp(lowpass),2=lowdamp(highpass)*/
- int idelaylen, rdelaylen;
- int idelayk;
- float lohifb, idelayfb, roomsize, rs; //rs is used to "normalise" the volume according to the roomsize
- int comblen[REV_COMBS * 2];
- int aplen[REV_APS * 2];
- Unison *bandwidth;
+ int lohidamptype; /**<0=disable,1=highdamp(lowpass),2=lowdamp(highpass)*/
+ int idelaylen, rdelaylen;
+ int idelayk;
+ float lohifb, idelayfb, roomsize, rs; //rs is used to "normalise" the volume according to the roomsize
+ int comblen[REV_COMBS * 2];
+ int aplen[REV_APS * 2];
+ Unison *bandwidth;
//Internal Variables
float *comb[REV_COMBS * 2];
- int combk[REV_COMBS * 2];
+ int combk[REV_COMBS * 2];
float combfb[REV_COMBS * 2]; /**<feedback-ul fiecarui filtru "comb"*/
float lpcomb[REV_COMBS * 2]; /**<pentru Filtrul LowPass*/
@@ -119,11 +119,10 @@ class Reverb:public Effect
int apk[REV_APS * 2];
- float *idelay;
- class AnalogFilter *lpf, *hpf; //filters
+ float *idelay;
+ class AnalogFilter * lpf, *hpf; //filters
void processmono(int ch, float *output, float *inputbuf);
};
#endif
-
diff --git a/src/Misc/Atomic.cpp b/src/Misc/Atomic.cpp
@@ -61,7 +61,7 @@ T Atomic<T>::operator++()
}
template<class T>
-T Atomic<T>::operator--(){
+T Atomic<T>::operator--() {
T tmp;
pthread_mutex_lock(&mutex);
tmp = --value;
diff --git a/src/Misc/Atomic.h b/src/Misc/Atomic.h
@@ -43,4 +43,3 @@ class Atomic
};
#include "Atomic.cpp"
#endif
-
diff --git a/src/Misc/Bank.cpp b/src/Misc/Bank.cpp
@@ -88,7 +88,7 @@ void Bank::setname(unsigned int ninstrument, const string &newname, int newslot)
return;
string newfilename;
- char tmpfilename[100 + 1];
+ char tmpfilename[100 + 1];
if(newslot >= 0)
snprintf(tmpfilename, 100, "%4d-%s", newslot + 1, newname.c_str());
@@ -105,7 +105,7 @@ void Bank::setname(unsigned int ninstrument, const string &newname, int newslot)
rename(ins[ninstrument].filename.c_str(), newfilename.c_str());
ins[ninstrument].filename = newfilename;
- ins[ninstrument].name = legalizeFilename(tmpfilename); //TODO limit name to PART_MAX_NAME_LEN
+ ins[ninstrument].name = legalizeFilename(tmpfilename); //TODO limit name to PART_MAX_NAME_LEN
}
/*
@@ -222,12 +222,11 @@ int Bank::loadbank(string bankdirname)
string name = filename;
//remove the file extension
- for(int i = name.size() - 1; i >= 2; i--) {
+ for(int i = name.size() - 1; i >= 2; i--)
if(name[i] == '.') {
- name = name.substr(0,i);
+ name = name.substr(0, i);
break;
}
- }
if(no != 0) //the instrument position in the bank is found
addtobank(no - 1, filename, name.substr(startname));
@@ -298,7 +297,7 @@ void Bank::swapslot(unsigned int n1, unsigned int n2)
setname(n1, getname(n1), n2);
setname(n2, getname(n2), n1);
- swap(ins[n2],ins[n1]);
+ swap(ins[n2], ins[n1]);
}
}
@@ -326,11 +325,12 @@ void Bank::rescanforbanks()
//remove duplicate bank names
int dupl = 0;
- for(int j = 0; j < (int) banks.size() - 1; ++j) {
- for(int i = j + 1; i <(int) banks.size(); ++i) {
+ for(int j = 0; j < (int) banks.size() - 1; ++j)
+ for(int i = j + 1; i < (int) banks.size(); ++i) {
if(banks[i].name == banks[j].name) {
//add a [1] to the first bankname and [n] to others
- banks[i].name = banks[i].name + '[' + stringFrom(dupl +2) + ']';
+ banks[i].name = banks[i].name + '['
+ + stringFrom(dupl + 2) + ']';
if(dupl == 0)
banks[j].name += "[1]";
@@ -339,7 +339,6 @@ void Bank::rescanforbanks()
else
dupl = 0;
}
- }
}
@@ -366,12 +365,12 @@ void Bank::scanrootdir(string rootdir)
if(dirname[0] == '.')
continue;
- bank.dir = rootdir + separator + dirname + '/';
+ bank.dir = rootdir + separator + dirname + '/';
bank.name = dirname;
//find out if the directory contains at least 1 instrument
bool isbank = false;
- DIR *d = opendir(bank.dir.c_str());
+ DIR *d = opendir(bank.dir.c_str());
if(d == NULL)
continue;
@@ -410,26 +409,24 @@ int Bank::addtobank(int pos, string filename, string name)
pos = -1; //force it to find a new free position
}
else
- if(pos >= BANK_SIZE)
- pos = -1;
+ if(pos >= BANK_SIZE)
+ pos = -1;
- if(pos < 0) { //find a free position
- for(int i = BANK_SIZE - 1; i >= 0; i--) {
+ if(pos < 0) //find a free position
+ for(int i = BANK_SIZE - 1; i >= 0; i--)
if(!ins[i].used) {
pos = i;
break;
}
- }
- }
if(pos < 0)
- return -1;//the bank is full
+ return -1; //the bank is full
deletefrombank(pos);
- ins[pos].used = true;
- ins[pos].name = name;
+ ins[pos].used = true;
+ ins[pos].name = name;
ins[pos].filename = dirname + '/' + filename;
//see if PADsynth is used
@@ -466,4 +463,3 @@ Bank::ins_t::ins_t()
{
info.PADsynth_used = false;
}
-
diff --git a/src/Misc/Bank.h b/src/Misc/Bank.h
@@ -42,7 +42,9 @@ class Bank
~Bank();
std::string getname(unsigned int ninstrument);
std::string getnamenumbered(unsigned int ninstrument);
- void setname(unsigned int ninstrument, const std::string &newname, int newslot); //if newslot==-1 then this is ignored, else it will be put on that slot
+ void setname(unsigned int ninstrument,
+ const std::string &newname,
+ int newslot); //if newslot==-1 then this is ignored, else it will be put on that slot
bool isPADsynth_used(unsigned int ninstrument);
/**returns true when slot is empty*/
@@ -89,7 +91,7 @@ class Bank
struct ins_t {
ins_t();
- bool used;
+ bool used;
std::string name;
std::string filename;
struct {
@@ -103,4 +105,3 @@ class Bank
};
#endif
-
diff --git a/src/Misc/Config.cpp b/src/Misc/Config.cpp
@@ -36,29 +36,29 @@ void Config::init()
{
maxstringsize = MAX_STRING_SIZE; //for ui
//defaults
- cfg.SampleRate = 44100;
- cfg.SoundBufferSize = 256;
- cfg.OscilSize = 1024;
- cfg.SwapStereo = 0;
+ cfg.SampleRate = 44100;
+ cfg.SoundBufferSize = 256;
+ cfg.OscilSize = 1024;
+ cfg.SwapStereo = 0;
cfg.LinuxOSSWaveOutDev = new char[MAX_STRING_SIZE];
snprintf(cfg.LinuxOSSWaveOutDev, MAX_STRING_SIZE, "/dev/dsp");
- cfg.LinuxOSSSeqInDev = new char[MAX_STRING_SIZE];
+ cfg.LinuxOSSSeqInDev = new char[MAX_STRING_SIZE];
snprintf(cfg.LinuxOSSSeqInDev, MAX_STRING_SIZE, "/dev/sequencer");
cfg.DumpFile = "zynaddsubfx_dump.txt";
- cfg.WindowsWaveOutId = 0;
- cfg.WindowsMidiInId = 0;
+ cfg.WindowsWaveOutId = 0;
+ cfg.WindowsMidiInId = 0;
- cfg.BankUIAutoClose = 0;
- cfg.DumpNotesToFile = 0;
- cfg.DumpAppend = 1;
+ cfg.BankUIAutoClose = 0;
+ cfg.DumpNotesToFile = 0;
+ cfg.DumpAppend = 1;
- cfg.GzipCompression = 3;
+ cfg.GzipCompression = 3;
- cfg.Interpolation = 0;
- cfg.CheckPADsynth = 1;
+ cfg.Interpolation = 0;
+ cfg.CheckPADsynth = 1;
cfg.UserInterfaceMode = 0;
cfg.VirKeybLayout = 1;
@@ -129,7 +129,7 @@ void Config::clearbankrootdirlist()
void Config::clearpresetsdirlist()
{
for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i)
- cfg.presetsDirList[i].clear();
+ cfg.presetsDirList[i].clear();
}
void Config::readConfig(const char *filename)
@@ -138,96 +138,94 @@ void Config::readConfig(const char *filename)
if(xmlcfg.loadXMLfile(filename) < 0)
return;
if(xmlcfg.enterbranch("CONFIGURATION")) {
- cfg.SampleRate = xmlcfg.getpar("sample_rate",
- cfg.SampleRate,
- 4000,
- 1024000);
+ cfg.SampleRate = xmlcfg.getpar("sample_rate",
+ cfg.SampleRate,
+ 4000,
+ 1024000);
cfg.SoundBufferSize = xmlcfg.getpar("sound_buffer_size",
- cfg.SoundBufferSize,
- 16,
- 8192);
+ cfg.SoundBufferSize,
+ 16,
+ 8192);
cfg.OscilSize = xmlcfg.getpar("oscil_size",
- cfg.OscilSize,
- MAX_AD_HARMONICS * 2,
- 131072);
- cfg.SwapStereo = xmlcfg.getpar("swap_stereo",
- cfg.SwapStereo,
- 0,
- 1);
+ cfg.OscilSize,
+ MAX_AD_HARMONICS * 2,
+ 131072);
+ cfg.SwapStereo = xmlcfg.getpar("swap_stereo",
+ cfg.SwapStereo,
+ 0,
+ 1);
cfg.BankUIAutoClose = xmlcfg.getpar("bank_window_auto_close",
- cfg.BankUIAutoClose,
- 0,
- 1);
+ cfg.BankUIAutoClose,
+ 0,
+ 1);
cfg.DumpNotesToFile = xmlcfg.getpar("dump_notes_to_file",
- cfg.DumpNotesToFile,
- 0,
- 1);
- cfg.DumpAppend = xmlcfg.getpar("dump_append",
- cfg.DumpAppend,
- 0,
- 1);
+ cfg.DumpNotesToFile,
+ 0,
+ 1);
+ cfg.DumpAppend = xmlcfg.getpar("dump_append",
+ cfg.DumpAppend,
+ 0,
+ 1);
cfg.DumpFile = xmlcfg.getparstr("dump_file", "");
cfg.GzipCompression = xmlcfg.getpar("gzip_compression",
- cfg.GzipCompression,
- 0,
- 9);
+ cfg.GzipCompression,
+ 0,
+ 9);
cfg.currentBankDir = xmlcfg.getparstr("bank_current", "");
- cfg.Interpolation = xmlcfg.getpar("interpolation",
+ cfg.Interpolation = xmlcfg.getpar("interpolation",
cfg.Interpolation,
0,
1);
cfg.CheckPADsynth = xmlcfg.getpar("check_pad_synth",
- cfg.CheckPADsynth,
- 0,
- 1);
+ cfg.CheckPADsynth,
+ 0,
+ 1);
cfg.UserInterfaceMode = xmlcfg.getpar("user_interface_mode",
- cfg.UserInterfaceMode,
- 0,
- 2);
- cfg.VirKeybLayout = xmlcfg.getpar("virtual_keyboard_layout",
- cfg.VirKeybLayout,
- 0,
- 10);
+ cfg.UserInterfaceMode,
+ 0,
+ 2);
+ cfg.VirKeybLayout = xmlcfg.getpar("virtual_keyboard_layout",
+ cfg.VirKeybLayout,
+ 0,
+ 10);
//get bankroot dirs
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i) {
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i)
if(xmlcfg.enterbranch("BANKROOT", i)) {
cfg.bankRootDirList[i] = xmlcfg.getparstr("bank_root", "");
xmlcfg.exitbranch();
}
- }
//get preset root dirs
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i) {
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i)
if(xmlcfg.enterbranch("PRESETSROOT", i)) {
cfg.presetsDirList[i] = xmlcfg.getparstr("presets_root", "");
xmlcfg.exitbranch();
}
- }
//linux stuff
xmlcfg.getparstr("linux_oss_wave_out_dev",
- cfg.LinuxOSSWaveOutDev,
- MAX_STRING_SIZE);
+ cfg.LinuxOSSWaveOutDev,
+ MAX_STRING_SIZE);
xmlcfg.getparstr("linux_oss_seq_in_dev",
- cfg.LinuxOSSSeqInDev,
- MAX_STRING_SIZE);
+ cfg.LinuxOSSSeqInDev,
+ MAX_STRING_SIZE);
//windows stuff
cfg.WindowsWaveOutId = xmlcfg.getpar("windows_wave_out_id",
- cfg.WindowsWaveOutId,
- 0,
- winwavemax);
- cfg.WindowsMidiInId = xmlcfg.getpar("windows_midi_in_id",
- cfg.WindowsMidiInId,
- 0,
- winmidimax);
+ cfg.WindowsWaveOutId,
+ 0,
+ winwavemax);
+ cfg.WindowsMidiInId = xmlcfg.getpar("windows_midi_in_id",
+ cfg.WindowsMidiInId,
+ 0,
+ winmidimax);
xmlcfg.exitbranch();
}
@@ -300,4 +298,3 @@ void Config::getConfigFileName(char *name, int namesize)
name[0] = 0;
snprintf(name, namesize, "%s%s", getenv("HOME"), "/.zynaddsubfxXML.cfg");
}
-
diff --git a/src/Misc/Config.h b/src/Misc/Config.h
@@ -46,9 +46,9 @@ class Config
std::string DumpFile;
std::string bankRootDirList[MAX_BANK_ROOT_DIRS], currentBankDir;
std::string presetsDirList[MAX_BANK_ROOT_DIRS];
- int CheckPADsynth;
- int UserInterfaceMode;
- int VirKeybLayout;
+ int CheckPADsynth;
+ int UserInterfaceMode;
+ int VirKeybLayout;
std::string LinuxALSAaudioDev;
std::string nameTag;
} cfg;
@@ -71,4 +71,3 @@ class Config
void getConfigFileName(char *name, int namesize);
};
#endif
-
diff --git a/src/Misc/Control.h b/src/Misc/Control.h
@@ -97,4 +97,3 @@ class ControlUser
};
#endif /* _CONTROL_H_ */
-
diff --git a/src/Misc/Dump.cpp b/src/Misc/Dump.cpp
@@ -107,12 +107,12 @@ void Dump::dumpcontroller(char chan, unsigned int type, int par)
if(file == NULL)
return;
switch(type) {
- case C_pitchwheel:
- fprintf(file, "P %d -> %d %d\n", tick, chan, par);
- break;
- default:
- fprintf(file, "C %d -> %d %d %d\n", tick, chan, type, par);
- break;
+ case C_pitchwheel:
+ fprintf(file, "P %d -> %d %d\n", tick, chan, par);
+ break;
+ default:
+ fprintf(file, "C %d -> %d %d %d\n", tick, chan, type, par);
+ break;
}
#ifndef JACKAUDIOOUT
if(k++ > 25) {
@@ -121,4 +121,3 @@ void Dump::dumpcontroller(char chan, unsigned int type, int par)
}
#endif
}
-
diff --git a/src/Misc/Dump.h b/src/Misc/Dump.h
@@ -61,4 +61,3 @@ class Dump
int keyspressed;
};
#endif
-
diff --git a/src/Misc/LASHClient.cpp b/src/Misc/LASHClient.cpp
@@ -101,4 +101,3 @@ void LASHClient::confirmevent(Event event)
if(event == Restore)
lash_send_event(client, lash_event_new_with_type(LASH_Restore_File));
}
-
diff --git a/src/Misc/LASHClient.h b/src/Misc/LASHClient.h
@@ -61,4 +61,3 @@ class LASHClient
#endif
-
diff --git a/src/Misc/Master.cpp b/src/Misc/Master.cpp
@@ -44,7 +44,7 @@ Master::Master()
pthread_mutex_init(&vumutex, NULL);
fft = new FFTwrapper(OSCIL_SIZE);
- shutup = 0;
+ shutup = 0;
for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
vuoutpeakpart[npart] = 1e-9;
fakepeakpart[npart] = 0;
@@ -99,8 +99,7 @@ void Master::defaults()
bool Master::mutexLock(lockset request)
{
- switch (request)
- {
+ switch(request) {
case MUTEX_TRYLOCK:
return !pthread_mutex_trylock(&mutex);
case MUTEX_LOCK:
@@ -123,13 +122,12 @@ Master &Master::getInstance()
void Master::noteOn(char chan, char note, char velocity)
{
if(velocity) {
- for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
if(chan == part[npart]->Prcvchn) {
fakepeakpart[npart] = velocity * 2;
if(part[npart]->Penabled)
part[npart]->NoteOn(note, velocity, keyshift);
}
- }
}
else
this->noteOff(chan, note);
@@ -159,16 +157,16 @@ void Master::setController(char chan, int type, int par)
if(ctl.getnrpn(&parhi, &parlo, &valhi, &vallo) == 0) //this is NRPN
//fprintf(stderr,"rcv. NRPN: %d %d %d %d\n",parhi,parlo,valhi,vallo);
switch(parhi) {
- case 0x04: //System Effects
- if(parlo < NUM_SYS_EFX)
- sysefx[parlo]->seteffectpar_nolock(valhi, vallo);
- ;
- break;
- case 0x08: //Insertion Effects
- if(parlo < NUM_INS_EFX)
- insefx[parlo]->seteffectpar_nolock(valhi, vallo);
- ;
- break;
+ case 0x04: //System Effects
+ if(parlo < NUM_SYS_EFX)
+ sysefx[parlo]->seteffectpar_nolock(valhi, vallo);
+ ;
+ break;
+ case 0x08: //Insertion Effects
+ if(parlo < NUM_INS_EFX)
+ insefx[parlo]->seteffectpar_nolock(valhi, vallo);
+ ;
+ break;
}
;
}
@@ -220,7 +218,7 @@ void Master::vuUpdate(const float *outl, const float *outr)
vuoutpeakpart[npart] = 1.0e-12f;
if(part[npart]->Penabled != 0) {
float *outl = part[npart]->partoutl,
- *outr = part[npart]->partoutr;
+ *outr = part[npart]->partoutr;
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmp = fabs(outl[i] + outr[i]);
if(tmp > vuoutpeakpart[npart])
@@ -229,8 +227,8 @@ void Master::vuUpdate(const float *outl, const float *outr)
vuoutpeakpart[npart] *= volume;
}
else
- if(fakepeakpart[npart] > 1)
- fakepeakpart[npart]--;
+ if(fakepeakpart[npart] > 1)
+ fakepeakpart[npart]--;
}
}
@@ -264,7 +262,7 @@ void Master::AudioOut(float *outl, float *outr)
{
//Swaps the Left channel with Right Channel
if(swaplr)
- swap(outl,outr);
+ swap(outl, outr);
//clean up the output samples (should not be needed?)
memset(outl, 0, sizeof(float) * SOUND_BUFFER_SIZE);
@@ -276,14 +274,13 @@ void Master::AudioOut(float *outl, float *outr)
part[npart]->ComputePartSmps();
//Insertion effects
- for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx) {
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx)
if(Pinsparts[nefx] >= 0) {
int efxpart = Pinsparts[nefx];
if(part[efxpart]->Penabled)
insefx[nefx]->out(part[efxpart]->partoutl,
part[efxpart]->partoutr);
}
- }
//Apply the part volumes and pannings (after insertion effects)
@@ -292,10 +289,10 @@ void Master::AudioOut(float *outl, float *outr)
continue;
Stereo<float> newvol(part[npart]->volume),
- oldvol(part[npart]->oldvolumel,
- part[npart]->oldvolumer);
+ oldvol(part[npart]->oldvolumel,
+ part[npart]->oldvolumer);
- float pan = part[npart]->panning;
+ float pan = part[npart]->panning;
if(pan < 0.5f)
newvol.l *= pan * 2.0f;
else
@@ -306,21 +303,20 @@ void Master::AudioOut(float *outl, float *outr)
|| ABOVE_AMPLITUDE_THRESHOLD(oldvol.r, newvol.r)) {
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
Stereo<float> vol(INTERPOLATE_AMPLITUDE(oldvol.l, newvol.l,
- i, SOUND_BUFFER_SIZE),
- INTERPOLATE_AMPLITUDE(oldvol.r, newvol.r,
- i, SOUND_BUFFER_SIZE));
+ i, SOUND_BUFFER_SIZE),
+ INTERPOLATE_AMPLITUDE(oldvol.r, newvol.r,
+ i, SOUND_BUFFER_SIZE));
part[npart]->partoutl[i] *= vol.l;
part[npart]->partoutr[i] *= vol.r;
}
part[npart]->oldvolumel = newvol.l;
part[npart]->oldvolumer = newvol.r;
}
- else {
+ else
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //the volume did not changed
part[npart]->partoutl[i] *= newvol.l;
part[npart]->partoutr[i] *= newvol.r;
}
- }
}
@@ -354,7 +350,7 @@ void Master::AudioOut(float *outl, float *outr)
}
// system effect send to next ones
- for(int nefxfrom = 0; nefxfrom < nefx; ++nefxfrom) {
+ for(int nefxfrom = 0; nefxfrom < nefx; ++nefxfrom)
if(Psysefxsend[nefxfrom][nefx] != 0) {
const float vol = sysefxsend[nefxfrom][nefx];
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
@@ -362,7 +358,6 @@ void Master::AudioOut(float *outl, float *outr)
tmpmixr[i] += sysefx[nefxfrom]->efxoutr[i] * vol;
}
}
- }
sysefx[nefx]->out(tmpmixl, tmpmixr);
@@ -378,14 +373,12 @@ void Master::AudioOut(float *outl, float *outr)
}
//Mix all parts
- for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
- if(part[npart]->Penabled) { //only mix active parts
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
+ if(part[npart]->Penabled) //only mix active parts
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //the volume did not changed
outl[i] += part[npart]->partoutl[i];
outr[i] += part[npart]->partoutr[i];
}
- }
- }
//Insertion effects for Master Out
for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx)
@@ -429,7 +422,7 @@ void Master::GetAudioOutSamples(size_t nsamples,
float *outr)
{
static float *bufl = new float[SOUND_BUFFER_SIZE],
- *bufr = new float[SOUND_BUFFER_SIZE];
+ *bufr = new float[SOUND_BUFFER_SIZE];
static off_t off = 0;
static size_t smps = 0;
@@ -444,21 +437,22 @@ void Master::GetAudioOutSamples(size_t nsamples,
while(nsamples) {
//use all available samples
if(nsamples >= smps) {
- memcpy(outl+out_off, bufl+off, sizeof(float) * smps);
- memcpy(outr+out_off, bufr+off, sizeof(float) * smps);
+ memcpy(outl + out_off, bufl + off, sizeof(float) * smps);
+ memcpy(outr + out_off, bufr + off, sizeof(float) * smps);
//generate samples
AudioOut(bufl, bufr);
- off = 0;
+ off = 0;
smps = SOUND_BUFFER_SIZE;
out_off += smps;
nsamples -= smps;
- } else { //use some samples
- memcpy(outl+out_off, bufl+off, sizeof(float) * nsamples);
- memcpy(outr+out_off, bufr+off, sizeof(float) * nsamples);
- smps -= nsamples;
- off += nsamples;
+ }
+ else { //use some samples
+ memcpy(outl + out_off, bufl + off, sizeof(float) * nsamples);
+ memcpy(outr + out_off, bufr + off, sizeof(float) * nsamples);
+ smps -= nsamples;
+ off += nsamples;
nsamples = 0;
}
}
@@ -751,4 +745,3 @@ void Master::getfromXML(XMLwrapper *xml)
xml->exitbranch();
}
}
-
diff --git a/src/Misc/Master.h b/src/Misc/Master.h
@@ -33,13 +33,15 @@
#include "Dump.h"
#include "XMLwrapper.h"
-typedef enum { MUTEX_TRYLOCK, MUTEX_LOCK, MUTEX_UNLOCK } lockset;
+typedef enum {
+ MUTEX_TRYLOCK, MUTEX_LOCK, MUTEX_UNLOCK
+} lockset;
extern Dump dump;
typedef struct vuData_t {
float outpeakl, outpeakr, maxoutpeakl, maxoutpeakr,
- rmspeakl, rmspeakr;
+ rmspeakl, rmspeakr;
int clipped;
} vuData;
@@ -142,15 +144,15 @@ class Master
//peaks for part VU-meters
/**\todo synchronize this with a mutex*/
- float vuoutpeakpart[NUM_MIDI_PARTS];
+ float vuoutpeakpart[NUM_MIDI_PARTS];
unsigned char fakepeakpart[NUM_MIDI_PARTS]; //this is used to compute the "peak" when the part is disabled
Controller ctl;
- bool swaplr; //if L and R are swapped
+ bool swaplr; //if L and R are swapped
//other objects
Microtonal microtonal;
- Bank bank;
+ Bank bank;
FFTwrapper *fft;
pthread_mutex_t mutex;
@@ -158,14 +160,13 @@ class Master
private:
- bool nullRun;
+ bool nullRun;
vuData vu;
- float volume;
- float sysefxvol[NUM_SYS_EFX][NUM_MIDI_PARTS];
- float sysefxsend[NUM_SYS_EFX][NUM_SYS_EFX];
- int keyshift;
+ float volume;
+ float sysefxvol[NUM_SYS_EFX][NUM_MIDI_PARTS];
+ float sysefxsend[NUM_SYS_EFX][NUM_SYS_EFX];
+ int keyshift;
};
#endif
-
diff --git a/src/Misc/Microtonal.cpp b/src/Misc/Microtonal.cpp
@@ -37,11 +37,11 @@ void Microtonal::defaults()
{
Pinvertupdown = 0;
Pinvertupdowncenter = 60;
- octavesize = 12;
- Penabled = 0;
- PAnote = 69;
- PAfreq = 440.0f;
- Pscaleshift = 64;
+ octavesize = 12;
+ Penabled = 0;
+ PAnote = 69;
+ PAfreq = 440.0f;
+ Pscaleshift = 64;
Pfirstkey = 0;
Plastkey = 127;
@@ -56,10 +56,10 @@ void Microtonal::defaults()
octave[i].tuning = tmpoctave[i].tuning = powf(
2,
(i % octavesize
- + 1) / 12.0f);
- octave[i].type = tmpoctave[i].type = 1;
- octave[i].x1 = tmpoctave[i].x1 = (i % octavesize + 1) * 100;
- octave[i].x2 = tmpoctave[i].x2 = 0;
+ + 1) / 12.0f);
+ octave[i].type = tmpoctave[i].type = 1;
+ octave[i].x1 = tmpoctave[i].x1 = (i % octavesize + 1) * 100;
+ octave[i].x2 = tmpoctave[i].x2 = 0;
}
octave[11].type = 2;
octave[11].x1 = 2;
@@ -106,12 +106,13 @@ float Microtonal::getnotefreq(int note, int keyshift) const
note = (int) Pinvertupdowncenter * 2 - note;
//compute global fine detune
- float globalfinedetunerap = powf(2.0f, (Pglobalfinedetune - 64.0f) / 1200.0f); //-64.0f .. 63.0f cents
+ float globalfinedetunerap = powf(2.0f,
+ (Pglobalfinedetune - 64.0f) / 1200.0f); //-64.0f .. 63.0f cents
if(Penabled == 0)
return powf(2.0f,
- (note - PAnote
- + keyshift) / 12.0f) * PAfreq * globalfinedetunerap; //12tET
+ (note - PAnote
+ + keyshift) / 12.0f) * PAfreq * globalfinedetunerap; //12tET
int scaleshift =
((int)Pscaleshift - 64 + (int) octavesize * 100) % octavesize;
@@ -145,7 +146,8 @@ float Microtonal::getnotefreq(int note, int keyshift) const
0) ? (1.0f) : (octave[(deltanote - 1) % octavesize].tuning);
if(deltanote != 0)
rap_anote_middlenote *=
- powf(octave[octavesize - 1].tuning, (deltanote - 1) / octavesize);
+ powf(octave[octavesize - 1].tuning,
+ (deltanote - 1) / octavesize);
if(minus != 0)
rap_anote_middlenote = 1.0f / rap_anote_middlenote;
@@ -178,14 +180,14 @@ float Microtonal::getnotefreq(int note, int keyshift) const
return freq * rap_keyshift;
}
else { //if the mapping is disabled
- int nt = note - PAnote + scaleshift;
- int ntkey = (nt + (int)octavesize * 100) % octavesize;
- int ntoct = (nt - ntkey) / octavesize;
+ int nt = note - PAnote + scaleshift;
+ int ntkey = (nt + (int)octavesize * 100) % octavesize;
+ int ntoct = (nt - ntkey) / octavesize;
float oct = octave[octavesize - 1].tuning;
float freq =
octave[(ntkey + octavesize - 1) % octavesize].tuning *powf(oct,
- ntoct)
+ ntoct)
* PAfreq;
if(ntkey == 0)
freq /= oct;
@@ -254,7 +256,7 @@ bool Microtonal::operator!=(const Microtonal µ) const
*/
int Microtonal::linetotunings(unsigned int nline, const char *line)
{
- int x1 = -1, x2 = -1, type = -1;
+ int x1 = -1, x2 = -1, type = -1;
float x = -1.0f, tmp, tuning = 1.0f;
if(strstr(line, "/") == NULL) {
if(strstr(line, ".") == NULL) { // M case (M=M/1)
@@ -288,16 +290,16 @@ int Microtonal::linetotunings(unsigned int nline, const char *line)
x = ((float) x1) / x2;
}
switch(type) {
- case 1:
- x1 = (int) floor(x);
- tmp = fmod(x, 1.0f);
- x2 = (int) (floor(tmp * 1e6));
- tuning = powf(2.0f, x / 1200.0f);
- break;
- case 2:
- x = ((float)x1) / x2;
- tuning = x;
- break;
+ case 1:
+ x1 = (int) floor(x);
+ tmp = fmod(x, 1.0f);
+ x2 = (int) (floor(tmp * 1e6));
+ tuning = powf(2.0f, x / 1200.0f);
+ break;
+ case 2:
+ x = ((float)x1) / x2;
+ tuning = x;
+ break;
}
tmpoctave[nline].tuning = tuning;
@@ -684,4 +686,3 @@ int Microtonal::loadXML(const char *filename)
delete (xml);
return 0;
}
-
diff --git a/src/Misc/Microtonal.h b/src/Misc/Microtonal.h
@@ -132,4 +132,3 @@ class Microtonal
};
#endif
-
diff --git a/src/Misc/Part.cpp b/src/Misc/Part.cpp
@@ -38,10 +38,10 @@
Part::Part(Microtonal *microtonal_, FFTwrapper *fft_, pthread_mutex_t *mutex_)
{
microtonal = microtonal_;
- fft = fft_;
- mutex = mutex_;
- partoutl = new float [SOUND_BUFFER_SIZE];
- partoutr = new float [SOUND_BUFFER_SIZE];
+ fft = fft_;
+ mutex = mutex_;
+ partoutl = new float [SOUND_BUFFER_SIZE];
+ partoutr = new float [SOUND_BUFFER_SIZE];
for(int n = 0; n < NUM_KIT_ITEMS; ++n) {
kit[n].Pname = new unsigned char [PART_MAX_NAME_LEN];
@@ -81,7 +81,7 @@ Part::Part(Microtonal *microtonal_, FFTwrapper *fft_, pthread_mutex_t *mutex_)
}
cleanup();
- Pname = new unsigned char [PART_MAX_NAME_LEN];
+ Pname = new unsigned char [PART_MAX_NAME_LEN];
oldvolumel = oldvolumer = 0.5f;
lastnote = -1;
@@ -100,12 +100,12 @@ void Part::defaults()
Ppolymode = 1;
Plegatomode = 0;
setPvolume(96);
- Pkeyshift = 64;
- Prcvchn = 0;
+ Pkeyshift = 64;
+ Prcvchn = 0;
setPpanning(64);
- Pvelsns = 64;
- Pveloffs = 64;
- Pkeylimit = 15;
+ Pvelsns = 64;
+ Pveloffs = 64;
+ Pkeylimit = 15;
defaultsinstrument();
ctl.defaults();
}
@@ -162,12 +162,11 @@ void Part::cleanup(bool final)
ctl.resetall();
for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx)
partefx[nefx]->cleanup();
- for(int n = 0; n < NUM_PART_EFX + 1; ++n) {
+ for(int n = 0; n < NUM_PART_EFX + 1; ++n)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partfxinputl[n][i] = final ? 0.0f : denormalkillbuf[i];
partfxinputr[n][i] = final ? 0.0f : denormalkillbuf[i];
}
- }
}
Part::~Part()
@@ -236,17 +235,17 @@ void Part::NoteOn(unsigned char note,
lastnote = note;
- pos = -1;
- for(i = 0; i < POLIPHONY; ++i) {
+ pos = -1;
+ for(i = 0; i < POLIPHONY; ++i)
if(partnote[i].status == KEY_OFF) {
pos = i;
break;
}
- }
if((Plegatomode != 0) && (Pdrummode == 0)) {
if(Ppolymode != 0) {
- fprintf( stderr,
+ fprintf(
+ stderr,
"ZynAddSubFX WARNING: Poly and Legato modes are both On, that should not happen ! ... Disabling Legato mode ! - (Part.cpp::NoteOn(..))\n");
Plegatomode = 0;
}
@@ -355,9 +354,15 @@ void Part::NoteOn(unsigned char note,
&& (partnote[pos].kititem[0].adnote != NULL)
&& (partnote[posb].kititem[0].adnote != NULL)) {
partnote[pos].kititem[0].adnote->legatonote(notebasefreq,
- vel, portamento, note, true);//'true' is to tell it it's being called from here.
+ vel,
+ portamento,
+ note,
+ true); //'true' is to tell it it's being called from here.
partnote[posb].kititem[0].adnote->legatonote(notebasefreq,
- vel, portamento, note, true);
+ vel,
+ portamento,
+ note,
+ true);
}
if((kit[0].Psubenabled != 0)
@@ -390,12 +395,14 @@ void Part::NoteOn(unsigned char note,
|| (lastnotecopy > kit[item].Pmaxkey))
continue; // We will not perform legato across 2 key regions.
- partnote[pos].kititem[ci].sendtoparteffect =
+ partnote[pos].kititem[ci].sendtoparteffect =
(kit[item].Psendtoparteffect <
- NUM_PART_EFX ? kit[item].Psendtoparteffect : NUM_PART_EFX);//if this parameter is 127 for "unprocessed"
+ NUM_PART_EFX ? kit[item].Psendtoparteffect :
+ NUM_PART_EFX); //if this parameter is 127 for "unprocessed"
partnote[posb].kititem[ci].sendtoparteffect =
(kit[item].Psendtoparteffect <
- NUM_PART_EFX ? kit[item].Psendtoparteffect : NUM_PART_EFX);
+ NUM_PART_EFX ? kit[item].Psendtoparteffect :
+ NUM_PART_EFX);
if((kit[item].Padenabled != 0) && (kit[item].adpars != NULL)
&& (partnote[pos].kititem[ci].adnote != NULL)
@@ -451,13 +458,28 @@ void Part::NoteOn(unsigned char note,
partnote[pos].kititem[0].sendtoparteffect = 0;
if(kit[0].Padenabled != 0)
partnote[pos].kititem[0].adnote = new ADnote(kit[0].adpars,
- &ctl, notebasefreq, vel, portamento, note, false);
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ false);
if(kit[0].Psubenabled != 0)
partnote[pos].kititem[0].subnote = new SUBnote(kit[0].subpars,
- &ctl, notebasefreq, vel, portamento, note, false);
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ false);
if(kit[0].Ppadenabled != 0)
partnote[pos].kititem[0].padnote = new PADnote(kit[0].padpars,
- &ctl, notebasefreq, vel, portamento, note, false);
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ false);
if((kit[0].Padenabled != 0) || (kit[0].Psubenabled != 0)
|| (kit[0].Ppadenabled != 0))
partnote[pos].itemsplaying++;
@@ -467,19 +489,36 @@ void Part::NoteOn(unsigned char note,
partnote[posb].kititem[0].sendtoparteffect = 0;
if(kit[0].Padenabled != 0)
partnote[posb].kititem[0].adnote = new ADnote(kit[0].adpars,
- &ctl, notebasefreq, vel, portamento, note, true);//true for silent.
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ true); //true for silent.
if(kit[0].Psubenabled != 0)
- partnote[posb].kititem[0].subnote = new SUBnote( kit[0].subpars,
- &ctl, notebasefreq, vel, portamento, note, true);
+ partnote[posb].kititem[0].subnote = new SUBnote(
+ kit[0].subpars,
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ true);
if(kit[0].Ppadenabled != 0)
- partnote[posb].kititem[0].padnote = new PADnote( kit[0].padpars,
- &ctl, notebasefreq, vel, portamento, note, true);
+ partnote[posb].kititem[0].padnote = new PADnote(
+ kit[0].padpars,
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ true);
if((kit[0].Padenabled != 0) || (kit[0].Psubenabled != 0)
|| (kit[0].Ppadenabled != 0))
partnote[posb].itemsplaying++;
}
}
- else { //init the notes for the "kit mode"
+ else //init the notes for the "kit mode"
for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
if(kit[item].Pmuted != 0)
continue;
@@ -491,40 +530,75 @@ void Part::NoteOn(unsigned char note,
//if this parameter is 127 for "unprocessed"
partnote[pos].kititem[ci].sendtoparteffect =
(kit[item].Psendtoparteffect < NUM_PART_EFX ?
- kit[item]. Psendtoparteffect : NUM_PART_EFX);
+ kit[item].Psendtoparteffect : NUM_PART_EFX);
if((kit[item].adpars != NULL) && ((kit[item].Padenabled) != 0))
- partnote[pos].kititem[ci].adnote = new ADnote( kit[item].adpars,
- &ctl, notebasefreq, vel, portamento, note, false);
+ partnote[pos].kititem[ci].adnote = new ADnote(
+ kit[item].adpars,
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ false);
if((kit[item].subpars != NULL) && ((kit[item].Psubenabled) != 0))
- partnote[pos].kititem[ci].subnote = new SUBnote( kit[item].subpars,
- &ctl, notebasefreq, vel, portamento, note, false);
+ partnote[pos].kititem[ci].subnote = new SUBnote(
+ kit[item].subpars,
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ false);
if((kit[item].padpars != NULL) && ((kit[item].Ppadenabled) != 0))
- partnote[pos].kititem[ci].padnote = new PADnote( kit[item].padpars,
- &ctl, notebasefreq, vel, portamento, note, false);
+ partnote[pos].kititem[ci].padnote = new PADnote(
+ kit[item].padpars,
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ false);
// Spawn another note (but silent) if legatomodevalid==true
if(legatomodevalid) {
partnote[posb].kititem[ci].sendtoparteffect =
(kit[item].Psendtoparteffect <
- NUM_PART_EFX ? kit[item].Psendtoparteffect : NUM_PART_EFX); //if this parameter is 127 for "unprocessed"
+ NUM_PART_EFX ? kit[item].Psendtoparteffect :
+ NUM_PART_EFX); //if this parameter is 127 for "unprocessed"
if((kit[item].adpars != NULL)
&& ((kit[item].Padenabled) != 0))
- partnote[posb].kititem[ci].adnote = new ADnote( kit[item].adpars,
- &ctl, notebasefreq, vel, portamento, note, true); //true for silent.
+ partnote[posb].kititem[ci].adnote = new ADnote(
+ kit[item].adpars,
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ true); //true for silent.
if((kit[item].subpars != NULL)
&& ((kit[item].Psubenabled) != 0))
partnote[posb].kititem[ci].subnote =
new SUBnote(kit[item].subpars,
- &ctl, notebasefreq, vel, portamento, note, true);
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ true);
if((kit[item].padpars != NULL)
&& ((kit[item].Ppadenabled) != 0))
partnote[posb].kititem[ci].padnote =
new PADnote(kit[item].padpars,
- &ctl, notebasefreq, vel, portamento, note, true);
+ &ctl,
+ notebasefreq,
+ vel,
+ portamento,
+ note,
+ true);
if((kit[item].adpars != NULL) || (kit[item].subpars != NULL))
partnote[posb].itemsplaying++;
@@ -539,7 +613,6 @@ void Part::NoteOn(unsigned char note,
break;
}
}
- }
}
//this only relase the keys if there is maximum number of keys allowed
@@ -564,7 +637,7 @@ void Part::NoteOff(unsigned char note) //relase the key
MonoMemRenote(); // To play most recent still held note.
else
RelaseNotePos(i);
- /// break;
+ /// break;
}
else //the sustain pedal is pushed
partnote[i].status = KEY_RELASED_AND_SUSTAINED;
@@ -577,88 +650,89 @@ void Part::NoteOff(unsigned char note) //relase the key
void Part::SetController(unsigned int type, int par)
{
switch(type) {
- case C_pitchwheel:
- ctl.setpitchwheel(par);
- break;
- case C_expression:
- ctl.setexpression(par);
- setPvolume(Pvolume); //update the volume
- break;
- case C_portamento:
- ctl.setportamento(par);
- break;
- case C_panning:
- ctl.setpanning(par);
- setPpanning(Ppanning); //update the panning
- break;
- case C_filtercutoff:
- ctl.setfiltercutoff(par);
- break;
- case C_filterq:
- ctl.setfilterq(par);
- break;
- case C_bandwidth:
- ctl.setbandwidth(par);
- break;
- case C_modwheel:
- ctl.setmodwheel(par);
- break;
- case C_fmamp:
- ctl.setfmamp(par);
- break;
- case C_volume:
- ctl.setvolume(par);
- if(ctl.volume.receive != 0)
- volume = ctl.volume.volume;
- else
- setPvolume(Pvolume);
- break;
- case C_sustain:
- ctl.setsustain(par);
- if(ctl.sustain.sustain == 0)
+ case C_pitchwheel:
+ ctl.setpitchwheel(par);
+ break;
+ case C_expression:
+ ctl.setexpression(par);
+ setPvolume(Pvolume); //update the volume
+ break;
+ case C_portamento:
+ ctl.setportamento(par);
+ break;
+ case C_panning:
+ ctl.setpanning(par);
+ setPpanning(Ppanning); //update the panning
+ break;
+ case C_filtercutoff:
+ ctl.setfiltercutoff(par);
+ break;
+ case C_filterq:
+ ctl.setfilterq(par);
+ break;
+ case C_bandwidth:
+ ctl.setbandwidth(par);
+ break;
+ case C_modwheel:
+ ctl.setmodwheel(par);
+ break;
+ case C_fmamp:
+ ctl.setfmamp(par);
+ break;
+ case C_volume:
+ ctl.setvolume(par);
+ if(ctl.volume.receive != 0)
+ volume = ctl.volume.volume;
+ else
+ setPvolume(Pvolume);
+ break;
+ case C_sustain:
+ ctl.setsustain(par);
+ if(ctl.sustain.sustain == 0)
+ RelaseSustainedKeys();
+ break;
+ case C_allsoundsoff:
+ AllNotesOff(); //Panic
+ break;
+ case C_resetallcontrollers:
+ ctl.resetall();
RelaseSustainedKeys();
- break;
- case C_allsoundsoff:
- AllNotesOff(); //Panic
- break;
- case C_resetallcontrollers:
- ctl.resetall();
- RelaseSustainedKeys();
- if(ctl.volume.receive != 0)
- volume = ctl.volume.volume;
- else
- setPvolume(Pvolume);
- setPvolume(Pvolume); //update the volume
- setPpanning(Ppanning); //update the panning
+ if(ctl.volume.receive != 0)
+ volume = ctl.volume.volume;
+ else
+ setPvolume(Pvolume);
+ setPvolume(Pvolume); //update the volume
+ setPpanning(Ppanning); //update the panning
- for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
- if(kit[item].adpars == NULL)
- continue;
- kit[item].adpars->GlobalPar.Reson->
- sendcontroller(C_resonance_center, 1.0f);
+ for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
+ if(kit[item].adpars == NULL)
+ continue;
+ kit[item].adpars->GlobalPar.Reson->
+ sendcontroller(C_resonance_center, 1.0f);
- kit[item].adpars->GlobalPar.Reson->
- sendcontroller(C_resonance_bandwidth, 1.0f);
- }
- //more update to add here if I add controllers
- break;
- case C_allnotesoff:
- RelaseAllKeys();
- break;
- case C_resonance_center:
- ctl.setresonancecenter(par);
- for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
- if(kit[item].adpars == NULL)
- continue;
- kit[item].adpars->GlobalPar.Reson->
- sendcontroller(C_resonance_center, ctl.resonancecenter.relcenter);
- }
- break;
- case C_resonance_bandwidth:
- ctl.setresonancebw(par);
- kit[0].adpars->GlobalPar.Reson->
- sendcontroller(C_resonance_bandwidth, ctl.resonancebandwidth.relbw);
- break;
+ kit[item].adpars->GlobalPar.Reson->
+ sendcontroller(C_resonance_bandwidth, 1.0f);
+ }
+ //more update to add here if I add controllers
+ break;
+ case C_allnotesoff:
+ RelaseAllKeys();
+ break;
+ case C_resonance_center:
+ ctl.setresonancecenter(par);
+ for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
+ if(kit[item].adpars == NULL)
+ continue;
+ kit[item].adpars->GlobalPar.Reson->
+ sendcontroller(C_resonance_center,
+ ctl.resonancecenter.relcenter);
+ }
+ break;
+ case C_resonance_bandwidth:
+ ctl.setresonancebw(par);
+ kit[0].adpars->GlobalPar.Reson->
+ sendcontroller(C_resonance_bandwidth, ctl.resonancebandwidth.relbw);
+ break;
}
}
/*
@@ -774,7 +848,7 @@ void Part::setkeylimit(unsigned char Pkeylimit)
notecount++;
int oldestnotepos = -1;
- if(notecount > keylimit) { //find out the oldest note
+ if(notecount > keylimit) //find out the oldest note
for(int i = 0; i < POLIPHONY; ++i) {
int maxtime = 0;
if(((partnote[i].status == KEY_PLAYING)
@@ -784,7 +858,6 @@ void Part::setkeylimit(unsigned char Pkeylimit)
oldestnotepos = i;
}
}
- }
if(oldestnotepos != -1)
RelaseNotePos(oldestnotepos);
}
@@ -810,9 +883,11 @@ void Part::RunNote(unsigned int k)
SynthNote **note;
if(type == 0)
note = &partnote[k].kititem[item].adnote;
- else if(type == 1)
+ else
+ if(type == 1)
note = &partnote[k].kititem[item].subnote;
- else if(type == 2)
+ else
+ if(type == 2)
note = &partnote[k].kititem[item].padnote;
//Process if it exists
@@ -847,12 +922,11 @@ void Part::RunNote(unsigned int k)
*/
void Part::ComputePartSmps()
{
- for(unsigned nefx = 0; nefx < NUM_PART_EFX + 1; ++nefx) {
+ for(unsigned nefx = 0; nefx < NUM_PART_EFX + 1; ++nefx)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partfxinputl[nefx][i] = 0.0f;
partfxinputr[nefx][i] = 0.0f;
}
- }
for(unsigned k = 0; k < POLIPHONY; ++k) {
if(partnote[k].status == KEY_OFF)
@@ -867,12 +941,11 @@ void Part::ComputePartSmps()
for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx) {
if(!Pefxbypass[nefx]) {
partefx[nefx]->out(partfxinputl[nefx], partfxinputr[nefx]);
- if(Pefxroute[nefx] == 2) {
+ if(Pefxroute[nefx] == 2)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partfxinputl[nefx + 1][i] += partefx[nefx]->efxoutl[i];
partfxinputr[nefx + 1][i] += partefx[nefx]->efxoutr[i];
}
- }
}
int routeto = ((Pefxroute[nefx] == 0) ? nefx + 1 : NUM_PART_EFX);
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
@@ -898,7 +971,6 @@ void Part::ComputePartSmps()
killallnotes = 0;
for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx)
partefx[nefx]->cleanup();
-
}
ctl.updateportamento();
}
@@ -909,7 +981,8 @@ void Part::ComputePartSmps()
void Part::setPvolume(char Pvolume_)
{
Pvolume = Pvolume_;
- volume = dB2rap((Pvolume - 96.0f) / 96.0f * 40.0f) * ctl.expression.relvolume;
+ volume =
+ dB2rap((Pvolume - 96.0f) / 96.0f * 40.0f) * ctl.expression.relvolume;
}
void Part::setPpanning(char Ppanning_)
@@ -1073,7 +1146,7 @@ int Part::saveXML(const char *filename)
return result;
}
-int Part::loadXMLinstrument(const char *filename)/*{*/
+int Part::loadXMLinstrument(const char *filename) /*{*/
{
XMLwrapper *xml = new XMLwrapper();
if(xml->loadXMLfile(filename) < 0) {
@@ -1088,14 +1161,14 @@ int Part::loadXMLinstrument(const char *filename)/*{*/
delete (xml);
return 0;
-}/*}*/
+} /*}*/
-void Part::applyparameters(bool lockmutex)/*{*/
+void Part::applyparameters(bool lockmutex) /*{*/
{
for(int n = 0; n < NUM_KIT_ITEMS; ++n)
if((kit[n].padpars != NULL) && (kit[n].Ppadenabled != 0))
kit[n].padpars->applyparameters(lockmutex);
-}/*}*/
+} /*}*/
void Part::getfromXMLinstrument(XMLwrapper *xml)
{
@@ -1169,10 +1242,10 @@ void Part::getfromXMLinstrument(XMLwrapper *xml)
xml->exitbranch();
}
- Pefxroute[nefx] = xml->getpar("route",
- Pefxroute[nefx],
- 0,
- NUM_PART_EFX);
+ Pefxroute[nefx] = xml->getpar("route",
+ Pefxroute[nefx],
+ 0,
+ NUM_PART_EFX);
partefx[nefx]->setdryonly(Pefxroute[nefx] == 2);
Pefxbypass[nefx] = xml->getparbool("bypass", Pefxbypass[nefx]);
xml->exitbranch();
@@ -1188,20 +1261,20 @@ void Part::getfromXML(XMLwrapper *xml)
setPvolume(xml->getpar127("volume", Pvolume));
setPpanning(xml->getpar127("panning", Ppanning));
- Pminkey = xml->getpar127("min_key", Pminkey);
- Pmaxkey = xml->getpar127("max_key", Pmaxkey);
- Pkeyshift = xml->getpar127("key_shift", Pkeyshift);
- Prcvchn = xml->getpar127("rcv_chn", Prcvchn);
+ Pminkey = xml->getpar127("min_key", Pminkey);
+ Pmaxkey = xml->getpar127("max_key", Pmaxkey);
+ Pkeyshift = xml->getpar127("key_shift", Pkeyshift);
+ Prcvchn = xml->getpar127("rcv_chn", Prcvchn);
- Pvelsns = xml->getpar127("velocity_sensing", Pvelsns);
- Pveloffs = xml->getpar127("velocity_offset", Pveloffs);
+ Pvelsns = xml->getpar127("velocity_sensing", Pvelsns);
+ Pveloffs = xml->getpar127("velocity_offset", Pveloffs);
Pnoteon = xml->getparbool("note_on", Pnoteon);
Ppolymode = xml->getparbool("poly_mode", Ppolymode);
Plegatomode = xml->getparbool("legato_mode", Plegatomode); //older versions
if(!Plegatomode)
Plegatomode = xml->getpar127("legato_mode", Plegatomode);
- Pkeylimit = xml->getpar127("key_limit", Pkeylimit);
+ Pkeylimit = xml->getpar127("key_limit", Pkeylimit);
if(xml->enterbranch("INSTRUMENT")) {
@@ -1214,4 +1287,3 @@ void Part::getfromXML(XMLwrapper *xml)
xml->exitbranch();
}
}
-
diff --git a/src/Misc/Part.h b/src/Misc/Part.h
@@ -132,7 +132,7 @@ class Part
float *partoutr; //Right channel output of the part
float *partfxinputl[NUM_PART_EFX + 1], //Left and right signal that pass thru part effects;
- *partfxinputr[NUM_PART_EFX + 1]; //partfxinput l/r [NUM_PART_EFX] is for "no effect" buffer
+ *partfxinputr[NUM_PART_EFX + 1]; //partfxinput l/r [NUM_PART_EFX] is for "no effect" buffer
enum NoteStatus {
KEY_OFF, KEY_PLAYING, KEY_RELASED_AND_SUSTAINED, KEY_RELASED
@@ -165,10 +165,10 @@ class Part
int note; //if there is no note playing, the "note"=-1
int itemsplaying;
struct {
- SynthNote *adnote,
- *subnote,
- *padnote;
- int sendtoparteffect;
+ SynthNote *adnote,
+ *subnote,
+ *padnote;
+ int sendtoparteffect;
} kititem[NUM_KIT_ITEMS];
int time;
};
@@ -189,10 +189,9 @@ class Part
PartNotes partnote[POLIPHONY];
- float oldfreq; //this is used for portamento
+ float oldfreq; //this is used for portamento
Microtonal *microtonal;
FFTwrapper *fft;
};
#endif
-
diff --git a/src/Misc/Recorder.cpp b/src/Misc/Recorder.cpp
@@ -46,7 +46,7 @@ int Recorder::preparefile(std::string filename_, int overwrite)
return 1;
}
- OutMgr::getInstance().wave->newFile(new WavFile(filename_, SAMPLE_RATE, 2));
+ OutMgr::getInstance(). wave->newFile(new WavFile(filename_, SAMPLE_RATE, 2));
status = 1; //ready
@@ -61,15 +61,15 @@ void Recorder::start()
void Recorder::stop()
{
- OutMgr::getInstance().wave->Stop();
- OutMgr::getInstance().wave->destroyFile();
+ OutMgr::getInstance(). wave->Stop();
+ OutMgr::getInstance(). wave->destroyFile();
status = 0;
}
void Recorder::pause()
{
status = 0;
- OutMgr::getInstance().wave->Stop();
+ OutMgr::getInstance(). wave->Stop();
}
int Recorder::recording()
@@ -83,9 +83,8 @@ int Recorder::recording()
void Recorder::triggernow()
{
if(status == 2) {
- if(notetrigger!=1) {
+ if(notetrigger != 1)
OutMgr::getInstance().wave->Start();
- }
notetrigger = 1;
}
}
diff --git a/src/Misc/Recorder.h b/src/Misc/Recorder.h
@@ -52,4 +52,3 @@ class Recorder
};
#endif
-
diff --git a/src/Misc/Stereo.cpp b/src/Misc/Stereo.cpp
@@ -36,4 +36,3 @@ Stereo<T> &Stereo<T>::operator=(const Stereo<T> &nstr)
r = nstr.r;
return *this;
}
-
diff --git a/src/Misc/Stereo.h b/src/Misc/Stereo.h
@@ -22,8 +22,7 @@
#define STEREO_H
template<class T>
-struct Stereo
-{
+struct Stereo {
public:
Stereo(const T &left, const T &right);
@@ -39,4 +38,3 @@ struct Stereo
};
#include "Stereo.cpp"
#endif
-
diff --git a/src/Misc/Util.cpp b/src/Misc/Util.cpp
@@ -35,12 +35,12 @@
#include <sched.h>
-int SAMPLE_RATE = 44100;
-int SOUND_BUFFER_SIZE = 256;
-int OSCIL_SIZE = 1024;
-prng_t prng_state = 0x1234;
+int SAMPLE_RATE = 44100;
+int SOUND_BUFFER_SIZE = 256;
+int OSCIL_SIZE = 1024;
+prng_t prng_state = 0x1234;
-Config config;
+Config config;
float *denormalkillbuf;
@@ -61,12 +61,12 @@ float VelF(float velocity, unsigned char scaling)
* Get the detune in cents
*/
float getdetune(unsigned char type,
- unsigned short int coarsedetune,
- unsigned short int finedetune)
+ unsigned short int coarsedetune,
+ unsigned short int finedetune)
{
float det = 0.0f, octdet = 0.0f, cdet = 0.0f, findet = 0.0f;
//Get Octave
- int octave = coarsedetune / 1024;
+ int octave = coarsedetune / 1024;
if(octave >= 8)
octave -= 16;
octdet = octave * 1200.0f;
@@ -80,23 +80,24 @@ float getdetune(unsigned char type,
switch(type) {
// case 1: is used for the default (see below)
- case 2:
- cdet = fabs(cdetune * 10.0f);
- findet = fabs(fdetune / 8192.0f) * 10.0f;
- break;
- case 3:
- cdet = fabs(cdetune * 100);
- findet = powf(10, fabs(fdetune / 8192.0f) * 3.0f) / 10.0f - 0.1f;
- break;
- case 4:
- cdet = fabs(cdetune * 701.95500087f); //perfect fifth
- findet = (powf(2, fabs(fdetune / 8192.0f) * 12.0f) - 1.0f) / 4095 * 1200;
- break;
- //case ...: need to update N_DETUNE_TYPES, if you'll add more
- default:
- cdet = fabs(cdetune * 50.0f);
- findet = fabs(fdetune / 8192.0f) * 35.0f; //almost like "Paul's Sound Designer 2"
- break;
+ case 2:
+ cdet = fabs(cdetune * 10.0f);
+ findet = fabs(fdetune / 8192.0f) * 10.0f;
+ break;
+ case 3:
+ cdet = fabs(cdetune * 100);
+ findet = powf(10, fabs(fdetune / 8192.0f) * 3.0f) / 10.0f - 0.1f;
+ break;
+ case 4:
+ cdet = fabs(cdetune * 701.95500087f); //perfect fifth
+ findet =
+ (powf(2, fabs(fdetune / 8192.0f) * 12.0f) - 1.0f) / 4095 * 1200;
+ break;
+ //case ...: need to update N_DETUNE_TYPES, if you'll add more
+ default:
+ cdet = fabs(cdetune * 50.0f);
+ findet = fabs(fdetune / 8192.0f) * 35.0f; //almost like "Paul's Sound Designer 2"
+ break;
}
if(finedetune < 8192)
findet = -findet;
@@ -152,26 +153,25 @@ void invSignal(float *sig, size_t len)
//Some memory pools for short term buffer use
//(avoid the use of new in RT thread(s))
-struct pool_entry{
- bool free;
+struct pool_entry {
+ bool free;
float *dat;
};
typedef std::vector<pool_entry> pool_t;
-typedef pool_t::iterator pool_itr_t;
+typedef pool_t::iterator pool_itr_t;
pool_t pool;
float *getTmpBuffer()
{
- for(pool_itr_t itr = pool.begin(); itr != pool.end(); ++itr) {
+ for(pool_itr_t itr = pool.begin(); itr != pool.end(); ++itr)
if(itr->free) { //Use Pool
itr->free = false;
return itr->dat;
}
- }
pool_entry p; //Extend Pool
p.free = false;
- p.dat = new float[SOUND_BUFFER_SIZE];
+ p.dat = new float[SOUND_BUFFER_SIZE];
pool.push_back(p);
return p.dat;
@@ -179,23 +179,24 @@ float *getTmpBuffer()
void returnTmpBuffer(float *buf)
{
- for(pool_itr_t itr = pool.begin(); itr != pool.end(); ++itr) {
+ for(pool_itr_t itr = pool.begin(); itr != pool.end(); ++itr)
if(itr->dat == buf) { //Return to Pool
itr->free = true;
return;
}
- }
- fprintf(stderr,"ERROR: invalid buffer returned %s %d\n",__FILE__,__LINE__);
+ fprintf(stderr,
+ "ERROR: invalid buffer returned %s %d\n",
+ __FILE__,
+ __LINE__);
}
void clearTmpBuffers(void)
{
for(pool_itr_t itr = pool.begin(); itr != pool.end(); ++itr) {
if(!itr->free) //Warn about used buffers
- warn("Temporary buffer (%p) about to be freed may be in use", itr->dat);
+ warn("Temporary buffer (%p) about to be freed may be in use",
+ itr->dat);
delete [] itr->dat;
}
pool.clear();
}
-
-
diff --git a/src/Misc/Util.h b/src/Misc/Util.h
@@ -36,8 +36,8 @@ bool fileexists(const char *filename);
#define N_DETUNE_TYPES 4 //the number of detune types
extern float getdetune(unsigned char type,
- unsigned short int coarsedetune,
- unsigned short int finedetune);
+ unsigned short int coarsedetune,
+ unsigned short int finedetune);
/**Try to set current thread to realtime priority program priority
* \todo see if the right pid is being sent
@@ -81,39 +81,38 @@ T stringTo(const char *x)
return ans;
}
-template <class T>
+template<class T>
T limit(T val, T min, T max)
{
- return (val < min ? min : (val > max ? max : val));
-}
+ return (val<min ? min : (val> max ? max : val));
+ }
//Random number generator
-typedef uint32_t prng_t;
-extern prng_t prng_state;
+ typedef uint32_t prng_t;
+ extern prng_t prng_state;
// Portable Pseudo-Random Number Generator
-inline prng_t prng_r(prng_t &p)
-{
- return (p = p * 1103515245 + 12345);
-}
+ inline prng_t prng_r(prng_t & p)
+ {
+ return p = p * 1103515245 + 12345;
+ }
-inline prng_t prng(void)
-{
- return prng_r(prng_state)&0x7fffffff;
-}
+ inline prng_t prng(void)
+ {
+ return prng_r(prng_state) & 0x7fffffff;
+ }
-inline void sprng(prng_t p)
-{
- prng_state = p;
-}
+ inline void sprng(prng_t p)
+ {
+ prng_state = p;
+ }
/*
* The random generator (0.0f..1.0f)
*/
-# define INT32_MAX (2147483647)
+# define INT32_MAX (2147483647)
#define RND (prng() / (INT32_MAX * 1.0f))
#endif
-
diff --git a/src/Misc/WavFile.cpp b/src/Misc/WavFile.cpp
@@ -26,14 +26,14 @@ using namespace std;
WavFile::WavFile(string filename, int samplerate, int channels)
:sampleswritten(0), samplerate(samplerate), channels(channels),
- file(fopen(filename.c_str(), "w"))
+ file(fopen(filename.c_str(), "w"))
{
if(file) {
cout << "INFO: Making space for wave file header" << endl;
//making space for the header written at destruction
char tmp[44];
- memset(tmp, 0, 44*sizeof(char));
+ memset(tmp, 0, 44 * sizeof(char));
fwrite(tmp, 1, 44, file);
}
}
@@ -53,15 +53,15 @@ WavFile::~WavFile()
fwrite("WAVEfmt ", 8, 1, file);
chunksize = 16;
fwrite(&chunksize, 4, 1, file);
- unsigned short int formattag = 1; //uncompresed wave
+ unsigned short int formattag = 1; //uncompresed wave
fwrite(&formattag, 2, 1, file);
- unsigned short int nchannels = channels; //stereo
+ unsigned short int nchannels = channels; //stereo
fwrite(&nchannels, 2, 1, file);
- unsigned int samplerate_ = samplerate; //samplerate
+ unsigned int samplerate_ = samplerate; //samplerate
fwrite(&samplerate_, 4, 1, file);
- unsigned int bytespersec = samplerate * 2 * channels; //bytes/sec
+ unsigned int bytespersec = samplerate * 2 * channels; //bytes/sec
fwrite(&bytespersec, 4, 1, file);
- unsigned short int blockalign = 2 * channels; //2 channels * 16 bits/8
+ unsigned short int blockalign = 2 * channels; //2 channels * 16 bits/8
fwrite(&blockalign, 2, 1, file);
unsigned short int bitspersample = 16;
fwrite(&bitspersample, 2, 1, file);
@@ -95,4 +95,3 @@ void WavFile::writeMonoSamples(int nsmps, short int *smps)
sampleswritten += nsmps;
}
}
-
diff --git a/src/Misc/WavFile.h b/src/Misc/WavFile.h
@@ -42,4 +42,3 @@ class WavFile
FILE *file;
};
#endif
-
diff --git a/src/Misc/WaveShapeSmps.cpp b/src/Misc/WaveShapeSmps.cpp
@@ -28,162 +28,162 @@ void waveShapeSmps(int n,
unsigned char type,
unsigned char drive)
{
- int i;
+ int i;
float ws = drive / 127.0f;
float tmpv;
switch(type) {
- case 1:
- ws = powf(10, ws * ws * 3.0f) - 1.0f + 0.001f; //Arctangent
- for(i = 0; i < n; ++i)
- smps[i] = atanf(smps[i] * ws) / atanf(ws);
- break;
- case 2:
- ws = ws * ws * 32.0f + 0.0001f; //Asymmetric
- if(ws < 1.0f)
- tmpv = sinf(ws) + 0.1f;
- else
- tmpv = 1.1f;
- for(i = 0; i < n; ++i)
- smps[i] = sinf(smps[i] * (0.1f + ws - ws * smps[i])) / tmpv;
- ;
- break;
- case 3:
- ws = ws * ws * ws * 20.0f + 0.0001f; //Pow
- for(i = 0; i < n; ++i) {
- smps[i] *= ws;
- if(fabs(smps[i]) < 1.0f) {
- smps[i] = (smps[i] - powf(smps[i], 3.0f)) * 3.0f;
- if(ws < 1.0f)
- smps[i] /= ws;
- }
+ case 1:
+ ws = powf(10, ws * ws * 3.0f) - 1.0f + 0.001f; //Arctangent
+ for(i = 0; i < n; ++i)
+ smps[i] = atanf(smps[i] * ws) / atanf(ws);
+ break;
+ case 2:
+ ws = ws * ws * 32.0f + 0.0001f; //Asymmetric
+ if(ws < 1.0f)
+ tmpv = sinf(ws) + 0.1f;
else
- smps[i] = 0.0f;
- }
- break;
- case 4:
- ws = ws * ws * ws * 32.0f + 0.0001f; //Sine
- if(ws < 1.57f)
- tmpv = sinf(ws);
- else
- tmpv = 1.0f;
- for(i = 0; i < n; ++i)
- smps[i] = sinf(smps[i] * ws) / tmpv;
- break;
- case 5:
- ws = ws * ws + 0.000001f; //Quantisize
- for(i = 0; i < n; ++i)
- smps[i] = floor(smps[i] / ws + 0.5f) * ws;
- break;
- case 6:
- ws = ws * ws * ws * 32 + 0.0001f; //Zigzag
- if(ws < 1.0f)
- tmpv = sinf(ws);
- else
- tmpv = 1.0f;
- for(i = 0; i < n; ++i)
- smps[i] = asinf(sinf(smps[i] * ws)) / tmpv;
- break;
- case 7:
- ws = powf(2.0f, -ws * ws * 8.0f); //Limiter
- for(i = 0; i < n; ++i) {
- float tmp = smps[i];
- if(fabs(tmp) > ws) {
- if(tmp >= 0.0f)
- smps[i] = 1.0f;
+ tmpv = 1.1f;
+ for(i = 0; i < n; ++i)
+ smps[i] = sinf(smps[i] * (0.1f + ws - ws * smps[i])) / tmpv;
+ ;
+ break;
+ case 3:
+ ws = ws * ws * ws * 20.0f + 0.0001f; //Pow
+ for(i = 0; i < n; ++i) {
+ smps[i] *= ws;
+ if(fabs(smps[i]) < 1.0f) {
+ smps[i] = (smps[i] - powf(smps[i], 3.0f)) * 3.0f;
+ if(ws < 1.0f)
+ smps[i] /= ws;
+ }
else
- smps[i] = -1.0f;
+ smps[i] = 0.0f;
}
+ break;
+ case 4:
+ ws = ws * ws * ws * 32.0f + 0.0001f; //Sine
+ if(ws < 1.57f)
+ tmpv = sinf(ws);
else
- smps[i] /= ws;
- }
- break;
- case 8:
- ws = powf(2.0f, -ws * ws * 8.0f); //Upper Limiter
- for(i = 0; i < n; ++i) {
- float tmp = smps[i];
- if(tmp > ws)
- smps[i] = ws;
- smps[i] *= 2.0f;
- }
- break;
- case 9:
- ws = powf(2.0f, -ws * ws * 8.0f); //Lower Limiter
- for(i = 0; i < n; ++i) {
- float tmp = smps[i];
- if(tmp < -ws)
- smps[i] = -ws;
- smps[i] *= 2.0f;
- }
- break;
- case 10:
- ws = (powf(2.0f, ws * 6.0f) - 1.0f) / powf(2.0f, 6.0f); //Inverse Limiter
- for(i = 0; i < n; ++i) {
- float tmp = smps[i];
- if(fabs(tmp) > ws) {
- if(tmp >= 0.0f)
- smps[i] = tmp - ws;
+ tmpv = 1.0f;
+ for(i = 0; i < n; ++i)
+ smps[i] = sinf(smps[i] * ws) / tmpv;
+ break;
+ case 5:
+ ws = ws * ws + 0.000001f; //Quantisize
+ for(i = 0; i < n; ++i)
+ smps[i] = floor(smps[i] / ws + 0.5f) * ws;
+ break;
+ case 6:
+ ws = ws * ws * ws * 32 + 0.0001f; //Zigzag
+ if(ws < 1.0f)
+ tmpv = sinf(ws);
+ else
+ tmpv = 1.0f;
+ for(i = 0; i < n; ++i)
+ smps[i] = asinf(sinf(smps[i] * ws)) / tmpv;
+ break;
+ case 7:
+ ws = powf(2.0f, -ws * ws * 8.0f); //Limiter
+ for(i = 0; i < n; ++i) {
+ float tmp = smps[i];
+ if(fabs(tmp) > ws) {
+ if(tmp >= 0.0f)
+ smps[i] = 1.0f;
+ else
+ smps[i] = -1.0f;
+ }
else
- smps[i] = tmp + ws;
+ smps[i] /= ws;
}
+ break;
+ case 8:
+ ws = powf(2.0f, -ws * ws * 8.0f); //Upper Limiter
+ for(i = 0; i < n; ++i) {
+ float tmp = smps[i];
+ if(tmp > ws)
+ smps[i] = ws;
+ smps[i] *= 2.0f;
+ }
+ break;
+ case 9:
+ ws = powf(2.0f, -ws * ws * 8.0f); //Lower Limiter
+ for(i = 0; i < n; ++i) {
+ float tmp = smps[i];
+ if(tmp < -ws)
+ smps[i] = -ws;
+ smps[i] *= 2.0f;
+ }
+ break;
+ case 10:
+ ws = (powf(2.0f, ws * 6.0f) - 1.0f) / powf(2.0f, 6.0f); //Inverse Limiter
+ for(i = 0; i < n; ++i) {
+ float tmp = smps[i];
+ if(fabs(tmp) > ws) {
+ if(tmp >= 0.0f)
+ smps[i] = tmp - ws;
+ else
+ smps[i] = tmp + ws;
+ }
+ else
+ smps[i] = 0;
+ }
+ break;
+ case 11:
+ ws = powf(5, ws * ws * 1.0f) - 1.0f; //Clip
+ for(i = 0; i < n; ++i)
+ smps[i] = smps[i]
+ * (ws + 0.5f) * 0.9999f - floor(
+ 0.5f + smps[i] * (ws + 0.5f) * 0.9999f);
+ break;
+ case 12:
+ ws = ws * ws * ws * 30 + 0.001f; //Asym2
+ if(ws < 0.3f)
+ tmpv = ws;
else
- smps[i] = 0;
- }
- break;
- case 11:
- ws = powf(5, ws * ws * 1.0f) - 1.0f; //Clip
- for(i = 0; i < n; ++i)
- smps[i] = smps[i]
- * (ws + 0.5f) * 0.9999f - floor(
- 0.5f + smps[i] * (ws + 0.5f) * 0.9999f);
- break;
- case 12:
- ws = ws * ws * ws * 30 + 0.001f; //Asym2
- if(ws < 0.3f)
- tmpv = ws;
- else
- tmpv = 1.0f;
- for(i = 0; i < n; ++i) {
- float tmp = smps[i] * ws;
- if((tmp > -2.0f) && (tmp < 1.0f))
- smps[i] = tmp * (1.0f - tmp) * (tmp + 2.0f) / tmpv;
- else
- smps[i] = 0.0f;
- }
- break;
- case 13:
- ws = ws * ws * ws * 32.0f + 0.0001f; //Pow2
- if(ws < 1.0f)
- tmpv = ws * (1 + ws) / 2.0f;
- else
- tmpv = 1.0f;
- for(i = 0; i < n; ++i) {
- float tmp = smps[i] * ws;
- if((tmp > -1.0f) && (tmp < 1.618034f))
- smps[i] = tmp * (1.0f - tmp) / tmpv;
- else
- if(tmp > 0.0f)
- smps[i] = -1.0f;
+ tmpv = 1.0f;
+ for(i = 0; i < n; ++i) {
+ float tmp = smps[i] * ws;
+ if((tmp > -2.0f) && (tmp < 1.0f))
+ smps[i] = tmp * (1.0f - tmp) * (tmp + 2.0f) / tmpv;
+ else
+ smps[i] = 0.0f;
+ }
+ break;
+ case 13:
+ ws = ws * ws * ws * 32.0f + 0.0001f; //Pow2
+ if(ws < 1.0f)
+ tmpv = ws * (1 + ws) / 2.0f;
else
- smps[i] = -2.0f;
- }
- break;
- case 14:
- ws = powf(ws, 5.0f) * 80.0f + 0.0001f; //sigmoid
- if(ws > 10.0f)
- tmpv = 0.5f;
- else
- tmpv = 0.5f - 1.0f / (expf(ws) + 1.0f);
- for(i = 0; i < n; ++i) {
- float tmp = smps[i] * ws;
- if(tmp < -10.0f)
- tmp = -10.0f;
+ tmpv = 1.0f;
+ for(i = 0; i < n; ++i) {
+ float tmp = smps[i] * ws;
+ if((tmp > -1.0f) && (tmp < 1.618034f))
+ smps[i] = tmp * (1.0f - tmp) / tmpv;
+ else
+ if(tmp > 0.0f)
+ smps[i] = -1.0f;
+ else
+ smps[i] = -2.0f;
+ }
+ break;
+ case 14:
+ ws = powf(ws, 5.0f) * 80.0f + 0.0001f; //sigmoid
+ if(ws > 10.0f)
+ tmpv = 0.5f;
else
- if(tmp > 10.0f)
- tmp = 10.0f;
- tmp = 0.5f - 1.0f / (expf(tmp) + 1.0f);
- smps[i] = tmp / tmpv;
- }
- break;
+ tmpv = 0.5f - 1.0f / (expf(ws) + 1.0f);
+ for(i = 0; i < n; ++i) {
+ float tmp = smps[i] * ws;
+ if(tmp < -10.0f)
+ tmp = -10.0f;
+ else
+ if(tmp > 10.0f)
+ tmp = 10.0f;
+ tmp = 0.5f - 1.0f / (expf(tmp) + 1.0f);
+ smps[i] = tmp / tmpv;
+ }
+ break;
}
}
diff --git a/src/Misc/XMLwrapper.cpp b/src/Misc/XMLwrapper.cpp
@@ -102,8 +102,8 @@ XMLwrapper::XMLwrapper()
minimal = true;
- node = tree = mxmlNewElement(MXML_NO_PARENT,
- "?xml version=\"1.0f\" encoding=\"UTF-8\"?");
+ node = tree = mxmlNewElement(MXML_NO_PARENT,
+ "?xml version=\"1.0f\" encoding=\"UTF-8\"?");
/* for mxml 2.1f (and older)
tree=mxmlNewElement(MXML_NO_PARENT,"?xml");
mxmlElementSetAttr(tree,"version","1.0f");
@@ -315,7 +315,8 @@ int XMLwrapper::loadXMLfile(const string &filename)
if(xmldata == NULL)
return -1; //the file could not be loaded or uncompressed
- root = tree = mxmlLoadString(NULL, trimLeadingWhite(xmldata), MXML_OPAQUE_CALLBACK);
+ root = tree = mxmlLoadString(NULL, trimLeadingWhite(
+ xmldata), MXML_OPAQUE_CALLBACK);
delete[] xmldata;
@@ -384,7 +385,8 @@ bool XMLwrapper::putXMLdata(const char *xmldata)
if(xmldata == NULL)
return false;
- root = tree = mxmlLoadString(NULL, trimLeadingWhite(xmldata), MXML_OPAQUE_CALLBACK);
+ root = tree = mxmlLoadString(NULL, trimLeadingWhite(
+ xmldata), MXML_OPAQUE_CALLBACK);
if(tree == NULL)
return false;
@@ -578,9 +580,9 @@ float XMLwrapper::getparreal(const char *name, float defaultpar) const
}
float XMLwrapper::getparreal(const char *name,
- float defaultpar,
- float min,
- float max) const
+ float defaultpar,
+ float min,
+ float max) const
{
float result = getparreal(name, defaultpar);
@@ -619,4 +621,3 @@ mxml_node_t *XMLwrapper::addparams(const char *name, unsigned int params,
}
return element;
}
-
diff --git a/src/Misc/XMLwrapper.h b/src/Misc/XMLwrapper.h
@@ -205,9 +205,9 @@ class XMLwrapper
* @param max The maximum value
*/
float getparreal(const char *name,
- float defaultpar,
- float min,
- float max) const;
+ float defaultpar,
+ float min,
+ float max) const;
bool minimal; /**<false if all parameters will be stored (used only for clipboard)*/
@@ -269,4 +269,3 @@ class XMLwrapper
};
#endif
-
diff --git a/src/Nio/AlsaEngine.cpp b/src/Nio/AlsaEngine.cpp
@@ -32,14 +32,13 @@ using namespace std;
AlsaEngine::AlsaEngine()
:AudioOut()
{
- audio.buffer = new short[SOUND_BUFFER_SIZE*2];
+ audio.buffer = new short[SOUND_BUFFER_SIZE * 2];
name = "ALSA";
audio.handle = NULL;
- midi.handle = NULL;
- midi.alsaId = -1;
+ midi.handle = NULL;
+ midi.alsaId = -1;
midi.pThread = 0;
-
}
AlsaEngine::~AlsaEngine()
@@ -50,7 +49,7 @@ AlsaEngine::~AlsaEngine()
void *AlsaEngine::_AudioThread(void *arg)
{
- return (static_cast<AlsaEngine*>(arg))->AudioThread();
+ return (static_cast<AlsaEngine *>(arg))->AudioThread();
}
void *AlsaEngine::AudioThread()
@@ -101,7 +100,7 @@ bool AlsaEngine::getAudioEn() const
void *AlsaEngine::_MidiThread(void *arg)
{
- return static_cast<AlsaEngine*>(arg)->MidiThread();
+ return static_cast<AlsaEngine *>(arg)->MidiThread();
}
@@ -110,21 +109,18 @@ void *AlsaEngine::MidiThread(void)
snd_seq_event_t *event;
MidiEvent ev;
set_realtime();
- while(snd_seq_event_input(midi.handle, &event) > 0)
- {
+ while(snd_seq_event_input(midi.handle, &event) > 0) {
//ensure ev is empty
ev.channel = 0;
- ev.num = 0;
- ev.value = 0;
- ev.type = 0;
+ ev.num = 0;
+ ev.value = 0;
+ ev.type = 0;
- if (!event)
+ if(!event)
continue;
- switch (event->type)
- {
+ switch(event->type) {
case SND_SEQ_EVENT_NOTEON:
- if (event->data.note.note)
- {
+ if(event->data.note.note) {
ev.type = M_NOTE;
ev.channel = event->data.note.channel;
ev.num = event->data.note.note;
@@ -166,12 +162,12 @@ void *AlsaEngine::MidiThread(void)
break;
case SND_SEQ_EVENT_PORT_SUBSCRIBED: // ports connected
- if (true)
+ if(true)
cout << "Info, alsa midi port connected" << endl;
break;
case SND_SEQ_EVENT_PORT_UNSUBSCRIBED: // ports disconnected
- if (true)
+ if(true)
cout << "Info, alsa midi port disconnected" << endl;
break;
@@ -180,9 +176,9 @@ void *AlsaEngine::MidiThread(void)
break;
default:
- if (true)
+ if(true)
cout << "Info, other non-handled midi event, type: "
- << (int)event->type << endl;
+ << (int)event->type << endl;
break;
}
snd_seq_free_event(event);
@@ -226,7 +222,7 @@ void AlsaEngine::stopMidi()
return;
snd_seq_t *handle = midi.handle;
- if (NULL != midi.handle && midi.pThread)
+ if((NULL != midi.handle) && midi.pThread)
pthread_cancel(midi.pThread);
midi.handle = NULL;
if(handle)
@@ -237,11 +233,10 @@ short *AlsaEngine::interleave(const Stereo<float *> &smps)
{
/**\todo TODO fix repeated allocation*/
short *shortInterleaved = audio.buffer;
- memset(shortInterleaved,0,bufferSize*2*sizeof(short));
- int idx = 0;//possible off by one error here
+ memset(shortInterleaved, 0, bufferSize * 2 * sizeof(short));
+ int idx = 0; //possible off by one error here
double scaled;
- for (int frame = 0; frame < bufferSize; ++frame)
- { // with a nod to libsamplerate ...
+ for(int frame = 0; frame < bufferSize; ++frame) { // with a nod to libsamplerate ...
scaled = smps.l[frame] * (8.0f * 0x10000000);
shortInterleaved[idx++] = (short int)(lrint(scaled) >> 16);
scaled = smps.r[frame] * (8.0f * 0x10000000);
@@ -257,10 +252,10 @@ bool AlsaEngine::openAudio()
int rc = 0;
/* Open PCM device for playback. */
- audio.handle=NULL;
+ audio.handle = NULL;
rc = snd_pcm_open(&audio.handle, "hw:0",
- SND_PCM_STREAM_PLAYBACK, 0);
- if (rc < 0) {
+ SND_PCM_STREAM_PLAYBACK, 0);
+ if(rc < 0) {
fprintf(stderr,
"unable to open pcm device: %s\n",
snd_strerror(rc));
@@ -277,30 +272,30 @@ bool AlsaEngine::openAudio()
/* Interleaved mode */
snd_pcm_hw_params_set_access(audio.handle, audio.params,
- SND_PCM_ACCESS_RW_INTERLEAVED);
+ SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(audio.handle, audio.params,
- SND_PCM_FORMAT_S16_LE);
+ SND_PCM_FORMAT_S16_LE);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(audio.handle, audio.params, 2);
audio.sampleRate = SAMPLE_RATE;
snd_pcm_hw_params_set_rate_near(audio.handle, audio.params,
- &audio.sampleRate, NULL);
+ &audio.sampleRate, NULL);
audio.frames = 512;
snd_pcm_hw_params_set_period_size_near(audio.handle,
- audio.params, &audio.frames, NULL);
+ audio.params, &audio.frames, NULL);
audio.periods = 4;
snd_pcm_hw_params_set_periods_near(audio.handle,
- audio.params, &audio.periods, NULL);
+ audio.params, &audio.periods, NULL);
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(audio.handle, audio.params);
- if (rc < 0) {
+ if(rc < 0) {
fprintf(stderr,
"unable to set hw parameters: %s\n",
snd_strerror(rc));
@@ -309,7 +304,9 @@ bool AlsaEngine::openAudio()
/* Set buffer size (in frames). The resulting latency is given by */
/* latency = periodsize * periods / (rate * bytes_per_frame) */
- snd_pcm_hw_params_set_buffer_size(audio.handle, audio.params, SOUND_BUFFER_SIZE);
+ snd_pcm_hw_params_set_buffer_size(audio.handle,
+ audio.params,
+ SOUND_BUFFER_SIZE);
//snd_pcm_hw_params_get_period_size(audio.params, &audio.frames, NULL);
//snd_pcm_hw_params_get_period_time(audio.params, &val, NULL);
@@ -332,21 +329,23 @@ void AlsaEngine::stopAudio()
pthread_join(audio.pThread, NULL);
snd_pcm_drain(handle);
if(snd_pcm_close(handle))
- cout << "Error: in snd_pcm_close " << __LINE__ << ' ' << __FILE__ << endl;
+ cout << "Error: in snd_pcm_close " << __LINE__ << ' ' << __FILE__
+ << endl;
}
void *AlsaEngine::processAudio()
{
- while (audio.handle) {
+ while(audio.handle) {
audio.buffer = interleave(getNext());
snd_pcm_t *handle = audio.handle;
int rc = snd_pcm_writei(handle, audio.buffer, SOUND_BUFFER_SIZE);
- if (rc == -EPIPE) {
+ if(rc == -EPIPE) {
/* EPIPE means underrun */
cerr << "underrun occurred" << endl;
snd_pcm_prepare(handle);
}
- else if (rc < 0)
+ else
+ if(rc < 0)
cerr << "error from writei: " << snd_strerror(rc) << endl;
}
return NULL;
diff --git a/src/Nio/AlsaEngine.h b/src/Nio/AlsaEngine.h
@@ -31,20 +31,20 @@
#include "OutMgr.h"
#include "../Misc/Stereo.h"
-class AlsaEngine : public AudioOut, MidiIn
+class AlsaEngine:public AudioOut, MidiIn
{
public:
AlsaEngine();
~AlsaEngine();
-
+
bool Start();
void Stop();
-
+
void setAudioEn(bool nval);
bool getAudioEn() const;
void setMidiEn(bool nval);
bool getMidiEn() const;
-
+
protected:
void *AudioThread();
static void *_AudioThread(void *arg);
@@ -60,20 +60,20 @@ class AlsaEngine : public AudioOut, MidiIn
short *interleave(const Stereo<float *> &smps);
struct {
- std::string device;
- snd_seq_t *handle;
- int alsaId;
- pthread_t pThread;
+ std::string device;
+ snd_seq_t *handle;
+ int alsaId;
+ pthread_t pThread;
} midi;
struct {
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
- unsigned int sampleRate;
+ unsigned int sampleRate;
snd_pcm_uframes_t frames;
- unsigned int periods;
- short *buffer;
- pthread_t pThread;
+ unsigned int periods;
+ short *buffer;
+ pthread_t pThread;
} audio;
void *processAudio();
diff --git a/src/Nio/AudioOut.cpp b/src/Nio/AudioOut.cpp
@@ -31,12 +31,11 @@ using namespace std;
#include "AudioOut.h"
AudioOut::AudioOut()
- :samplerate(SAMPLE_RATE),bufferSize(SOUND_BUFFER_SIZE)
+ :samplerate(SAMPLE_RATE), bufferSize(SOUND_BUFFER_SIZE)
{}
AudioOut::~AudioOut()
-{
-}
+{}
void AudioOut::setSamplerate(int _samplerate)
{
diff --git a/src/Nio/AudioOut.h b/src/Nio/AudioOut.h
@@ -27,7 +27,7 @@
#include "../globals.h"
#include "Engine.h"
-class AudioOut : public virtual Engine
+class AudioOut:public virtual Engine
{
public:
AudioOut();
@@ -46,8 +46,8 @@ class AudioOut : public virtual Engine
void bufferingSize(int nBuffering);
int bufferingSize();
- virtual void setAudioEn(bool nval)=0;
- virtual bool getAudioEn() const=0;
+ virtual void setAudioEn(bool nval) = 0;
+ virtual bool getAudioEn() const = 0;
protected:
/**Get the next sample for output.
@@ -59,4 +59,3 @@ class AudioOut : public virtual Engine
};
#endif
-
diff --git a/src/Nio/Engine.cpp b/src/Nio/Engine.cpp
@@ -22,8 +22,7 @@
#include "Engine.h"
Engine::Engine()
-{};
+{}
Engine::~Engine()
-{};
-
+{}
diff --git a/src/Nio/Engine.h b/src/Nio/Engine.h
@@ -32,9 +32,9 @@ class Engine
/**Start the Driver with all capabilities
* @return true on success*/
- virtual bool Start()=0;
+ virtual bool Start() = 0;
/**Completely stop the Driver*/
- virtual void Stop()=0;
+ virtual void Stop() = 0;
std::string name;
};
diff --git a/src/Nio/EngineMgr.cpp b/src/Nio/EngineMgr.cpp
@@ -64,33 +64,30 @@ EngineMgr::EngineMgr()
defaultOut = dynamic_cast<AudioOut *>(defaultEng);
- defaultIn = dynamic_cast<MidiIn *>(defaultEng);
-};
+ defaultIn = dynamic_cast<MidiIn *>(defaultEng);
+}
EngineMgr::~EngineMgr()
{
- for(list<Engine*>::iterator itr = engines.begin();
- itr != engines.end(); ++itr) {
- delete *itr;
- }
+ for(list<Engine *>::iterator itr = engines.begin();
+ itr != engines.end(); ++itr)
+ delete *itr;
}
Engine *EngineMgr::getEng(string name)
{
transform(name.begin(), name.end(), name.begin(), ::toupper);
- for(list<Engine*>::iterator itr = engines.begin();
- itr != engines.end(); ++itr) {
- if((*itr)->name == name) {
+ for(list<Engine *>::iterator itr = engines.begin();
+ itr != engines.end(); ++itr)
+ if((*itr)->name == name)
return *itr;
- }
- }
return NULL;
}
bool EngineMgr::start()
{
bool expected = true;
- if(!(defaultOut&&defaultIn)) {
+ if(!(defaultOut && defaultIn)) {
cerr << "ERROR: It looks like someone broke the Nio Output\n"
<< " Attempting to recover by defaulting to the\n"
<< " Null Engine." << endl;
@@ -98,32 +95,31 @@ bool EngineMgr::start()
defaultIn = dynamic_cast<MidiIn *>(getEng("NULL"));
}
- OutMgr::getInstance().currentOut = defaultOut;
- InMgr::getInstance().current = defaultIn;
+ OutMgr::getInstance(). currentOut = defaultOut;
+ InMgr::getInstance(). current = defaultIn;
//open up the default output(s)
cout << "Starting Audio: " << defaultOut->name << endl;
defaultOut->setAudioEn(true);
- if(defaultOut->getAudioEn()) {
+ if(defaultOut->getAudioEn())
cout << "Audio Started" << endl;
- }
- else {
+ else {
expected = false;
cerr << "ERROR: The default audio output failed to open!" << endl;
- OutMgr::getInstance().currentOut = dynamic_cast<AudioOut *>(getEng("NULL"));
- OutMgr::getInstance().currentOut->setAudioEn(true);
+ OutMgr::getInstance(). currentOut =
+ dynamic_cast<AudioOut *>(getEng("NULL"));
+ OutMgr::getInstance(). currentOut->setAudioEn(true);
}
cout << "Starting MIDI: " << defaultIn->name << endl;
defaultIn->setMidiEn(true);
- if(defaultIn->getMidiEn()) {
+ if(defaultIn->getMidiEn())
cout << "MIDI Started" << endl;
- }
else { //recover
expected = false;
cerr << "ERROR: The default MIDI input failed to open!" << endl;
- InMgr::getInstance().current = dynamic_cast<MidiIn *>(getEng("NULL"));
- InMgr::getInstance().current->setMidiEn(true);
+ InMgr::getInstance(). current = dynamic_cast<MidiIn *>(getEng("NULL"));
+ InMgr::getInstance(). current->setMidiEn(true);
}
//Show if expected drivers were booted
@@ -132,29 +128,27 @@ bool EngineMgr::start()
void EngineMgr::stop()
{
- for(list<Engine*>::iterator itr = engines.begin();
- itr != engines.end(); ++itr)
+ for(list<Engine *>::iterator itr = engines.begin();
+ itr != engines.end(); ++itr)
(*itr)->Stop();
}
bool EngineMgr::setInDefault(string name)
{
- MidiIn *chosen;
- if((chosen = dynamic_cast<MidiIn *>(getEng(name)))){ //got the input
- defaultIn = chosen;
- return true;
- }
- return false;
+ MidiIn *chosen;
+ if((chosen = dynamic_cast<MidiIn *>(getEng(name)))) { //got the input
+ defaultIn = chosen;
+ return true;
+ }
+ return false;
}
bool EngineMgr::setOutDefault(string name)
{
- AudioOut *chosen;
- if((chosen = dynamic_cast<AudioOut *>(getEng(name)))){ //got the output
- defaultOut = chosen;
- return true;
- }
- return false;
+ AudioOut *chosen;
+ if((chosen = dynamic_cast<AudioOut *>(getEng(name)))) { //got the output
+ defaultOut = chosen;
+ return true;
+ }
+ return false;
}
-
-
diff --git a/src/Nio/EngineMgr.h b/src/Nio/EngineMgr.h
@@ -41,4 +41,3 @@ class EngineMgr
EngineMgr();
};
#endif
-
diff --git a/src/Nio/InMgr.cpp b/src/Nio/InMgr.cpp
@@ -6,21 +6,21 @@
using namespace std;
-ostream &operator<<(ostream &out, const MidiEvent& ev)
+ostream &operator<<(ostream &out, const MidiEvent &ev)
{
if(ev.type == M_NOTE)
- out << "MidiNote: note(" << ev.num << ")\n"
- << " channel(" << ev.channel << ")\n"
- << " velocity(" << ev.value << ")";
+ out << "MidiNote: note(" << ev.num << ")\n"
+ << " channel(" << ev.channel << ")\n"
+ << " velocity(" << ev.value << ")";
else
- out << "MidiCtl: controller(" << ev.num << ")\n"
- << " channel(" << ev.channel << ")\n"
- << " value(" << ev.value << ")";
+ out << "MidiCtl: controller(" << ev.num << ")\n"
+ << " channel(" << ev.channel << ")\n"
+ << " value(" << ev.value << ")";
return out;
}
MidiEvent::MidiEvent()
- :channel(0),type(0),num(0),value(0)
+ :channel(0), type(0), num(0), value(0)
{}
InMgr &InMgr::getInstance()
@@ -106,4 +106,3 @@ MidiIn *InMgr::getIn(string name)
EngineMgr &eng = EngineMgr::getInstance();
return dynamic_cast<MidiIn *>(eng.getEng(name));
}
-
diff --git a/src/Nio/InMgr.h b/src/Nio/InMgr.h
@@ -5,13 +5,12 @@
#include <semaphore.h>
#include "SafeQueue.h"
-enum midi_type{
+enum midi_type {
M_NOTE = 1,
M_CONTROLLER = 2
}; //type=1 for note, type=2 for controller
-struct MidiEvent
-{
+struct MidiEvent {
MidiEvent();
int channel; //the midi channel for the event
int type; //type=1 for note, type=2 for controller
@@ -41,11 +40,10 @@ class InMgr
class MidiIn *getIn(std::string name);
SafeQueue<MidiEvent> queue;
sem_t work;
- class MidiIn *current;
+ class MidiIn * current;
/**the link to the rest of zyn*/
- class Master &master;
+ class Master & master;
};
#endif
-
diff --git a/src/Nio/JackEngine.cpp b/src/Nio/JackEngine.cpp
@@ -37,10 +37,9 @@ JackEngine::JackEngine()
{
name = "JACK";
audio.jackSamplerate = 0;
- audio.jackNframes = 0;
- for (int i = 0; i < 2; ++i)
- {
- audio.ports[i] = NULL;
+ audio.jackNframes = 0;
+ for(int i = 0; i < 2; ++i) {
+ audio.ports[i] = NULL;
audio.portBuffs[i] = NULL;
}
midi.inport = NULL;
@@ -54,24 +53,26 @@ bool JackEngine::connectServer(string server)
string clientname = "zynaddsubfx";
- string postfix = Nio::getInstance().getPostfix();
+ string postfix = Nio::getInstance().getPostfix();
if(!postfix.empty())
clientname += "_" + postfix;
jack_status_t jackstatus;
bool use_server_name = server.size() && server.compare("default") != 0;
jack_options_t jopts = (jack_options_t)
- (((use_server_name) ? JackServerName : JackNullOption)
- | ((autostart_jack) ? JackNullOption : JackNoStartServer));
- if (use_server_name)
+ (((use_server_name) ? JackServerName :
+ JackNullOption)
+ | ((autostart_jack) ? JackNullOption :
+ JackNoStartServer));
+ if(use_server_name)
jackClient = jack_client_open(clientname.c_str(), jopts, &jackstatus,
- server.c_str());
+ server.c_str());
else
jackClient = jack_client_open(clientname.c_str(), jopts, &jackstatus);
- if (NULL != jackClient)
+ if(NULL != jackClient)
return true;
else
cerr << "Error, failed to open jack client on server: " << server
- << " status " << jackstatus << endl;
+ << " status " << jackstatus << endl;
return false;
return true;
@@ -80,24 +81,21 @@ bool JackEngine::connectServer(string server)
bool JackEngine::connectJack()
{
connectServer("");
- if (NULL != jackClient)
- {
+ if(NULL != jackClient) {
setBufferSize(jack_get_buffer_size(jackClient));
int chk;
jack_set_error_function(_errorCallback);
jack_set_info_function(_infoCallback);
if(jack_set_buffer_size_callback(jackClient, _bufferSizeCallback, this))
cerr << "Error setting the bufferSize callback" << endl;
- if ((chk = jack_set_xrun_callback(jackClient, _xrunCallback, this)))
+ if((chk = jack_set_xrun_callback(jackClient, _xrunCallback, this)))
cerr << "Error setting jack xrun callback" << endl;
- if (jack_set_process_callback(jackClient, _processCallback, this))
- {
+ if(jack_set_process_callback(jackClient, _processCallback, this)) {
cerr << "Error, JackEngine failed to set process callback" << endl;
return false;
}
- if (jack_activate(jackClient))
- {
- cerr << "Error, failed to activate jack client" << endl;;
+ if(jack_activate(jackClient)) {
+ cerr << "Error, failed to activate jack client" << endl;
return false;
}
@@ -164,31 +162,38 @@ bool JackEngine::openAudio()
const char *portnames[] = { "out_1", "out_2" };
- for (int port = 0; port < 2; ++port)
- {
- audio.ports[port] = jack_port_register(jackClient, portnames[port],
- JACK_DEFAULT_AUDIO_TYPE,
- JackPortIsOutput | JackPortIsTerminal, 0);
- }
- if (NULL != audio.ports[0] && NULL != audio.ports[1])
- {
+ for(int port = 0; port < 2; ++port)
+ audio.ports[port] = jack_port_register(
+ jackClient,
+ portnames[port],
+ JACK_DEFAULT_AUDIO_TYPE,
+ JackPortIsOutput
+ | JackPortIsTerminal,
+ 0);
+ if((NULL != audio.ports[0]) && (NULL != audio.ports[1])) {
audio.jackSamplerate = jack_get_sample_rate(jackClient);
- audio.jackNframes = jack_get_buffer_size(jackClient);
+ audio.jackNframes = jack_get_buffer_size(jackClient);
//Attempt to autoConnect when specified
- if(Nio::getInstance().autoConnect)
- {
- const char **outPorts = jack_get_ports(jackClient, NULL, NULL,
- JackPortIsPhysical|JackPortIsInput);
+ if(Nio::getInstance().autoConnect) {
+ const char **outPorts = jack_get_ports(
+ jackClient,
+ NULL,
+ NULL,
+ JackPortIsPhysical
+ | JackPortIsInput);
if(outPorts != NULL) {
//Verify that stereo is available
assert(outPorts[0]);
assert(outPorts[1]);
//Connect to physical outputs
- jack_connect(jackClient, jack_port_name(audio.ports[0]), outPorts[0]);
- jack_connect(jackClient, jack_port_name(audio.ports[1]), outPorts[1]);
- } else
+ jack_connect(jackClient, jack_port_name(
+ audio.ports[0]), outPorts[0]);
+ jack_connect(jackClient, jack_port_name(
+ audio.ports[1]), outPorts[1]);
+ }
+ else
cerr << "Warning, No outputs to autoconnect to" << endl;
}
return true;
@@ -200,11 +205,10 @@ bool JackEngine::openAudio()
void JackEngine::stopAudio()
{
- for (int i = 0; i < 2; ++i)
- {
+ for(int i = 0; i < 2; ++i) {
jack_port_t *port = audio.ports[i];
audio.ports[i] = NULL;
- if (NULL != port)
+ if(NULL != port)
jack_port_unregister(jackClient, port);
}
if(!getMidiEn())
@@ -238,7 +242,7 @@ void JackEngine::stopMidi()
int JackEngine::clientId()
{
- if (NULL != jackClient)
+ if(NULL != jackClient)
return (long)jack_client_thread_id(jackClient);
else
return -1;
@@ -246,7 +250,7 @@ int JackEngine::clientId()
string JackEngine::clientName()
{
- if (NULL != jackClient)
+ if(NULL != jackClient)
return string(jack_get_client_name(jackClient));
else
cerr << "Error, clientName() with null jackClient" << endl;
@@ -255,30 +259,28 @@ string JackEngine::clientName()
int JackEngine::_processCallback(jack_nframes_t nframes, void *arg)
{
- return static_cast<JackEngine*>(arg)->processCallback(nframes);
+ return static_cast<JackEngine *>(arg)->processCallback(nframes);
}
int JackEngine::processCallback(jack_nframes_t nframes)
{
bool okaudio = true;
- if (NULL != audio.ports[0] && NULL != audio.ports[1])
+ if((NULL != audio.ports[0]) && (NULL != audio.ports[1]))
okaudio = processAudio(nframes);
- if (okaudio)
- handleMidi(nframes);
+ if(okaudio)
+ handleMidi(nframes);
return okaudio ? 0 : -1;
}
bool JackEngine::processAudio(jack_nframes_t nframes)
{
- for (int port = 0; port < 2; ++port)
- {
+ for(int port = 0; port < 2; ++port) {
audio.portBuffs[port] =
- (jsample_t*)jack_port_get_buffer(audio.ports[port], nframes);
- if (NULL == audio.portBuffs[port])
- {
+ (jsample_t *)jack_port_get_buffer(audio.ports[port], nframes);
+ if(NULL == audio.portBuffs[port]) {
cerr << "Error, failed to get jack audio port buffer: "
- << port << endl;
+ << port << endl;
return false;
}
}
@@ -286,10 +288,9 @@ bool JackEngine::processAudio(jack_nframes_t nframes)
Stereo<float *> smp = getNext();
//Assumes size of smp.l == nframes
- memcpy(audio.portBuffs[0], smp.l, bufferSize*sizeof(float));
- memcpy(audio.portBuffs[1], smp.r, bufferSize*sizeof(float));
+ memcpy(audio.portBuffs[0], smp.l, bufferSize * sizeof(float));
+ memcpy(audio.portBuffs[1], smp.r, bufferSize * sizeof(float));
return true;
-
}
int JackEngine::_xrunCallback(void * /*/arg*/)
@@ -310,7 +311,7 @@ void JackEngine::_infoCallback(const char *msg)
int JackEngine::_bufferSizeCallback(jack_nframes_t nframes, void *arg)
{
- return static_cast<JackEngine*>(arg)->bufferSizeCallback(nframes);
+ return static_cast<JackEngine *>(arg)->bufferSizeCallback(nframes);
}
int JackEngine::bufferSizeCallback(jack_nframes_t nframes)
@@ -331,7 +332,7 @@ void JackEngine::handleMidi(unsigned long frames)
unsigned char type;
while(jack_midi_event_get(&jack_midi_event, midi_buf,
- event_index++) == 0) {
+ event_index++) == 0) {
MidiEvent ev;
midi_data = jack_midi_event.buffer;
type = midi_data[0] & 0xF0;
@@ -339,30 +340,30 @@ void JackEngine::handleMidi(unsigned long frames)
switch(type) {
case 0x80: /* note-off */
- ev.type = M_NOTE;
- ev.num = midi_data[1];
- ev.value = 0;
+ ev.type = M_NOTE;
+ ev.num = midi_data[1];
+ ev.value = 0;
InMgr::getInstance().putEvent(ev);
break;
case 0x90: /* note-on */
- ev.type = M_NOTE;
- ev.num = midi_data[1];
- ev.value = midi_data[2];
+ ev.type = M_NOTE;
+ ev.num = midi_data[1];
+ ev.value = midi_data[2];
InMgr::getInstance().putEvent(ev);
break;
case 0xB0: /* controller */
- ev.type = M_CONTROLLER;
- ev.num = midi_data[1];
- ev.value = midi_data[2];
+ ev.type = M_CONTROLLER;
+ ev.num = midi_data[1];
+ ev.value = midi_data[2];
InMgr::getInstance().putEvent(ev);
break;
case 0xE0: /* pitch bend */
- ev.type = M_CONTROLLER;
- ev.num = C_pitchwheel;
- ev.value = ((midi_data[2] << 7) | midi_data[1]) - 8192;
+ ev.type = M_CONTROLLER;
+ ev.num = C_pitchwheel;
+ ev.value = ((midi_data[2] << 7) | midi_data[1]) - 8192;
InMgr::getInstance().putEvent(ev);
break;
@@ -370,4 +371,3 @@ void JackEngine::handleMidi(unsigned long frames)
}
}
}
-
diff --git a/src/Nio/JackEngine.h b/src/Nio/JackEngine.h
@@ -31,11 +31,11 @@
typedef jack_default_audio_sample_t jsample_t;
-class JackEngine : public AudioOut, MidiIn
+class JackEngine:public AudioOut, MidiIn
{
public:
JackEngine();
- ~JackEngine() { };
+ ~JackEngine() { }
bool Start();
void Stop();
@@ -46,8 +46,8 @@ class JackEngine : public AudioOut, MidiIn
void setAudioEn(bool nval);
bool getAudioEn() const;
- unsigned int getSamplerate() { return audio.jackSamplerate; };
- unsigned int getBuffersize() { return audio.jackNframes; };
+ unsigned int getSamplerate() { return audio.jackSamplerate; }
+ unsigned int getBuffersize() { return audio.jackNframes; }
std::string clientName();
int clientId();
@@ -72,14 +72,14 @@ class JackEngine : public AudioOut, MidiIn
bool openMidi();
void stopMidi();
- jack_client_t *jackClient;
- struct audio{
- unsigned int jackSamplerate;
- unsigned int jackNframes;
- jack_port_t *ports[2];
- jsample_t *portBuffs[2];
+ jack_client_t *jackClient;
+ struct audio {
+ unsigned int jackSamplerate;
+ unsigned int jackNframes;
+ jack_port_t *ports[2];
+ jsample_t *portBuffs[2];
} audio;
- struct midi{
+ struct midi {
jack_port_t *inport;
} midi;
diff --git a/src/Nio/MidiIn.cpp b/src/Nio/MidiIn.cpp
@@ -24,12 +24,13 @@
#include "../globals.h"
#include "InMgr.h"
-void MidiIn::midiProcess(unsigned char head, unsigned char num, unsigned char value)
+void MidiIn::midiProcess(unsigned char head,
+ unsigned char num,
+ unsigned char value)
{
- MidiEvent ev;
+ MidiEvent ev;
unsigned char chan = head & 0x0f;
- switch(head & 0xf0)
- {
+ switch(head & 0xf0) {
case 0x80: //Note Off
ev.type = M_NOTE;
ev.channel = chan;
diff --git a/src/Nio/MidiIn.h b/src/Nio/MidiIn.h
@@ -28,15 +28,16 @@
#include "Engine.h"
/**This class is inherited by all the Midi input classes*/
-class MidiIn : public virtual Engine
+class MidiIn:public virtual Engine
{
public:
/**Enables or disables driver based upon value*/
- virtual void setMidiEn(bool nval)=0;
+ virtual void setMidiEn(bool nval) = 0;
/**Returns if driver is initialized*/
- virtual bool getMidiEn() const=0;
- static void midiProcess(unsigned char head, unsigned char num, unsigned char value);
+ virtual bool getMidiEn() const = 0;
+ static void midiProcess(unsigned char head,
+ unsigned char num,
+ unsigned char value);
};
#endif
-
diff --git a/src/Nio/Nio.cpp b/src/Nio/Nio.cpp
@@ -14,11 +14,11 @@ Nio &Nio::getInstance()
}
Nio::Nio()
-:autoConnect(false),
- in(InMgr::getInstance()),//Enable input wrapper
- out(OutMgr::getInstance()),//Initialize the Output Systems
- eng(EngineMgr::getInstance()),//Initialize The Engines
- postfix("")//no default postfix
+ :autoConnect(false),
+ in(InMgr::getInstance()), //Enable input wrapper
+ out(OutMgr::getInstance()), //Initialize the Output Systems
+ eng(EngineMgr::getInstance()), //Initialize The Engines
+ postfix("") //no default postfix
{}
Nio::~Nio()
@@ -28,14 +28,14 @@ Nio::~Nio()
bool Nio::start()
{
- return eng.start();//Drivers start your engines!
+ return eng.start(); //Drivers start your engines!
}
void Nio::stop()
{
eng.stop();
}
-
+
int Nio::setDefaultSource(string name)
{
if(name.empty())
@@ -47,27 +47,26 @@ int Nio::setDefaultSource(string name)
}
return 0;
}
-
+
int Nio::setDefaultSink(string name)
{
if(name.empty())
return 0;
- if(!eng.setOutDefault(name)) {
+ if(!eng.setOutDefault(name))
cerr << "There is no output for " << name << endl;
- }
return 0;
}
bool Nio::setSource(string name)
{
- return in.setSource(name);
+ return in.setSource(name);
}
bool Nio::setSink(string name)
{
- return out.setSink(name);
+ return out.setSink(name);
}
void Nio::setPostfix(std::string post)
@@ -80,12 +79,12 @@ std::string Nio::getPostfix(void) const
return postfix;
}
-
+
set<string> Nio::getSources() const
{
set<string> sources;
for(list<Engine *>::iterator itr = eng.engines.begin();
- itr != eng.engines.end(); ++itr)
+ itr != eng.engines.end(); ++itr)
if(dynamic_cast<MidiIn *>(*itr))
sources.insert((*itr)->name);
return sources;
@@ -95,19 +94,18 @@ set<string> Nio::getSinks() const
{
set<string> sinks;
for(list<Engine *>::iterator itr = eng.engines.begin();
- itr != eng.engines.end(); ++itr)
+ itr != eng.engines.end(); ++itr)
if(dynamic_cast<AudioOut *>(*itr))
sinks.insert((*itr)->name);
return sinks;
}
-
+
string Nio::getSource() const
{
- return in.getSource();
+ return in.getSource();
}
string Nio::getSink() const
{
- return out.getSink();
+ return out.getSink();
}
-
diff --git a/src/Nio/Nio.h b/src/Nio/Nio.h
@@ -1,10 +1,10 @@
-#ifndef NIO_H
+#ifndef NIO_H
#define NIO_H
#include <string>
#include <set>
/**Interface to Nio Subsystem
- *
+ *
* Should be only externally included header */
class Nio
{
@@ -34,11 +34,10 @@ class Nio
private:
Nio();
- class InMgr ∈
- class OutMgr &out;
- class EngineMgr ŋ
+ class InMgr & in;
+ class OutMgr & out;
+ class EngineMgr & eng;
std::string postfix;
};
#endif
-
diff --git a/src/Nio/NulEngine.cpp b/src/Nio/NulEngine.cpp
@@ -37,7 +37,7 @@ NulEngine::NulEngine()
void *NulEngine::_AudioThread(void *arg)
{
- return (static_cast<NulEngine*>(arg))->AudioThread();
+ return (static_cast<NulEngine *>(arg))->AudioThread();
}
void *NulEngine::AudioThread()
@@ -52,9 +52,9 @@ void *NulEngine::AudioThread()
playing_until.tv_usec = now.tv_usec;
playing_until.tv_sec = now.tv_sec;
}
- else {
+ else {
remaining = (playing_until.tv_usec - now.tv_usec)
- + (playing_until.tv_sec - now.tv_sec) * 1000000;
+ + (playing_until.tv_sec - now.tv_sec) * 1000000;
if(remaining > 10000) //Don't sleep() less than 10ms.
//This will add latency...
usleep(remaining - 10000);
@@ -71,8 +71,7 @@ void *NulEngine::AudioThread()
}
NulEngine::~NulEngine()
-{
-}
+{}
bool NulEngine::Start()
{
@@ -89,7 +88,7 @@ void NulEngine::setAudioEn(bool nval)
{
if(nval) {
if(!getAudioEn()) {
- pthread_t *thread = new pthread_t;
+ pthread_t *thread = new pthread_t;
pthread_attr_t attr;
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);
@@ -97,13 +96,12 @@ void NulEngine::setAudioEn(bool nval)
pthread_create(pThread, &attr, _AudioThread, this);
}
}
- else {
- if(getAudioEn()) {
- pthread_t *thread = pThread;
- pThread = NULL;
- pthread_join(*thread, NULL);
- delete thread;
- }
+ else
+ if(getAudioEn()) {
+ pthread_t *thread = pThread;
+ pThread = NULL;
+ pthread_join(*thread, NULL);
+ delete thread;
}
}
@@ -111,4 +109,3 @@ bool NulEngine::getAudioEn() const
{
return pThread;
}
-
diff --git a/src/Nio/NulEngine.h b/src/Nio/NulEngine.h
@@ -29,7 +29,7 @@
#include "AudioOut.h"
#include "MidiIn.h"
-class NulEngine: public AudioOut, MidiIn
+class NulEngine:public AudioOut, MidiIn
{
public:
NulEngine();
@@ -41,8 +41,8 @@ class NulEngine: public AudioOut, MidiIn
void setAudioEn(bool nval);
bool getAudioEn() const;
- void setMidiEn(bool){};
- bool getMidiEn() const{return true;};
+ void setMidiEn(bool) {}
+ bool getMidiEn() const {return true;}
protected:
void *AudioThread();
@@ -50,8 +50,7 @@ class NulEngine: public AudioOut, MidiIn
private:
struct timeval playing_until;
- pthread_t *pThread;
+ pthread_t *pThread;
};
#endif
-
diff --git a/src/Nio/OssEngine.cpp b/src/Nio/OssEngine.cpp
@@ -65,7 +65,7 @@ bool OssEngine::openAudio()
int snd_fragment = 0x00080009; //fragment size (?);
int snd_stereo = 1; //stereo;
int snd_format = AFMT_S16_LE;
- int snd_samplerate = SAMPLE_RATE;;
+ int snd_samplerate = SAMPLE_RATE;
audio.handle = open(config.cfg.LinuxOSSWaveOutDev, O_WRONLY, 0);
if(audio.handle == -1) {
@@ -79,7 +79,7 @@ bool OssEngine::openAudio()
ioctl(audio.handle, SNDCTL_DSP_SPEED, &snd_samplerate);
ioctl(audio.handle, SNDCTL_DSP_SAMPLESIZE, &snd_bitsize);
ioctl(audio.handle, SNDCTL_DSP_SETFRAGMENT, &snd_fragment);
-
+
if(!getMidiEn()) {
pthread_attr_t attr;
pthread_attr_init(&attr);
@@ -102,7 +102,7 @@ void OssEngine::stopAudio()
pthread_join(*engThread, NULL);
delete engThread;
engThread = NULL;
-
+
close(handle);
}
@@ -159,15 +159,14 @@ bool OssEngine::openMidi()
{
int handle = midi.handle;
if(handle != -1)
- return true;//already open
+ return true; //already open
handle = open(config.cfg.LinuxOSSSeqInDev, O_RDONLY, 0);
- if(-1 == handle) {
+ if(-1 == handle)
return false;
- }
midi.handle = handle;
-
+
if(!getAudioEn()) {
pthread_attr_t attr;
pthread_attr_init(&attr);
@@ -186,29 +185,27 @@ void OssEngine::stopMidi()
return;
midi.handle = -1;
-
+
if(!getAudioEn() && engThread) {
pthread_join(*engThread, NULL);
delete engThread;
engThread = NULL;
}
-
+
close(handle);
}
void *OssEngine::_thread(void *arg)
{
- return (static_cast<OssEngine*>(arg))->thread();
+ return (static_cast<OssEngine *>(arg))->thread();
}
void *OssEngine::thread()
{
unsigned char tmp[4] = {0, 0, 0, 0};
set_realtime();
- while (getAudioEn() || getMidiEn())
- {
- if(getAudioEn())
- {
+ while(getAudioEn() || getMidiEn()) {
+ if(getAudioEn()) {
const Stereo<float *> smps = getNext();
float l, r;
@@ -219,13 +216,13 @@ void *OssEngine::thread()
if(l < -1.0f)
l = -1.0f;
else
- if(l > 1.0f)
- l = 1.0f;
+ if(l > 1.0f)
+ l = 1.0f;
if(r < -1.0f)
r = -1.0f;
else
- if(r > 1.0f)
- r = 1.0f;
+ if(r > 1.0f)
+ r = 1.0f;
audio.smps[i * 2] = (short int) (l * 32767.0f);
audio.smps[i * 2 + 1] = (short int) (r * 32767.0f);
@@ -238,33 +235,36 @@ void *OssEngine::thread()
}
//Collect up to 30 midi events
- for (int k = 0; k < 30 && getMidiEn(); ++k) {
+ for(int k = 0; k < 30 && getMidiEn(); ++k) {
static char escaped;
memset(tmp, 0, 4);
- if (escaped) {
- tmp[0] = escaped;
+ if(escaped) {
+ tmp[0] = escaped;
escaped = 0;
- } else {
+ }
+ else {
getMidi(tmp);
- if (!(tmp[0] & 0x80))
+ if(!(tmp[0] & 0x80))
continue;
}
getMidi(tmp + 1);
- if (tmp[1] & 0x80) {
+ if(tmp[1] & 0x80) {
escaped = tmp[1];
- tmp[1] = 0;
- } else {
+ tmp[1] = 0;
+ }
+ else {
getMidi(tmp + 2);
- if (tmp[2] & 0x80) {
+ if(tmp[2] & 0x80) {
escaped = tmp[2];
- tmp[2] = 0;
- } else {
+ tmp[2] = 0;
+ }
+ else {
getMidi(tmp + 3);
- if (tmp[3] & 0x80) {
+ if(tmp[3] & 0x80) {
escaped = tmp[3];
- tmp[3] = 0;
+ tmp[3] = 0;
}
}
}
@@ -279,4 +279,3 @@ void OssEngine::getMidi(unsigned char *midiPtr)
{
read(midi.handle, midiPtr, 1);
}
-
diff --git a/src/Nio/OssEngine.h b/src/Nio/OssEngine.h
@@ -28,7 +28,7 @@
#include "AudioOut.h"
#include "MidiIn.h"
-class OssEngine: public AudioOut, MidiIn
+class OssEngine:public AudioOut, MidiIn
{
public:
OssEngine();
@@ -55,7 +55,7 @@ class OssEngine: public AudioOut, MidiIn
bool openAudio();
void stopAudio();
- struct audio{
+ struct audio {
int handle;
short int *smps; //Samples to be sent to soundcard
bool en;
@@ -66,12 +66,11 @@ class OssEngine: public AudioOut, MidiIn
void stopMidi();
void getMidi(unsigned char *midiPtr);
- struct midi{
- int handle;
+ struct midi {
+ int handle;
bool en;
bool run;
} midi;
};
#endif
-
diff --git a/src/Nio/OutMgr.cpp b/src/Nio/OutMgr.cpp
@@ -7,8 +7,8 @@
#include "InMgr.h"
#include "WavEngine.h"
#include "../Misc/Master.h"
-#include "../Misc/Util.h"//for set_realtime()
-#include "../Samples/Sample.h"//for resampling
+#include "../Misc/Util.h" //for set_realtime()
+#include "../Samples/Sample.h" //for resampling
using namespace std;
@@ -20,16 +20,18 @@ OutMgr &OutMgr::getInstance()
OutMgr::OutMgr()
:wave(new WavEngine()),
- priBuf(new float[4096],new float[4096]),priBuffCurrent(priBuf),master(Master::getInstance())
+ priBuf(new float[4096],
+ new float[4096]), priBuffCurrent(priBuf),
+ master(Master::getInstance())
{
currentOut = NULL;
- stales = 0;
- master = Master::getInstance();
+ stales = 0;
+ master = Master::getInstance();
//init samples
outr = new float[SOUND_BUFFER_SIZE];
outl = new float[SOUND_BUFFER_SIZE];
-};
+}
OutMgr::~OutMgr()
{
@@ -62,7 +64,7 @@ const Stereo<float *> OutMgr::tick(unsigned int frameSize)
pthread_mutex_lock(&(master.mutex));
master.AudioOut(outl, outr);
pthread_mutex_unlock(&(master.mutex));
- addSmps(outl,outr);
+ addSmps(outl, outr);
}
Stereo<float *> ans = priBuffCurrent;
ans.l -= frameSize;
@@ -117,19 +119,23 @@ string OutMgr::getSink() const
void OutMgr::addSmps(float *l, float *r)
{
//allow wave file to syphon off stream
- wave->push(Stereo<float *>(l,r),SOUND_BUFFER_SIZE);
+ wave->push(Stereo<float *>(l, r), SOUND_BUFFER_SIZE);
if(currentOut->getSampleRate() != SAMPLE_RATE) { //we need to resample
//cout << "BAD RESAMPLING" << endl;
- Stereo<Sample> smps(Sample(SOUND_BUFFER_SIZE, l), Sample(SOUND_BUFFER_SIZE, r));
- smps.l.resample(SAMPLE_RATE,currentOut->getSampleRate());
- smps.r.resample(SAMPLE_RATE,currentOut->getSampleRate());
- memcpy(priBuffCurrent.l, smps.l.c_buf(), SOUND_BUFFER_SIZE*sizeof(float));
- memcpy(priBuffCurrent.r, smps.r.c_buf(), SOUND_BUFFER_SIZE*sizeof(float));
+ Stereo<Sample> smps(Sample(SOUND_BUFFER_SIZE, l), Sample(
+ SOUND_BUFFER_SIZE,
+ r));
+ smps.l.resample(SAMPLE_RATE, currentOut->getSampleRate());
+ smps.r.resample(SAMPLE_RATE, currentOut->getSampleRate());
+ memcpy(priBuffCurrent.l, smps.l.c_buf(), SOUND_BUFFER_SIZE
+ * sizeof(float));
+ memcpy(priBuffCurrent.r, smps.r.c_buf(), SOUND_BUFFER_SIZE
+ * sizeof(float));
}
else { //just copy the samples
- memcpy(priBuffCurrent.l, l, SOUND_BUFFER_SIZE*sizeof(float));
- memcpy(priBuffCurrent.r, r, SOUND_BUFFER_SIZE*sizeof(float));
+ memcpy(priBuffCurrent.l, l, SOUND_BUFFER_SIZE * sizeof(float));
+ memcpy(priBuffCurrent.r, r, SOUND_BUFFER_SIZE * sizeof(float));
}
priBuffCurrent.l += SOUND_BUFFER_SIZE;
priBuffCurrent.r += SOUND_BUFFER_SIZE;
@@ -142,9 +148,8 @@ void OutMgr::removeStaleSmps()
return;
//memset is possibly unneeded
- memset(priBuf.l, '0', 4096*sizeof(float));
- memset(priBuf.r, '0', 4096*sizeof(float));
+ memset(priBuf.l, '0', 4096 * sizeof(float));
+ memset(priBuf.r, '0', 4096 * sizeof(float));
priBuffCurrent = priBuf;
stales = 0;
}
-
diff --git a/src/Nio/OutMgr.h b/src/Nio/OutMgr.h
@@ -21,7 +21,7 @@ class OutMgr
/**Request a new set of samples
* @param n number of requested samples (defaults to 1)
* @return -1 for locking issues 0 for valid request*/
- void requestSamples(unsigned int n=1);
+ void requestSamples(unsigned int n = 1);
/**Gets requested driver
* @param name case unsensitive name of driver
@@ -39,15 +39,15 @@ class OutMgr
std::string getSink() const;
- class WavEngine *wave; /**<The Wave Recorder*/
+ class WavEngine * wave; /**<The Wave Recorder*/
friend class EngineMgr;
private:
OutMgr();
void addSmps(float *l, float *r);
- unsigned int storedSmps() const {return priBuffCurrent.l - priBuf.l;};
+ unsigned int storedSmps() const {return priBuffCurrent.l - priBuf.l;}
void removeStaleSmps();
- AudioOut *currentOut;/**<The current output driver*/
+ AudioOut *currentOut; /**<The current output driver*/
sem_t requested;
@@ -57,10 +57,9 @@ class OutMgr
float *outl;
float *outr;
- class Master &master;
+ class Master & master;
int stales;
};
#endif
-
diff --git a/src/Nio/PaEngine.cpp b/src/Nio/PaEngine.cpp
@@ -45,13 +45,13 @@ bool PaEngine::Start()
PaStreamParameters outputParameters;
outputParameters.device = Pa_GetDefaultOutputDevice();
- if (outputParameters.device == paNoDevice) {
+ if(outputParameters.device == paNoDevice) {
cerr << "Error: No default output device." << endl;
Pa_Terminate();
return false;
}
- outputParameters.channelCount = 2; /* stereo output */
- outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
+ outputParameters.channelCount = 2; /* stereo output */
+ outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency =
Pa_GetDeviceInfo(outputParameters.device)->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
@@ -82,10 +82,12 @@ bool PaEngine::getAudioEn() const
return stream;
}
-int PaEngine::PAprocess(const void *inputBuffer, void *outputBuffer,
- unsigned long framesPerBuffer,
- const PaStreamCallbackTimeInfo *outTime, PaStreamCallbackFlags flags,
- void *userData)
+int PaEngine::PAprocess(const void *inputBuffer,
+ void *outputBuffer,
+ unsigned long framesPerBuffer,
+ const PaStreamCallbackTimeInfo *outTime,
+ PaStreamCallbackFlags flags,
+ void *userData)
{
(void) inputBuffer;
(void) outTime;
@@ -114,4 +116,3 @@ void PaEngine::Stop()
stream = NULL;
Pa_Terminate();
}
-
diff --git a/src/Nio/PaEngine.h b/src/Nio/PaEngine.h
@@ -27,7 +27,7 @@
#include "../globals.h"
#include "AudioOut.h"
-class PaEngine: public AudioOut
+class PaEngine:public AudioOut
{
public:
PaEngine();
@@ -40,10 +40,12 @@ class PaEngine: public AudioOut
bool getAudioEn() const;
protected:
- static int PAprocess(const void *inputBuffer, void *outputBuffer,
- unsigned long framesPerBuffer,
- const PaStreamCallbackTimeInfo *outTime, PaStreamCallbackFlags flags,
- void *userData);
+ static int PAprocess(const void *inputBuffer,
+ void *outputBuffer,
+ unsigned long framesPerBuffer,
+ const PaStreamCallbackTimeInfo *outTime,
+ PaStreamCallbackFlags flags,
+ void *userData);
int process(float *out, unsigned long framesPerBuffer);
private:
PaStream *stream;
@@ -53,4 +55,3 @@ class PaEngine: public AudioOut
void PAfinish();
#endif
-
diff --git a/src/Nio/SafeQueue.cpp b/src/Nio/SafeQueue.cpp
@@ -1,9 +1,9 @@
template<class T>
SafeQueue<T>::SafeQueue(size_t maxlen)
- :writePtr(0),readPtr(0),bufSize(maxlen)
+ :writePtr(0), readPtr(0), bufSize(maxlen)
{
- sem_init(&w_space, PTHREAD_PROCESS_PRIVATE, maxlen-1);
+ sem_init(&w_space, PTHREAD_PROCESS_PRIVATE, maxlen - 1);
sem_init(&r_space, PTHREAD_PROCESS_PRIVATE, 0);
buffer = new T[maxlen];
}
@@ -47,10 +47,10 @@ int SafeQueue<T>::push(const T &in)
//ok, there is space to write
size_t w = (writePtr + 1) % bufSize;
buffer[w] = in;
- writePtr = w;
+ writePtr = w;
//adjust ranges
- sem_wait(&w_space);//guaranteed not to wait
+ sem_wait(&w_space); //guaranteed not to wait
sem_post(&r_space);
return 0;
}
@@ -63,11 +63,11 @@ int SafeQueue<T>::pop(T &out)
//ok, there is space to read
size_t r = (readPtr + 1) % bufSize;
- out = buffer[r];
+ out = buffer[r];
readPtr = r;
//adjust ranges
- sem_wait(&r_space);//guaranteed not to wait
+ sem_wait(&r_space); //guaranteed not to wait
sem_post(&w_space);
return 0;
}
diff --git a/src/Nio/SafeQueue.h b/src/Nio/SafeQueue.h
@@ -7,7 +7,7 @@
/**
* C++ thread safe lockless queue
* Based off of jack's ringbuffer*/
-template <class T>
+template<class T>
class SafeQueue
{
public:
diff --git a/src/Nio/WavEngine.cpp b/src/Nio/WavEngine.cpp
@@ -26,7 +26,7 @@
using namespace std;
WavEngine::WavEngine()
- :AudioOut(), file(NULL), buffer(SAMPLE_RATE*4), pThread(NULL)
+ :AudioOut(), file(NULL), buffer(SAMPLE_RATE * 4), pThread(NULL)
{
sem_init(&work, PTHREAD_PROCESS_PRIVATE, 0);
}
@@ -92,7 +92,9 @@ void WavEngine::newFile(WavFile *_file)
//check state
if(!file->good())
- cerr << "ERROR: WavEngine handed bad file output WavEngine::newFile()" << endl;
+ cerr
+ << "ERROR: WavEngine handed bad file output WavEngine::newFile()"
+ << endl;
}
void WavEngine::destroyFile()
@@ -104,21 +106,24 @@ void WavEngine::destroyFile()
void *WavEngine::_AudioThread(void *arg)
{
- return (static_cast<WavEngine*>(arg))->AudioThread();
+ return (static_cast<WavEngine *>(arg))->AudioThread();
}
void *WavEngine::AudioThread()
{
- short *recordbuf_16bit = new short[2*SOUND_BUFFER_SIZE];
+ short *recordbuf_16bit = new short[2 * SOUND_BUFFER_SIZE];
- while(!sem_wait(&work) && pThread)
- {
+ while(!sem_wait(&work) && pThread) {
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- float left=0.0f, right=0.0f;
+ float left = 0.0f, right = 0.0f;
buffer.pop(left);
buffer.pop(right);
- recordbuf_16bit[2*i] = limit((int)(left * 32767.0f), -32768, 32767);
- recordbuf_16bit[2*i+1] = limit((int)(right * 32767.0f), -32768, 32767);
+ recordbuf_16bit[2 * i] = limit((int)(left * 32767.0f),
+ -32768,
+ 32767);
+ recordbuf_16bit[2 * i + 1] = limit((int)(right * 32767.0f),
+ -32768,
+ 32767);
}
file->writeStereoSamples(SOUND_BUFFER_SIZE, recordbuf_16bit);
}
@@ -127,4 +132,3 @@ void *WavEngine::AudioThread()
return NULL;
}
-
diff --git a/src/Nio/WavEngine.h b/src/Nio/WavEngine.h
@@ -29,7 +29,7 @@
#include "SafeQueue.h"
class WavFile;
-class WavEngine: public AudioOut
+class WavEngine:public AudioOut
{
public:
WavEngine();
@@ -39,8 +39,8 @@ class WavEngine: public AudioOut
bool Start();
void Stop();
- void setAudioEn(bool /*nval*/){};
- bool getAudioEn() const{return true;};
+ void setAudioEn(bool /*nval*/) {}
+ bool getAudioEn() const {return true;}
void push(Stereo<float *> smps, size_t len);
@@ -53,10 +53,9 @@ class WavEngine: public AudioOut
private:
WavFile *file;
- sem_t work;
+ sem_t work;
SafeQueue<float> buffer;
pthread_t *pThread;
};
#endif
-
diff --git a/src/Output/DSSIaudiooutput.cpp b/src/Output/DSSIaudiooutput.cpp
@@ -43,7 +43,9 @@ using std::vector;
// LADSPA is essentially a C handle based API; This plug-in implementation is
// a C++ OO one so we need stub functions to map from C API calls to C++ object
// method calls.
-void DSSIaudiooutput::stub_connectPort(LADSPA_Handle instance, unsigned long port, LADSPA_Data * data)
+void DSSIaudiooutput::stub_connectPort(LADSPA_Handle instance,
+ unsigned long port,
+ LADSPA_Data *data)
{
getInstance(instance)->connectPort(port, data);
}
@@ -53,7 +55,8 @@ void DSSIaudiooutput::stub_activate(LADSPA_Handle instance)
getInstance(instance)->activate();
}
-void DSSIaudiooutput::stub_run(LADSPA_Handle instance, unsigned long sample_count)
+void DSSIaudiooutput::stub_run(LADSPA_Handle instance,
+ unsigned long sample_count)
{
getInstance(instance)->run(sample_count);
}
@@ -66,7 +69,7 @@ void DSSIaudiooutput::stub_deactivate(LADSPA_Handle instance)
void DSSIaudiooutput::stub_cleanup(LADSPA_Handle instance)
{
- DSSIaudiooutput* plugin_instance = getInstance(instance);
+ DSSIaudiooutput *plugin_instance = getInstance(instance);
plugin_instance->cleanup();
delete plugin_instance;
}
@@ -83,23 +86,30 @@ const LADSPA_Descriptor *ladspa_descriptor(unsigned long index)
// DSSI is essentially a C handle based API; This plug-in implementation is
// a C++ OO one so we need stub functions to map from C API calls to C++ object
// method calls.
-const DSSI_Program_Descriptor* DSSIaudiooutput::stub_getProgram (LADSPA_Handle instance, unsigned long index)
+const DSSI_Program_Descriptor *DSSIaudiooutput::stub_getProgram(
+ LADSPA_Handle instance,
+ unsigned long index)
{
return getInstance(instance)->getProgram(index);
}
-void DSSIaudiooutput::stub_selectProgram(LADSPA_Handle instance, unsigned long bank, unsigned long program)
+void DSSIaudiooutput::stub_selectProgram(LADSPA_Handle instance,
+ unsigned long bank,
+ unsigned long program)
{
getInstance(instance)->selectProgram(bank, program);
}
-int DSSIaudiooutput::stub_getMidiControllerForPort(LADSPA_Handle instance, unsigned long port)
+int DSSIaudiooutput::stub_getMidiControllerForPort(LADSPA_Handle instance,
+ unsigned long port)
{
return getInstance(instance)->getMidiControllerForPort(port);
}
-void DSSIaudiooutput::stub_runSynth(LADSPA_Handle instance, unsigned long sample_count,
- snd_seq_event_t *events, unsigned long event_count)
+void DSSIaudiooutput::stub_runSynth(LADSPA_Handle instance,
+ unsigned long sample_count,
+ snd_seq_event_t *events,
+ unsigned long event_count)
{
getInstance(instance)->runSynth(sample_count, events, event_count);
}
@@ -129,16 +139,13 @@ const DSSI_Descriptor *dssi_descriptor(unsigned long index)
* @param s_rate [in] the sample rate
* @return the plug-in instance handle if successful else NULL
*/
-LADSPA_Handle DSSIaudiooutput::instantiate(const LADSPA_Descriptor * descriptor, unsigned long s_rate)
+LADSPA_Handle DSSIaudiooutput::instantiate(const LADSPA_Descriptor *descriptor,
+ unsigned long s_rate)
{
if(descriptor->UniqueID == dssiDescriptor->LADSPA_Plugin->UniqueID)
- {
return (LADSPA_Handle)(new DSSIaudiooutput(s_rate));
- }
else
- {
return NULL;
- }
}
/**
@@ -159,15 +166,15 @@ LADSPA_Handle DSSIaudiooutput::instantiate(const LADSPA_Descriptor * descriptor,
* @param port [in] the port to be connected
* @param data [in] the data buffer to write to / read from
*/
-void DSSIaudiooutput::connectPort(unsigned long port, LADSPA_Data * data)
+void DSSIaudiooutput::connectPort(unsigned long port, LADSPA_Data *data)
{
- switch (port) {
- case 0:
- outl = data;
- break;
- case 1:
- outr = data;
- break;
+ switch(port) {
+ case 0:
+ outl = data;
+ break;
+ case 1:
+ outr = data;
+ break;
}
}
@@ -188,8 +195,7 @@ void DSSIaudiooutput::connectPort(unsigned long port, LADSPA_Data * data)
* too much code here seems to cause time-out problems in jack-dssi-host.
*/
void DSSIaudiooutput::activate()
-{
-}
+{}
/**
* Runs an instance of a plug-in for a block.
@@ -208,7 +214,7 @@ void DSSIaudiooutput::activate()
*/
void DSSIaudiooutput::run(unsigned long sample_count)
{
- runSynth(sample_count,NULL,(unsigned long)0);
+ runSynth(sample_count, NULL, (unsigned long)0);
}
/**
@@ -223,9 +229,7 @@ void DSSIaudiooutput::run(unsigned long sample_count)
* Currently this function does nothing.
*/
void DSSIaudiooutput::deactivate()
-{
-
-}
+{}
/**
* Deletes a plug-in instance that is no longer required.
@@ -242,8 +246,7 @@ void DSSIaudiooutput::deactivate()
* Currently cleanup is deferred to the destructor that is invoked after cleanup()
*/
void DSSIaudiooutput::cleanup()
-{
-}
+{}
/**
* Initial entry point for the LADSPA plug-in library.
@@ -263,9 +266,10 @@ void DSSIaudiooutput::cleanup()
* @param index [in] the index number of the plug-in within the library.
* @return if index is in range, a pointer to the plug-in descriptor is returned, else NULL
*/
-const LADSPA_Descriptor* DSSIaudiooutput::getLadspaDescriptor(unsigned long index)
+const LADSPA_Descriptor *DSSIaudiooutput::getLadspaDescriptor(
+ unsigned long index)
{
- if(index > 0 || dssiDescriptor == NULL)
+ if((index > 0) || (dssiDescriptor == NULL))
return NULL;
else
return dssiDescriptor->LADSPA_Plugin;
@@ -302,7 +306,7 @@ const LADSPA_Descriptor* DSSIaudiooutput::getLadspaDescriptor(unsigned long inde
* guaranteed to be valid only until the next call to get_program,
* deactivate, or configure, on the same plug-in instance, or NULL if index is out of range.
*/
-const DSSI_Program_Descriptor* DSSIaudiooutput::getProgram (unsigned long index)
+const DSSI_Program_Descriptor *DSSIaudiooutput::getProgram(unsigned long index)
{
static DSSI_Program_Descriptor retVal;
@@ -310,20 +314,17 @@ const DSSI_Program_Descriptor* DSSIaudiooutput::getProgram (unsigned long index)
initBanks();
/* Make sure that the bank containing the instrument has been mapped */
- while (index >= programMap.size() && mapNextBank())
+ while(index >= programMap.size() && mapNextBank())
/* DO NOTHING MORE */;
if(index >= programMap.size())
- {
/* No more instruments */
return NULL;
- }
- else
- {
+ else {
/* OK, return the instrument */
- retVal.Name = programMap[index].name.c_str();
+ retVal.Name = programMap[index].name.c_str();
retVal.Program = programMap[index].program;
- retVal.Bank = programMap[index].bank;
+ retVal.Bank = programMap[index].bank;
return &retVal;
}
}
@@ -349,11 +350,9 @@ void DSSIaudiooutput::selectProgram(unsigned long bank, unsigned long program)
{
initBanks();
// cerr << "selectProgram(" << (bank & 0x7F) << ':' << ((bank >> 7) & 0x7F) << "," << program << ")" << '\n';
- if(bank < master->bank.banks.size() && program < BANK_SIZE)
- {
- const std::string bankdir = master->bank.banks[ bank ].dir;
- if(!bankdir.empty())
- {
+ if((bank < master->bank.banks.size()) && (program < BANK_SIZE)) {
+ const std::string bankdir = master->bank.banks[bank].dir;
+ if(!bankdir.empty()) {
pthread_mutex_lock(&master->mutex);
/* We have to turn off the CheckPADsynth functionality, else
@@ -422,55 +421,58 @@ int DSSIaudiooutput::getMidiControllerForPort(unsigned long port)
* events.
* @param event_count [in] the number of entries in the `events` block
*/
-void DSSIaudiooutput::runSynth(unsigned long sample_count, snd_seq_event_t *events, unsigned long event_count)
+void DSSIaudiooutput::runSynth(unsigned long sample_count,
+ snd_seq_event_t *events,
+ unsigned long event_count)
{
- unsigned long from_frame = 0;
- unsigned long event_index = 0;
+ unsigned long from_frame = 0;
+ unsigned long event_index = 0;
unsigned long next_event_frame = 0;
unsigned long to_frame = 0;
pthread_mutex_lock(&master->mutex);
do {
/* Find the time of the next event, if any */
- if(events == NULL || event_index >= event_count)
+ if((events == NULL) || (event_index >= event_count))
next_event_frame = ULONG_MAX;
else
next_event_frame = events[event_index].time.tick;
/* find the end of the sub-sample to be processed this time round... */
/* if the next event falls within the desired sample interval... */
- if(next_event_frame < sample_count && next_event_frame >= to_frame)
+ if((next_event_frame < sample_count) && (next_event_frame >= to_frame))
/* set the end to be at that event */
to_frame = next_event_frame;
else
/* ...else go for the whole remaining sample */
to_frame = sample_count;
- if(from_frame<to_frame)
- {
+ if(from_frame < to_frame) {
// call master to fill from `from_frame` to `to_frame`:
- master->GetAudioOutSamples(to_frame - from_frame, (int)sampleRate, &(outl[from_frame]), &(outr[from_frame]));
+ master->GetAudioOutSamples(to_frame - from_frame,
+ (int)sampleRate,
+ &(outl[from_frame]),
+ &(outr[from_frame]));
// next sub-sample please...
from_frame = to_frame;
}
// Now process any event(s) at the current timing point
- while(events != NULL && event_index < event_count && events[event_index].time.tick == to_frame)
- {
+ while(events != NULL && event_index < event_count
+ && events[event_index].time.tick == to_frame) {
if(events[event_index].type == SND_SEQ_EVENT_NOTEON)
- {
- master->noteOn(events[event_index].data.note.channel, events[event_index].data.note.note, events[event_index].data.note.velocity);
- }
- else if(events[event_index].type == SND_SEQ_EVENT_NOTEOFF)
- {
- master->noteOff(events[event_index].data.note.channel, events[event_index].data.note.note);
- }
- else if(events[event_index].type == SND_SEQ_EVENT_CONTROLLER)
- {
- master->setController(events[event_index].data.control.channel, events[event_index].data.control.param, events[event_index].data.control.value);
- }
+ master->noteOn(events[event_index].data.note.channel,
+ events[event_index].data.note.note,
+ events[event_index].data.note.velocity);
+ else
+ if(events[event_index].type == SND_SEQ_EVENT_NOTEOFF)
+ master->noteOff(events[event_index].data.note.channel,
+ events[event_index].data.note.note);
else
- {
- }
+ if(events[event_index].type == SND_SEQ_EVENT_CONTROLLER)
+ master->setController(events[event_index].data.control.channel,
+ events[event_index].data.control.param,
+ events[event_index].data.control.value);
+ else {}
event_index++;
}
@@ -497,9 +499,9 @@ void DSSIaudiooutput::runSynth(unsigned long sample_count, snd_seq_event_t *even
* @param index [in] the index number of the plug-in within the library.
* @return if index is in range, a pointer to the plug-in descriptor is returned, else NULL
*/
-const DSSI_Descriptor* DSSIaudiooutput::getDssiDescriptor(unsigned long index)
+const DSSI_Descriptor *DSSIaudiooutput::getDssiDescriptor(unsigned long index)
{
- if(index > 0 || dssiDescriptor == NULL)
+ if((index > 0) || (dssiDescriptor == NULL))
return NULL;
else
return dssiDescriptor;
@@ -510,64 +512,67 @@ const DSSI_Descriptor* DSSIaudiooutput::getDssiDescriptor(unsigned long index)
//
// Initialise the DSSI descriptor, statically:
-DSSI_Descriptor* DSSIaudiooutput::dssiDescriptor = DSSIaudiooutput::initDssiDescriptor();
+DSSI_Descriptor *DSSIaudiooutput:: dssiDescriptor =
+ DSSIaudiooutput::initDssiDescriptor();
/**
* Initializes the DSSI (and LADSPA) descriptor, returning it is an object.
*/
-DSSI_Descriptor* DSSIaudiooutput::initDssiDescriptor()
+DSSI_Descriptor *DSSIaudiooutput::initDssiDescriptor()
{
- DSSI_Descriptor* newDssiDescriptor = new DSSI_Descriptor;
-
- LADSPA_PortDescriptor* newPortDescriptors;
- const char** newPortNames;
- LADSPA_PortRangeHint* newPortRangeHints;
-
- if (newDssiDescriptor)
- {
- LADSPA_Descriptor* newLadspaDescriptor = new LADSPA_Descriptor;
- if (newLadspaDescriptor)
- {
- newLadspaDescriptor->UniqueID = 100;
- newLadspaDescriptor->Label = "ZASF";
+ DSSI_Descriptor *newDssiDescriptor = new DSSI_Descriptor;
+
+ LADSPA_PortDescriptor *newPortDescriptors;
+ const char **newPortNames;
+ LADSPA_PortRangeHint *newPortRangeHints;
+
+ if(newDssiDescriptor) {
+ LADSPA_Descriptor *newLadspaDescriptor = new LADSPA_Descriptor;
+ if(newLadspaDescriptor) {
+ newLadspaDescriptor->UniqueID = 100;
+ newLadspaDescriptor->Label = "ZASF";
newLadspaDescriptor->Properties = 0;
- newLadspaDescriptor->Name = "ZynAddSubFX";
- newLadspaDescriptor->Maker = "Nasca Octavian Paul <[email protected]>";
+ newLadspaDescriptor->Name = "ZynAddSubFX";
+ newLadspaDescriptor->Maker =
+ "Nasca Octavian Paul <[email protected]>";
newLadspaDescriptor->Copyright = "GNU General Public License v.2";
newLadspaDescriptor->PortCount = 2;
- newPortNames = new const char *[newLadspaDescriptor->PortCount];
+ newPortNames = new const char *[newLadspaDescriptor->PortCount];
newPortNames[0] = "Output L";
newPortNames[1] = "Output R";
newLadspaDescriptor->PortNames = newPortNames;
- newPortDescriptors = new LADSPA_PortDescriptor[newLadspaDescriptor->PortCount];
+ newPortDescriptors =
+ new LADSPA_PortDescriptor[newLadspaDescriptor->PortCount];
newPortDescriptors[0] = LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO;
newPortDescriptors[1] = LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO;
newLadspaDescriptor->PortDescriptors = newPortDescriptors;
- newPortRangeHints = new LADSPA_PortRangeHint[newLadspaDescriptor->PortCount];
+ newPortRangeHints =
+ new LADSPA_PortRangeHint[newLadspaDescriptor->PortCount];
newPortRangeHints[0].HintDescriptor = 0;
newPortRangeHints[1].HintDescriptor = 0;
newLadspaDescriptor->PortRangeHints = newPortRangeHints;
- newLadspaDescriptor->activate = stub_activate;
- newLadspaDescriptor->cleanup = stub_cleanup;
+ newLadspaDescriptor->activate = stub_activate;
+ newLadspaDescriptor->cleanup = stub_cleanup;
newLadspaDescriptor->connect_port = stub_connectPort;
- newLadspaDescriptor->deactivate = stub_deactivate;
- newLadspaDescriptor->instantiate = instantiate;
+ newLadspaDescriptor->deactivate = stub_deactivate;
+ newLadspaDescriptor->instantiate = instantiate;
newLadspaDescriptor->run = stub_run;
newLadspaDescriptor->run_adding = NULL;
newLadspaDescriptor->set_run_adding_gain = NULL;
}
- newDssiDescriptor->LADSPA_Plugin = newLadspaDescriptor;
+ newDssiDescriptor->LADSPA_Plugin = newLadspaDescriptor;
newDssiDescriptor->DSSI_API_Version = 1;
- newDssiDescriptor->configure = NULL;
+ newDssiDescriptor->configure = NULL;
newDssiDescriptor->get_program = stub_getProgram;
- newDssiDescriptor->get_midi_controller_for_port = stub_getMidiControllerForPort;
- newDssiDescriptor->select_program = stub_selectProgram;
- newDssiDescriptor->run_synth = stub_runSynth;
- newDssiDescriptor->run_synth_adding = NULL;
+ newDssiDescriptor->get_midi_controller_for_port =
+ stub_getMidiControllerForPort;
+ newDssiDescriptor->select_program = stub_selectProgram;
+ newDssiDescriptor->run_synth = stub_runSynth;
+ newDssiDescriptor->run_synth_adding = NULL;
newDssiDescriptor->run_multiple_synths = NULL;
newDssiDescriptor->run_multiple_synths_adding = NULL;
}
@@ -583,9 +588,9 @@ DSSI_Descriptor* DSSIaudiooutput::initDssiDescriptor()
* @param instance [in]
* @return the instance
*/
-DSSIaudiooutput* DSSIaudiooutput::getInstance(LADSPA_Handle instance)
+DSSIaudiooutput *DSSIaudiooutput::getInstance(LADSPA_Handle instance)
{
- return (DSSIaudiooutput*)(instance);
+ return (DSSIaudiooutput *)(instance);
}
/**
@@ -597,16 +602,17 @@ DSSIaudiooutput* DSSIaudiooutput::getInstance(LADSPA_Handle instance)
*/
DSSIaudiooutput::DSSIaudiooutput(unsigned long sampleRate)
{
- SAMPLE_RATE = sampleRate;
+ SAMPLE_RATE = sampleRate;
- this->sampleRate = sampleRate;
+ this->sampleRate = sampleRate;
this->banksInited = false;
config.init();
sprng(time(NULL));
- denormalkillbuf=new float [SOUND_BUFFER_SIZE];
- for (int i=0;i<SOUND_BUFFER_SIZE;i++) denormalkillbuf[i]=(RND-0.5f)*1e-16;
+ denormalkillbuf = new float [SOUND_BUFFER_SIZE];
+ for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ denormalkillbuf[i] = (RND - 0.5f) * 1e-16;
this->master = new Master();
}
@@ -616,16 +622,14 @@ DSSIaudiooutput::DSSIaudiooutput(unsigned long sampleRate)
* @return
*/
DSSIaudiooutput::~DSSIaudiooutput()
-{
-}
+{}
/**
* Ensures the list of bank (directories) has been initialised.
*/
void DSSIaudiooutput::initBanks(void)
{
- if(!banksInited)
- {
+ if(!banksInited) {
pthread_mutex_lock(&master->mutex);
master->bank.rescanforbanks();
banksInited = true;
@@ -641,20 +645,22 @@ void DSSIaudiooutput::initBanks(void)
* @param _name [in] instrument / sample name
* @return
*/
-DSSIaudiooutput::ProgramDescriptor::ProgramDescriptor(unsigned long _bank, unsigned long _program, char* _name) :
- bank(_bank), program(_program), name(_name)
-{
-}
+DSSIaudiooutput::ProgramDescriptor::ProgramDescriptor(unsigned long _bank,
+ unsigned long _program,
+ char *_name)
+ :bank(_bank), program(_program), name(_name)
+{}
/**
* The map of programs available; held as a single shared statically allocated object.
*/
-vector <DSSIaudiooutput::ProgramDescriptor> DSSIaudiooutput::programMap = vector<DSSIaudiooutput::ProgramDescriptor>();
+vector<DSSIaudiooutput::ProgramDescriptor> DSSIaudiooutput:: programMap =
+ vector<DSSIaudiooutput::ProgramDescriptor>();
/**
* Index controlling the map of banks
*/
-long DSSIaudiooutput::bankNoToMap = 1;
+long DSSIaudiooutput:: bankNoToMap = 1;
/**
* Queries and maps the next available bank of instruments.
@@ -666,25 +672,22 @@ long DSSIaudiooutput::bankNoToMap = 1;
bool DSSIaudiooutput::mapNextBank()
{
pthread_mutex_lock(&master->mutex);
- Bank& bank = master->bank;
- bool retval;
- if(bankNoToMap >= (int)bank.banks.size() || bank.banks[bankNoToMap].dir.empty())
- {
+ Bank &bank = master->bank;
+ bool retval;
+ if((bankNoToMap >= (int)bank.banks.size())
+ || bank.banks[bankNoToMap].dir.empty())
retval = false;
- }
- else
- {
+ else {
bank.loadbank(bank.banks[bankNoToMap].dir);
- for(unsigned long instrument = 0; instrument < BANK_SIZE; ++instrument)
- {
+ for(unsigned long instrument = 0; instrument < BANK_SIZE;
+ ++instrument) {
string insName = bank.getname(instrument);
- if(!insName.empty() && insName[0] != '\0' && insName[0] != ' ')
- {
- programMap.push_back(ProgramDescriptor(bankNoToMap,instrument,
- const_cast<char*>(insName.c_str())));
- }
+ if(!insName.empty() && (insName[0] != '\0') && (insName[0] != ' '))
+ programMap.push_back(ProgramDescriptor(bankNoToMap, instrument,
+ const_cast<char *>(
+ insName.c_str())));
}
- bankNoToMap ++;
+ bankNoToMap++;
retval = true;
}
pthread_mutex_unlock(&master->mutex);
diff --git a/src/Output/DSSIaudiooutput.h b/src/Output/DSSIaudiooutput.h
@@ -33,79 +33,91 @@
class DSSIaudiooutput
{
-public:
- //
- // Static stubs for LADSPA member functions
- //
- static void stub_connectPort(LADSPA_Handle instance, unsigned long port, LADSPA_Data * data);
- static void stub_activate(LADSPA_Handle instance);
- static void stub_run(LADSPA_Handle instance, unsigned long sample_count);
- static void stub_deactivate(LADSPA_Handle Instance);
- static void stub_cleanup(LADSPA_Handle instance);
-
- //
- // Static stubs for DSSI member functions
- //
- static const DSSI_Program_Descriptor* stub_getProgram (LADSPA_Handle instance, unsigned long Index);
- static void stub_selectProgram(LADSPA_Handle instance, unsigned long bank, unsigned long program);
- static int stub_getMidiControllerForPort(LADSPA_Handle instance, unsigned long port);
- static void stub_runSynth(LADSPA_Handle instance, unsigned long sample_count,
- snd_seq_event_t *events, unsigned long event_count);
-
- /*
- * LADSPA member functions
- */
- static LADSPA_Handle instantiate(const LADSPA_Descriptor * descriptor, unsigned long s_rate);
- void connectPort(unsigned long port, LADSPA_Data * data);
- void activate();
- void run(unsigned long sample_count);
- void deactivate();
- void cleanup();
- static const LADSPA_Descriptor* getLadspaDescriptor(unsigned long index);
-
- /*
- * DSSI member functions
- */
- const DSSI_Program_Descriptor* getProgram (unsigned long Index);
- void selectProgram(unsigned long bank, unsigned long program);
- int getMidiControllerForPort(unsigned long port);
- void runSynth(unsigned long sample_count, snd_seq_event_t *events, unsigned long event_count);
- static const DSSI_Descriptor* getDssiDescriptor(unsigned long index);
-
- struct ProgramDescriptor
- {
- unsigned long bank;
- unsigned long program;
- std::string name;
- ProgramDescriptor(unsigned long _bank, unsigned long _program, char* _name);
- };
-
-private:
-
- DSSIaudiooutput(unsigned long sampleRate);
- ~DSSIaudiooutput();
- static DSSI_Descriptor* initDssiDescriptor();
- static DSSIaudiooutput* getInstance(LADSPA_Handle instance);
- void initBanks();
- bool mapNextBank();
-
- LADSPA_Data *outl;
- LADSPA_Data *outr;
- long sampleRate;
- Master* master;
- static DSSI_Descriptor* dssiDescriptor;
- static std::string bankDirNames[];
- static
- std::vector <ProgramDescriptor> programMap;
-
- /**
- * Flag controlling the list of bank directories
- */
- bool banksInited;
-
- static
- long bankNoToMap;
+ public:
+ //
+ // Static stubs for LADSPA member functions
+ //
+ static void stub_connectPort(LADSPA_Handle instance,
+ unsigned long port,
+ LADSPA_Data *data);
+ static void stub_activate(LADSPA_Handle instance);
+ static void stub_run(LADSPA_Handle instance, unsigned long sample_count);
+ static void stub_deactivate(LADSPA_Handle Instance);
+ static void stub_cleanup(LADSPA_Handle instance);
+
+ //
+ // Static stubs for DSSI member functions
+ //
+ static const DSSI_Program_Descriptor *stub_getProgram(
+ LADSPA_Handle instance,
+ unsigned long Index);
+ static void stub_selectProgram(LADSPA_Handle instance,
+ unsigned long bank,
+ unsigned long program);
+ static int stub_getMidiControllerForPort(LADSPA_Handle instance,
+ unsigned long port);
+ static void stub_runSynth(LADSPA_Handle instance,
+ unsigned long sample_count,
+ snd_seq_event_t *events,
+ unsigned long event_count);
+
+ /*
+ * LADSPA member functions
+ */
+ static LADSPA_Handle instantiate(const LADSPA_Descriptor *descriptor,
+ unsigned long s_rate);
+ void connectPort(unsigned long port, LADSPA_Data *data);
+ void activate();
+ void run(unsigned long sample_count);
+ void deactivate();
+ void cleanup();
+ static const LADSPA_Descriptor *getLadspaDescriptor(unsigned long index);
+
+ /*
+ * DSSI member functions
+ */
+ const DSSI_Program_Descriptor *getProgram(unsigned long Index);
+ void selectProgram(unsigned long bank, unsigned long program);
+ int getMidiControllerForPort(unsigned long port);
+ void runSynth(unsigned long sample_count,
+ snd_seq_event_t *events,
+ unsigned long event_count);
+ static const DSSI_Descriptor *getDssiDescriptor(unsigned long index);
+
+ struct ProgramDescriptor {
+ unsigned long bank;
+ unsigned long program;
+ std::string name;
+ ProgramDescriptor(unsigned long _bank,
+ unsigned long _program,
+ char *_name);
+ };
+
+ private:
+
+ DSSIaudiooutput(unsigned long sampleRate);
+ ~DSSIaudiooutput();
+ static DSSI_Descriptor *initDssiDescriptor();
+ static DSSIaudiooutput *getInstance(LADSPA_Handle instance);
+ void initBanks();
+ bool mapNextBank();
+
+ LADSPA_Data *outl;
+ LADSPA_Data *outr;
+ long sampleRate;
+ Master *master;
+ static DSSI_Descriptor *dssiDescriptor;
+ static std::string bankDirNames[];
+ static
+ std::vector<ProgramDescriptor> programMap;
+
+ /**
+ * Flag controlling the list of bank directories
+ */
+ bool banksInited;
+
+ static
+ long bankNoToMap;
};
#endif
-
diff --git a/src/Params/ADnoteParameters.cpp b/src/Params/ADnoteParameters.cpp
@@ -53,17 +53,17 @@ ADnoteGlobalParam::ADnoteGlobalParam()
{
FreqEnvelope = new EnvelopeParams(0, 0);
FreqEnvelope->ASRinit(64, 50, 64, 60);
- FreqLfo = new LFOParams(70, 0, 64, 0, 0, 0, 0, 0);
+ FreqLfo = new LFOParams(70, 0, 64, 0, 0, 0, 0, 0);
- AmpEnvelope = new EnvelopeParams(64, 1);
+ AmpEnvelope = new EnvelopeParams(64, 1);
AmpEnvelope->ADSRinit_dB(0, 40, 127, 25);
AmpLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 1);
GlobalFilter = new FilterParams(2, 94, 40);
FilterEnvelope = new EnvelopeParams(0, 1);
FilterEnvelope->ADSRinit_filter(64, 40, 64, 70, 60, 64);
- FilterLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 2);
- Reson = new Resonance();
+ FilterLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 2);
+ Reson = new Resonance();
}
void ADnoteParameters::defaults()
@@ -98,7 +98,7 @@ void ADnoteGlobalParam::defaults()
PPunchTime = 60;
PPunchStretch = 64;
PPunchVelocitySensing = 72;
- Hrandgrouping = 0;
+ Hrandgrouping = 0;
/* Filter Global Parameters*/
PFilterVelocityScale = 64;
@@ -119,14 +119,14 @@ void ADnoteParameters::defaults(int n)
void ADnoteVoiceParam::defaults()
{
- Enabled = 0;
+ Enabled = 0;
Unison_size = 1;
Unison_frequency_spread = 60;
Unison_stereo_spread = 64;
Unison_vibratto = 64;
- Unison_vibratto_speed = 64;
- Unison_invert_phase = 0;
+ Unison_vibratto_speed = 64;
+ Unison_invert_phase = 0;
Type = 0;
Pfixedfreq = 0;
@@ -137,14 +137,14 @@ void ADnoteVoiceParam::defaults()
PextFMoscil = -1;
Poscilphase = 64;
PFMoscilphase = 64;
- PDelay = 0;
- PVolume = 100;
- PVolumeminus = 0;
- PPanning = 64; //center
- PDetune = 8192; //8192=0
- PCoarseDetune = 0;
- PDetuneType = 0;
- PFreqLfoEnabled = 0;
+ PDelay = 0;
+ PVolume = 100;
+ PVolumeminus = 0;
+ PPanning = 64; //center
+ PDetune = 8192; //8192=0
+ PCoarseDetune = 0;
+ PDetuneType = 0;
+ PFreqLfoEnabled = 0;
PFreqEnvelopeEnabled = 0;
PAmpEnvelopeEnabled = 0;
PAmpLfoEnabled = 0;
@@ -155,9 +155,9 @@ void ADnoteVoiceParam::defaults()
PFMEnabled = 0;
//I use the internal oscillator (-1)
- PFMVoice = -1;
+ PFMVoice = -1;
- PFMVolume = 90;
+ PFMVolume = 90;
PFMVolumeDamp = 64;
PFMDetune = 8192;
PFMCoarseDetune = 0;
@@ -198,22 +198,22 @@ void ADnoteVoiceParam::enable(FFTwrapper *fft, Resonance *Reson)
OscilSmp = new OscilGen(fft, Reson);
FMSmp = new OscilGen(fft, NULL);
- AmpEnvelope = new EnvelopeParams(64, 1);
+ AmpEnvelope = new EnvelopeParams(64, 1);
AmpEnvelope->ADSRinit_dB(0, 100, 127, 100);
- AmpLfo = new LFOParams(90, 32, 64, 0, 0, 30, 0, 1);
+ AmpLfo = new LFOParams(90, 32, 64, 0, 0, 30, 0, 1);
- FreqEnvelope = new EnvelopeParams(0, 0);
+ FreqEnvelope = new EnvelopeParams(0, 0);
FreqEnvelope->ASRinit(30, 40, 64, 60);
- FreqLfo = new LFOParams(50, 40, 0, 0, 0, 0, 0, 0);
+ FreqLfo = new LFOParams(50, 40, 0, 0, 0, 0, 0, 0);
VoiceFilter = new FilterParams(2, 50, 60);
FilterEnvelope = new EnvelopeParams(0, 0);
FilterEnvelope->ADSRinit_filter(90, 70, 40, 70, 10, 40);
- FilterLfo = new LFOParams(50, 20, 64, 0, 0, 0, 0, 2);
+ FilterLfo = new LFOParams(50, 20, 64, 0, 0, 0, 0, 2);
FMFreqEnvelope = new EnvelopeParams(0, 0);
FMFreqEnvelope->ASRinit(20, 90, 40, 80);
- FMAmpEnvelope = new EnvelopeParams(64, 1);
+ FMAmpEnvelope = new EnvelopeParams(64, 1);
FMAmpEnvelope->ADSRinit(80, 90, 127, 100);
}
@@ -285,7 +285,7 @@ ADnoteParameters::~ADnoteParameters()
}
int ADnoteParameters::get_unison_size_index(int nvoice) {
- int index = 0;
+ int index = 0;
if(nvoice >= NUM_VOICES)
return 0;
int unison = VoicePar[nvoice].Unison_size;
@@ -337,7 +337,7 @@ void ADnoteParameters::add2XMLsection(XMLwrapper *xml, int n)
&& (fmoscilused == 0)) && (xml->minimal))
return;
- VoicePar[nvoice].add2XML(xml,fmoscilused);
+ VoicePar[nvoice].add2XML(xml, fmoscilused);
}
void ADnoteVoiceParam::add2XML(XMLwrapper *xml, bool fmoscilused)
@@ -560,15 +560,15 @@ void ADnoteGlobalParam::getfromXML(XMLwrapper *xml)
PVolume = xml->getpar127("volume", PVolume);
PPanning = xml->getpar127("panning", PPanning);
PAmpVelocityScaleFunction = xml->getpar127("velocity_sensing",
- PAmpVelocityScaleFunction);
+ PAmpVelocityScaleFunction);
PPunchStrength = xml->getpar127("punch_strength", PPunchStrength);
PPunchTime = xml->getpar127("punch_time", PPunchTime);
PPunchStretch = xml->getpar127("punch_stretch", PPunchStretch);
PPunchVelocitySensing = xml->getpar127("punch_velocity_sensing",
- PPunchVelocitySensing);
- Hrandgrouping = xml->getpar127("harmonic_randomness_grouping",
- Hrandgrouping);
+ PPunchVelocitySensing);
+ Hrandgrouping = xml->getpar127("harmonic_randomness_grouping",
+ Hrandgrouping);
if(xml->enterbranch("AMPLITUDE_ENVELOPE")) {
AmpEnvelope->getfromXML(xml);
@@ -603,8 +603,9 @@ void ADnoteGlobalParam::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("FILTER_PARAMETERS")) {
PFilterVelocityScale = xml->getpar127("velocity_sensing_amplitude",
- PFilterVelocityScale);
- PFilterVelocityScaleFunction = xml->getpar127("velocity_sensing",
+ PFilterVelocityScale);
+ PFilterVelocityScaleFunction = xml->getpar127(
+ "velocity_sensing",
PFilterVelocityScaleFunction);
xml->enterbranch("FILTER");
@@ -654,20 +655,22 @@ void ADnoteVoiceParam::getfromXML(XMLwrapper *xml, unsigned nvoice)
{
Enabled = xml->getparbool("enabled", 0);
Unison_size = xml->getpar127("unison_size", Unison_size);
- Unison_frequency_spread = xml->getpar127( "unison_frequency_spread",
- Unison_frequency_spread);
- Unison_stereo_spread = xml->getpar127( "unison_stereo_spread",
- Unison_stereo_spread);
- Unison_vibratto = xml->getpar127( "unison_vibratto", Unison_vibratto);
- Unison_vibratto_speed = xml->getpar127( "unison_vibratto_speed", Unison_vibratto_speed);
- Unison_invert_phase = xml->getpar127( "unison_invert_phase", Unison_invert_phase);
-
- Type = xml->getpar127("type", Type);
- PDelay = xml->getpar127("delay", PDelay);
+ Unison_frequency_spread = xml->getpar127("unison_frequency_spread",
+ Unison_frequency_spread);
+ Unison_stereo_spread = xml->getpar127("unison_stereo_spread",
+ Unison_stereo_spread);
+ Unison_vibratto = xml->getpar127("unison_vibratto", Unison_vibratto);
+ Unison_vibratto_speed = xml->getpar127("unison_vibratto_speed",
+ Unison_vibratto_speed);
+ Unison_invert_phase = xml->getpar127("unison_invert_phase",
+ Unison_invert_phase);
+
+ Type = xml->getpar127("type", Type);
+ PDelay = xml->getpar127("delay", PDelay);
Presonance = xml->getparbool("resonance", Presonance);
- Pextoscil = xml->getpar("ext_oscil", -1, -1, nvoice - 1);
- PextFMoscil = xml->getpar("ext_fm_oscil", -1, -1, nvoice - 1);
+ Pextoscil = xml->getpar("ext_oscil", -1, -1, nvoice - 1);
+ PextFMoscil = xml->getpar("ext_fm_oscil", -1, -1, nvoice - 1);
Poscilphase = xml->getpar127("oscil_phase", Poscilphase);
PFMoscilphase = xml->getpar127("oscil_fm_phase", PFMoscilphase);
@@ -685,9 +688,11 @@ void ADnoteVoiceParam::getfromXML(XMLwrapper *xml, unsigned nvoice)
PPanning = xml->getpar127("panning", PPanning);
PVolume = xml->getpar127("volume", PVolume);
PVolumeminus = xml->getparbool("volume_minus", PVolumeminus);
- PAmpVelocityScaleFunction = xml->getpar127("velocity_sensing", PAmpVelocityScaleFunction);
+ PAmpVelocityScaleFunction = xml->getpar127("velocity_sensing",
+ PAmpVelocityScaleFunction);
- PAmpEnvelopeEnabled = xml->getparbool("amp_envelope_enabled", PAmpEnvelopeEnabled);
+ PAmpEnvelopeEnabled = xml->getparbool("amp_envelope_enabled",
+ PAmpEnvelopeEnabled);
if(xml->enterbranch("AMPLITUDE_ENVELOPE")) {
AmpEnvelope->getfromXML(xml);
xml->exitbranch();
@@ -702,13 +707,13 @@ void ADnoteVoiceParam::getfromXML(XMLwrapper *xml, unsigned nvoice)
}
if(xml->enterbranch("FREQUENCY_PARAMETERS")) {
- Pfixedfreq = xml->getparbool("fixed_freq", Pfixedfreq);
- PfixedfreqET = xml->getpar127("fixed_freq_et", PfixedfreqET);
- PDetune = xml->getpar("detune", PDetune, 0, 16383);
+ Pfixedfreq = xml->getparbool("fixed_freq", Pfixedfreq);
+ PfixedfreqET = xml->getpar127("fixed_freq_et", PfixedfreqET);
+ PDetune = xml->getpar("detune", PDetune, 0, 16383);
PCoarseDetune = xml->getpar("coarse_detune", PCoarseDetune, 0, 16383);
- PDetuneType = xml->getpar127( "detune_type", PDetuneType);
- PFreqEnvelopeEnabled = xml->getparbool( "freq_envelope_enabled",
- PFreqEnvelopeEnabled);
+ PDetuneType = xml->getpar127("detune_type", PDetuneType);
+ PFreqEnvelopeEnabled = xml->getparbool("freq_envelope_enabled",
+ PFreqEnvelopeEnabled);
if(xml->enterbranch("FREQUENCY_ENVELOPE")) {
FreqEnvelope->getfromXML(xml);
@@ -730,13 +735,15 @@ void ADnoteVoiceParam::getfromXML(XMLwrapper *xml, unsigned nvoice)
xml->exitbranch();
}
- PFilterEnvelopeEnabled = xml->getparbool("filter_envelope_enabled", PFilterEnvelopeEnabled);
+ PFilterEnvelopeEnabled = xml->getparbool("filter_envelope_enabled",
+ PFilterEnvelopeEnabled);
if(xml->enterbranch("FILTER_ENVELOPE")) {
FilterEnvelope->getfromXML(xml);
xml->exitbranch();
}
- PFilterLfoEnabled = xml->getparbool("filter_lfo_enabled", PFilterLfoEnabled);
+ PFilterLfoEnabled = xml->getparbool("filter_lfo_enabled",
+ PFilterLfoEnabled);
if(xml->enterbranch("FILTER_LFO")) {
FilterLfo->getfromXML(xml);
xml->exitbranch();
@@ -745,12 +752,14 @@ void ADnoteVoiceParam::getfromXML(XMLwrapper *xml, unsigned nvoice)
}
if(xml->enterbranch("FM_PARAMETERS")) {
- PFMVoice = xml->getpar("input_voice", PFMVoice, -1, nvoice - 1);
+ PFMVoice = xml->getpar("input_voice", PFMVoice, -1, nvoice - 1);
PFMVolume = xml->getpar127("volume", PFMVolume);
PFMVolumeDamp = xml->getpar127("volume_damp", PFMVolumeDamp);
- PFMVelocityScaleFunction = xml->getpar127("velocity_sensing", PFMVelocityScaleFunction);
+ PFMVelocityScaleFunction = xml->getpar127("velocity_sensing",
+ PFMVelocityScaleFunction);
- PFMAmpEnvelopeEnabled = xml->getparbool("amp_envelope_enabled", PFMAmpEnvelopeEnabled);
+ PFMAmpEnvelopeEnabled = xml->getparbool("amp_envelope_enabled",
+ PFMAmpEnvelopeEnabled);
if(xml->enterbranch("AMPLITUDE_ENVELOPE")) {
FMAmpEnvelope->getfromXML(xml);
xml->exitbranch();
@@ -758,11 +767,14 @@ void ADnoteVoiceParam::getfromXML(XMLwrapper *xml, unsigned nvoice)
if(xml->enterbranch("MODULATOR")) {
PFMDetune = xml->getpar("detune", PFMDetune, 0, 16383);
- PFMCoarseDetune = xml->getpar("coarse_detune", PFMCoarseDetune, 0, 16383);
- PFMDetuneType = xml->getpar127("detune_type", PFMDetuneType);
+ PFMCoarseDetune = xml->getpar("coarse_detune",
+ PFMCoarseDetune,
+ 0,
+ 16383);
+ PFMDetuneType = xml->getpar127("detune_type", PFMDetuneType);
PFMFreqEnvelopeEnabled = xml->getparbool("freq_envelope_enabled",
- PFMFreqEnvelopeEnabled);
+ PFMFreqEnvelopeEnabled);
if(xml->enterbranch("FREQUENCY_ENVELOPE")) {
FMFreqEnvelope->getfromXML(xml);
xml->exitbranch();
@@ -778,4 +790,3 @@ void ADnoteVoiceParam::getfromXML(XMLwrapper *xml, unsigned nvoice)
xml->exitbranch();
}
}
-
diff --git a/src/Params/ADnoteParameters.h b/src/Params/ADnoteParameters.h
@@ -64,9 +64,9 @@ struct ADnoteGlobalParam {
unsigned short int PCoarseDetune; //coarse detune+octave
unsigned char PDetuneType; //detune type
- unsigned char PBandwidth; //how much the relative fine detunes of the voices are changed
+ unsigned char PBandwidth; //how much the relative fine detunes of the voices are changed
- EnvelopeParams *FreqEnvelope; //Frequency Envelope
+ EnvelopeParams *FreqEnvelope; //Frequency Envelope
LFOParams *FreqLfo; //Frequency LFO
@@ -78,18 +78,18 @@ struct ADnoteGlobalParam {
1 - left
64 - center
127 - right */
- unsigned char PPanning;
+ unsigned char PPanning;
- unsigned char PVolume;
+ unsigned char PVolume;
- unsigned char PAmpVelocityScaleFunction;
+ unsigned char PAmpVelocityScaleFunction;
EnvelopeParams *AmpEnvelope;
- LFOParams *AmpLfo;
+ LFOParams *AmpLfo;
- unsigned char PPunchStrength, PPunchTime, PPunchStretch,
- PPunchVelocitySensing;
+ unsigned char PPunchStrength, PPunchTime, PPunchStretch,
+ PPunchVelocitySensing;
/******************************************
* FILTER GLOBAL PARAMETERS *
@@ -100,11 +100,11 @@ struct ADnoteGlobalParam {
unsigned char PFilterVelocityScale;
// filter velocity sensing
- unsigned char PFilterVelocityScaleFunction;
+ unsigned char PFilterVelocityScaleFunction;
EnvelopeParams *FilterEnvelope;
- LFOParams *FilterLfo;
+ LFOParams *FilterLfo;
// RESONANCE
Resonance *Reson;
@@ -120,7 +120,7 @@ struct ADnoteGlobalParam {
/***********************************************************/
struct ADnoteVoiceParam {
void getfromXML(XMLwrapper *xml, unsigned nvoice);
- void add2XML(XMLwrapper *xml,bool fmoscilused);
+ void add2XML(XMLwrapper *xml, bool fmoscilused);
void defaults();
void enable(FFTwrapper *fft, Resonance *Reson);
void kill();
@@ -285,7 +285,7 @@ struct ADnoteVoiceParam {
EnvelopeParams *FMAmpEnvelope;
};
-class ADnoteParameters : public PresetsArray
+class ADnoteParameters:public PresetsArray
{
public:
ADnoteParameters(FFTwrapper *fft_);
@@ -314,4 +314,3 @@ class ADnoteParameters : public PresetsArray
};
#endif
-
diff --git a/src/Params/Controller.cpp b/src/Params/Controller.cpp
@@ -51,17 +51,17 @@ void Controller::defaults()
portamento.portamento = 0;
portamento.used = 0;
- portamento.proportional = 0;
- portamento.propRate = 80;
- portamento.propDepth = 90;
- portamento.receive = 1;
+ portamento.proportional = 0;
+ portamento.propRate = 80;
+ portamento.propDepth = 90;
+ portamento.receive = 1;
portamento.time = 64;
portamento.updowntimestretch = 64;
- portamento.pitchthresh = 3;
- portamento.pitchthreshtype = 1;
- portamento.noteusing = -1;
- resonancecenter.depth = 64;
- resonancebandwidth.depth = 64;
+ portamento.pitchthresh = 3;
+ portamento.pitchthreshtype = 1;
+ portamento.noteusing = -1;
+ resonancecenter.depth = 64;
+ resonancebandwidth.depth = 64;
initportamento(440.0f, 440.0f, false); // Now has a third argument
setportamento(0);
@@ -152,7 +152,8 @@ void Controller::setmodwheel(int value)
{
modwheel.data = value;
if(modwheel.exponential == 0) {
- float tmp = powf(25.0f, powf(modwheel.depth / 127.0f, 1.5f) * 2.0f) / 25.0f;
+ float tmp =
+ powf(25.0f, powf(modwheel.depth / 127.0f, 1.5f) * 2.0f) / 25.0f;
if((value < 64) && (modwheel.depth >= 64))
tmp = 1.0f;
modwheel.relmod = (value / 64.0f - 1.0f) * tmp + 1.0f;
@@ -222,23 +223,27 @@ int Controller::initportamento(float oldfreq,
//Linear functors could also make this nicer
if(oldfreq > newfreq) //2 is the center of propRate
portamentotime *=
- powf(oldfreq / newfreq / (portamento.propRate / 127.0f * 3 + .05),
- (portamento.propDepth / 127.0f * 1.6f + .2));
+ powf(oldfreq / newfreq
+ / (portamento.propRate / 127.0f * 3 + .05),
+ (portamento.propDepth / 127.0f * 1.6f + .2));
else //1 is the center of propDepth
portamentotime *=
- powf(newfreq / oldfreq / (portamento.propRate / 127.0f * 3 + .05),
- (portamento.propDepth / 127.0f * 1.6f + .2));
+ powf(newfreq / oldfreq
+ / (portamento.propRate / 127.0f * 3 + .05),
+ (portamento.propDepth / 127.0f * 1.6f + .2));
}
if((portamento.updowntimestretch >= 64) && (newfreq < oldfreq)) {
if(portamento.updowntimestretch == 127)
return 0;
- portamentotime *= powf(0.1f, (portamento.updowntimestretch - 64) / 63.0f);
+ portamentotime *= powf(0.1f,
+ (portamento.updowntimestretch - 64) / 63.0f);
}
if((portamento.updowntimestretch < 64) && (newfreq > oldfreq)) {
if(portamento.updowntimestretch == 0)
return 0;
- portamentotime *= powf(0.1f, (64.0f - portamento.updowntimestretch) / 64.0f);
+ portamentotime *= powf(0.1f,
+ (64.0f - portamento.updowntimestretch) / 64.0f);
}
//printf("%f->%f : Time %f\n",oldfreq,newfreq,portamentotime);
@@ -247,8 +252,8 @@ int Controller::initportamento(float oldfreq,
portamento.origfreqrap = oldfreq / newfreq;
float tmprap = ((portamento.origfreqrap > 1.0f) ?
- (portamento.origfreqrap) :
- (1.0f / portamento.origfreqrap));
+ (portamento.origfreqrap) :
+ (1.0f / portamento.origfreqrap));
float thresholdrap = powf(2.0f, portamento.pitchthresh / 12.0f);
if((portamento.pitchthreshtype == 0) && (tmprap - 0.00001f > thresholdrap))
@@ -310,24 +315,24 @@ int Controller::getnrpn(int *parhi, int *parlo, int *valhi, int *vallo)
void Controller::setparameternumber(unsigned int type, int value)
{
switch(type) {
- case C_nrpnhi:
- NRPN.parhi = value;
- NRPN.valhi = -1;
- NRPN.vallo = -1; //clear the values
- break;
- case C_nrpnlo:
- NRPN.parlo = value;
- NRPN.valhi = -1;
- NRPN.vallo = -1; //clear the values
- break;
- case C_dataentryhi:
- if((NRPN.parhi >= 0) && (NRPN.parlo >= 0))
- NRPN.valhi = value;
- break;
- case C_dataentrylo:
- if((NRPN.parhi >= 0) && (NRPN.parlo >= 0))
- NRPN.vallo = value;
- break;
+ case C_nrpnhi:
+ NRPN.parhi = value;
+ NRPN.valhi = -1;
+ NRPN.vallo = -1; //clear the values
+ break;
+ case C_nrpnlo:
+ NRPN.parlo = value;
+ NRPN.valhi = -1;
+ NRPN.vallo = -1; //clear the values
+ break;
+ case C_dataentryhi:
+ if((NRPN.parhi >= 0) && (NRPN.parlo >= 0))
+ NRPN.valhi = value;
+ break;
+ case C_dataentrylo:
+ if((NRPN.parhi >= 0) && (NRPN.parlo >= 0))
+ NRPN.vallo = value;
+ break;
}
}
@@ -369,14 +374,14 @@ void Controller::getfromXML(XMLwrapper *xml)
-6400,
6400);
- expression.receive = xml->getparbool("expression_receive",
- expression.receive);
- panning.depth = xml->getpar127("panning_depth", panning.depth);
- filtercutoff.depth = xml->getpar127("filter_cutoff_depth",
- filtercutoff.depth);
- filterq.depth = xml->getpar127("filter_q_depth", filterq.depth);
+ expression.receive = xml->getparbool("expression_receive",
+ expression.receive);
+ panning.depth = xml->getpar127("panning_depth", panning.depth);
+ filtercutoff.depth = xml->getpar127("filter_cutoff_depth",
+ filtercutoff.depth);
+ filterq.depth = xml->getpar127("filter_q_depth", filterq.depth);
bandwidth.depth = xml->getpar127("bandwidth_depth", bandwidth.depth);
- modwheel.depth = xml->getpar127("mod_wheel_depth", modwheel.depth);
+ modwheel.depth = xml->getpar127("mod_wheel_depth", modwheel.depth);
modwheel.exponential = xml->getparbool("mod_wheel_exponential",
modwheel.exponential);
fmamp.receive = xml->getparbool("fm_amp_receive",
@@ -392,24 +397,23 @@ void Controller::getfromXML(XMLwrapper *xml)
portamento.time);
portamento.pitchthresh = xml->getpar127("portamento_pitchthresh",
portamento.pitchthresh);
- portamento.pitchthreshtype = xml->getpar127("portamento_pitchthreshtype",
- portamento.pitchthreshtype);
- portamento.portamento = xml->getpar127("portamento_portamento",
- portamento.portamento);
+ portamento.pitchthreshtype = xml->getpar127("portamento_pitchthreshtype",
+ portamento.pitchthreshtype);
+ portamento.portamento = xml->getpar127("portamento_portamento",
+ portamento.portamento);
portamento.updowntimestretch = xml->getpar127(
"portamento_updowntimestretch",
portamento.updowntimestretch);
- portamento.proportional = xml->getpar127("portamento_proportional",
- portamento.proportional);
- portamento.propRate = xml->getpar127("portamento_proprate",
- portamento.propRate);
- portamento.propDepth = xml->getpar127("portamento_propdepth",
- portamento.propDepth);
+ portamento.proportional = xml->getpar127("portamento_proportional",
+ portamento.proportional);
+ portamento.propRate = xml->getpar127("portamento_proprate",
+ portamento.propRate);
+ portamento.propDepth = xml->getpar127("portamento_propdepth",
+ portamento.propDepth);
- resonancecenter.depth = xml->getpar127("resonance_center_depth",
- resonancecenter.depth);
+ resonancecenter.depth = xml->getpar127("resonance_center_depth",
+ resonancecenter.depth);
resonancebandwidth.depth = xml->getpar127("resonance_bandwidth_depth",
resonancebandwidth.depth);
}
-
diff --git a/src/Params/Controller.h b/src/Params/Controller.h
@@ -74,59 +74,59 @@ class Controller
// Controllers values
struct { //Pitch Wheel
- int data;
+ int data;
short int bendrange; //bendrange is in cents
- float relfreq; //the relative frequency (default is 1.0f)
+ float relfreq; //the relative frequency (default is 1.0f)
} pitchwheel;
struct { //Expression
- int data;
- float relvolume;
+ int data;
+ float relvolume;
unsigned char receive;
} expression;
struct { //Panning
- int data;
- float pan;
+ int data;
+ float pan;
unsigned char depth;
} panning;
struct { //Filter cutoff
- int data;
- float relfreq;
+ int data;
+ float relfreq;
unsigned char depth;
} filtercutoff;
struct { //Filter Q
- int data;
- float relq;
+ int data;
+ float relq;
unsigned char depth;
} filterq;
struct { //Bandwidth
- int data;
- float relbw;
+ int data;
+ float relbw;
unsigned char depth;
unsigned char exponential;
} bandwidth;
struct { //Modulation Wheel
- int data;
- float relmod;
+ int data;
+ float relmod;
unsigned char depth;
unsigned char exponential;
} modwheel;
struct { //FM amplitude
- int data;
- float relamp;
+ int data;
+ float relamp;
unsigned char receive;
} fmamp;
struct { //Volume
- int data;
- float volume;
+ int data;
+ float volume;
unsigned char receive;
} volume;
@@ -179,7 +179,7 @@ class Controller
* This will be linear with respect to x.*/
float freqrap;
/**this is used by the Part for knowing which note uses the portamento*/
- int noteusing;
+ int noteusing;
/**if a the portamento is used by a note
* \todo see if this can be a bool*/
int used;
@@ -195,14 +195,14 @@ class Controller
} portamento;
struct { //Resonance Center Frequency
- int data;
- float relcenter;
+ int data;
+ float relcenter;
unsigned char depth;
} resonancecenter;
struct { //Resonance Bandwidth
- int data;
- float relbw;
+ int data;
+ float relbw;
unsigned char depth;
} resonancebandwidth;
@@ -218,4 +218,3 @@ class Controller
};
#endif
-
diff --git a/src/Params/EnvelopeParams.cpp b/src/Params/EnvelopeParams.cpp
@@ -43,7 +43,7 @@ EnvelopeParams::EnvelopeParams(unsigned char Penvstretch_,
Penvdt[i] = 32;
Penvval[i] = 64;
}
- Penvdt[0] = 0; //no used
+ Penvdt[0] = 0; //no used
Penvsustain = 1;
Penvpoints = 1;
Envmode = 1;
@@ -149,57 +149,57 @@ void EnvelopeParams::ASRinit_bw(char A_val, char A_dt, char R_val, char R_dt)
void EnvelopeParams::converttofree()
{
switch(Envmode) {
- case 1:
- Penvpoints = 4;
- Penvsustain = 2;
- Penvval[0] = 0;
- Penvdt[1] = PA_dt;
- Penvval[1] = 127;
- Penvdt[2] = PD_dt;
- Penvval[2] = PS_val;
- Penvdt[3] = PR_dt;
- Penvval[3] = 0;
- break;
- case 2:
- Penvpoints = 4;
- Penvsustain = 2;
- Penvval[0] = 0;
- Penvdt[1] = PA_dt;
- Penvval[1] = 127;
- Penvdt[2] = PD_dt;
- Penvval[2] = PS_val;
- Penvdt[3] = PR_dt;
- Penvval[3] = 0;
- break;
- case 3:
- Penvpoints = 3;
- Penvsustain = 1;
- Penvval[0] = PA_val;
- Penvdt[1] = PA_dt;
- Penvval[1] = 64;
- Penvdt[2] = PR_dt;
- Penvval[2] = PR_val;
- break;
- case 4:
- Penvpoints = 4;
- Penvsustain = 2;
- Penvval[0] = PA_val;
- Penvdt[1] = PA_dt;
- Penvval[1] = PD_val;
- Penvdt[2] = PD_dt;
- Penvval[2] = 64;
- Penvdt[3] = PR_dt;
- Penvval[3] = PR_val;
- break;
- case 5:
- Penvpoints = 3;
- Penvsustain = 1;
- Penvval[0] = PA_val;
- Penvdt[1] = PA_dt;
- Penvval[1] = 64;
- Penvdt[2] = PR_dt;
- Penvval[2] = PR_val;
- break;
+ case 1:
+ Penvpoints = 4;
+ Penvsustain = 2;
+ Penvval[0] = 0;
+ Penvdt[1] = PA_dt;
+ Penvval[1] = 127;
+ Penvdt[2] = PD_dt;
+ Penvval[2] = PS_val;
+ Penvdt[3] = PR_dt;
+ Penvval[3] = 0;
+ break;
+ case 2:
+ Penvpoints = 4;
+ Penvsustain = 2;
+ Penvval[0] = 0;
+ Penvdt[1] = PA_dt;
+ Penvval[1] = 127;
+ Penvdt[2] = PD_dt;
+ Penvval[2] = PS_val;
+ Penvdt[3] = PR_dt;
+ Penvval[3] = 0;
+ break;
+ case 3:
+ Penvpoints = 3;
+ Penvsustain = 1;
+ Penvval[0] = PA_val;
+ Penvdt[1] = PA_dt;
+ Penvval[1] = 64;
+ Penvdt[2] = PR_dt;
+ Penvval[2] = PR_val;
+ break;
+ case 4:
+ Penvpoints = 4;
+ Penvsustain = 2;
+ Penvval[0] = PA_val;
+ Penvdt[1] = PA_dt;
+ Penvval[1] = PD_val;
+ Penvdt[2] = PD_dt;
+ Penvval[2] = 64;
+ Penvdt[3] = PR_dt;
+ Penvval[3] = PR_val;
+ break;
+ case 5:
+ Penvpoints = 3;
+ Penvsustain = 1;
+ Penvval[0] = PA_val;
+ Penvdt[1] = PA_dt;
+ Penvval[1] = 64;
+ Penvdt[2] = PR_dt;
+ Penvval[2] = PR_val;
+ break;
}
}
@@ -222,7 +222,7 @@ void EnvelopeParams::add2XML(XMLwrapper *xml)
xml->addpar("S_val", PS_val);
xml->addpar("R_val", PR_val);
- if((Pfreemode != 0) || (!xml->minimal)) {
+ if((Pfreemode != 0) || (!xml->minimal))
for(int i = 0; i < Penvpoints; ++i) {
xml->beginbranch("POINT", i);
if(i != 0)
@@ -230,14 +230,13 @@ void EnvelopeParams::add2XML(XMLwrapper *xml)
xml->addpar("val", Penvval[i]);
xml->endbranch();
}
- }
}
void EnvelopeParams::getfromXML(XMLwrapper *xml)
{
- Pfreemode = xml->getparbool("free_mode", Pfreemode);
+ Pfreemode = xml->getparbool("free_mode", Pfreemode);
Penvpoints = xml->getpar127("env_points", Penvpoints);
Penvsustain = xml->getpar127("env_sustain", Penvsustain);
Penvstretch = xml->getpar127("env_stretch", Penvstretch);
@@ -295,4 +294,3 @@ void EnvelopeParams::store2defaults()
DS_val = PS_val;
DR_val = PR_val;
}
-
diff --git a/src/Params/EnvelopeParams.h b/src/Params/EnvelopeParams.h
@@ -87,4 +87,3 @@ class EnvelopeParams:public Presets
};
#endif
-
diff --git a/src/Params/FilterParams.cpp b/src/Params/FilterParams.cpp
@@ -29,12 +29,12 @@
FilterParams::FilterParams(unsigned char Ptype_,
unsigned char Pfreq_,
unsigned char Pq_)
- : PresetsArray()
+ :PresetsArray()
{
setpresettype("Pfilter");
- Dtype = Ptype_;
- Dfreq = Pfreq_;
- Dq = Pq_;
+ Dtype = Ptype_;
+ Dfreq = Pfreq_;
+ Dq = Pq_;
changed = false;
defaults();
@@ -46,14 +46,14 @@ FilterParams::~FilterParams()
void FilterParams::defaults()
{
- Ptype = Dtype;
- Pfreq = Dfreq;
- Pq = Dq;
+ Ptype = Dtype;
+ Pfreq = Dfreq;
+ Pq = Dq;
Pstages = 0;
Pfreqtrack = 64;
- Pgain = 64;
- Pcategory = 0;
+ Pgain = 64;
+ Pcategory = 0;
Pnumformants = 3;
Pformantslowness = 64;
@@ -67,9 +67,9 @@ void FilterParams::defaults()
Psequencestretch = 40;
Psequencereversed = 0;
- Pcenterfreq = 64; //1 kHz
- Poctavesfreq = 64;
- Pvowelclearness = 64;
+ Pcenterfreq = 64; //1 kHz
+ Poctavesfreq = 64;
+ Pvowelclearness = 64;
}
void FilterParams::defaults(int n)
@@ -94,24 +94,23 @@ void FilterParams::getfromFilterParams(FilterParams *pars)
if(pars == NULL)
return;
- Ptype = pars->Ptype;
- Pfreq = pars->Pfreq;
- Pq = pars->Pq;
+ Ptype = pars->Ptype;
+ Pfreq = pars->Pfreq;
+ Pq = pars->Pq;
Pstages = pars->Pstages;
Pfreqtrack = pars->Pfreqtrack;
- Pgain = pars->Pgain;
- Pcategory = pars->Pcategory;
+ Pgain = pars->Pgain;
+ Pcategory = pars->Pcategory;
Pnumformants = pars->Pnumformants;
Pformantslowness = pars->Pformantslowness;
- for(int j = 0; j < FF_MAX_VOWELS; ++j) {
+ for(int j = 0; j < FF_MAX_VOWELS; ++j)
for(int i = 0; i < FF_MAX_FORMANTS; ++i) {
Pvowels[j].formants[i].freq = pars->Pvowels[j].formants[i].freq;
Pvowels[j].formants[i].q = pars->Pvowels[j].formants[i].q;
Pvowels[j].formants[i].amp = pars->Pvowels[j].formants[i].amp;
}
- }
Psequencesize = pars->Psequencesize;
for(int i = 0; i < FF_MAX_SEQUENCE; ++i)
@@ -119,9 +118,9 @@ void FilterParams::getfromFilterParams(FilterParams *pars)
Psequencestretch = pars->Psequencestretch;
Psequencereversed = pars->Psequencereversed;
- Pcenterfreq = pars->Pcenterfreq;
- Poctavesfreq = pars->Poctavesfreq;
- Pvowelclearness = pars->Pvowelclearness;
+ Pcenterfreq = pars->Pcenterfreq;
+ Poctavesfreq = pars->Poctavesfreq;
+ Pvowelclearness = pars->Pvowelclearness;
}
@@ -202,7 +201,8 @@ void FilterParams::formantfilterH(int nvowel, int nfreqs, float *freqs)
filter_q = getformantq(Pvowels[nvowel].formants[nformant].q) * getq();
if(Pstages > 0)
filter_q =
- (filter_q > 1.0f ? powf(filter_q, 1.0f / (Pstages + 1)) : filter_q);
+ (filter_q >
+ 1.0f ? powf(filter_q, 1.0f / (Pstages + 1)) : filter_q);
filter_amp = getformantamp(Pvowels[nvowel].formants[nformant].amp);
@@ -213,11 +213,11 @@ void FilterParams::formantfilterH(int nvowel, int nfreqs, float *freqs)
cs = cosf(omega);
alpha = sn / (2 * filter_q);
float tmp = 1 + alpha;
- c[0] = alpha / tmp *sqrt(filter_q + 1);
- c[1] = 0;
- c[2] = -alpha / tmp *sqrt(filter_q + 1);
- d[1] = -2 * cs / tmp * (-1);
- d[2] = (1 - alpha) / tmp * (-1);
+ c[0] = alpha / tmp *sqrt(filter_q + 1);
+ c[1] = 0;
+ c[2] = -alpha / tmp *sqrt(filter_q + 1);
+ d[1] = -2 * cs / tmp * (-1);
+ d[2] = (1 - alpha) / tmp * (-1);
}
else
continue;
@@ -339,11 +339,11 @@ void FilterParams::getfromXMLsection(XMLwrapper *xml, int n)
"freq",
Pvowels[nvowel
].formants[nformant].freq);
- Pvowels[nvowel].formants[nformant].amp = xml->getpar127(
+ Pvowels[nvowel].formants[nformant].amp = xml->getpar127(
"amp",
Pvowels[nvowel
].formants[nformant].amp);
- Pvowels[nvowel].formants[nformant].q =
+ Pvowels[nvowel].formants[nformant].q =
xml->getpar127("q", Pvowels[nvowel].formants[nformant].q);
xml->exitbranch();
}
@@ -352,10 +352,10 @@ void FilterParams::getfromXMLsection(XMLwrapper *xml, int n)
void FilterParams::getfromXML(XMLwrapper *xml)
{
//filter parameters
- Pcategory = xml->getpar127("category", Pcategory);
- Ptype = xml->getpar127("type", Ptype);
- Pfreq = xml->getpar127("freq", Pfreq);
- Pq = xml->getpar127("q", Pq);
+ Pcategory = xml->getpar127("category", Pcategory);
+ Ptype = xml->getpar127("type", Ptype);
+ Pfreq = xml->getpar127("freq", Pfreq);
+ Pq = xml->getpar127("q", Pq);
Pstages = xml->getpar127("stages", Pstages);
Pfreqtrack = xml->getpar127("freq_track", Pfreqtrack);
Pgain = xml->getpar127("gain", Pgain);
@@ -390,4 +390,3 @@ void FilterParams::getfromXML(XMLwrapper *xml)
xml->exitbranch();
}
}
-
diff --git a/src/Params/FilterParams.h b/src/Params/FilterParams.h
@@ -27,7 +27,7 @@
#include "../Misc/XMLwrapper.h"
#include "PresetsArray.h"
-class FilterParams : public PresetsArray
+class FilterParams:public PresetsArray
{
public:
FilterParams(unsigned char Ptype_,
@@ -100,4 +100,3 @@ class FilterParams : public PresetsArray
};
#endif
-
diff --git a/src/Params/LFOParams.cpp b/src/Params/LFOParams.cpp
@@ -25,7 +25,7 @@
#include "../globals.h"
#include "LFOParams.h"
-int LFOParams::time;
+int LFOParams:: time;
LFOParams::LFOParams(char Pfreq_,
char Pintensity_,
@@ -37,17 +37,17 @@ LFOParams::LFOParams(char Pfreq_,
char fel_):Presets()
{
switch(fel_) {
- case 0:
- setpresettype("Plfofrequency");
- break;
- case 1:
- setpresettype("Plfoamplitude");
- break;
- case 2:
- setpresettype("Plfofilter");
- break;
+ case 0:
+ setpresettype("Plfofrequency");
+ break;
+ case 1:
+ setpresettype("Plfoamplitude");
+ break;
+ case 2:
+ setpresettype("Plfofilter");
+ break;
}
- Dfreq = Pfreq_;
+ Dfreq = Pfreq_;
Dintensity = Pintensity_;
Dstartphase = Pstartphase_;
DLFOtype = PLFOtype_;
@@ -65,7 +65,7 @@ LFOParams::~LFOParams()
void LFOParams::defaults()
{
- Pfreq = Dfreq / 127.0f;
+ Pfreq = Dfreq / 127.0f;
Pintensity = Dintensity;
Pstartphase = Dstartphase;
PLFOtype = DLFOtype;
@@ -92,7 +92,7 @@ void LFOParams::add2XML(XMLwrapper *xml)
void LFOParams::getfromXML(XMLwrapper *xml)
{
- Pfreq = xml->getparreal("freq", Pfreq, 0.0f, 1.0f);
+ Pfreq = xml->getparreal("freq", Pfreq, 0.0f, 1.0f);
Pintensity = xml->getpar127("intensity", Pintensity);
Pstartphase = xml->getpar127("start_phase", Pstartphase);
PLFOtype = xml->getpar127("lfo_type", PLFOtype);
@@ -102,4 +102,3 @@ void LFOParams::getfromXML(XMLwrapper *xml)
Pstretch = xml->getpar127("stretch", Pstretch);
Pcontinous = xml->getparbool("continous", Pcontinous);
}
-
diff --git a/src/Params/LFOParams.h b/src/Params/LFOParams.h
@@ -45,7 +45,7 @@ class LFOParams:public Presets
void getfromXML(XMLwrapper *xml);
/* MIDI Parameters*/
- float Pfreq; /**<frequency*/
+ float Pfreq; /**<frequency*/
unsigned char Pintensity; /**<intensity*/
unsigned char Pstartphase; /**<start phase (0=random)*/
unsigned char PLFOtype; /**<LFO type (sin,triangle,square,ramp,...)*/
@@ -69,4 +69,3 @@ class LFOParams:public Presets
};
#endif
-
diff --git a/src/Params/PADnoteParameters.cpp b/src/Params/PADnoteParameters.cpp
@@ -28,25 +28,25 @@ PADnoteParameters::PADnoteParameters(FFTwrapper *fft_,
{
setpresettype("Ppadsyth");
- fft = fft_;
- mutex = mutex_;
+ fft = fft_;
+ mutex = mutex_;
resonance = new Resonance();
oscilgen = new OscilGen(fft_, resonance);
oscilgen->ADvsPAD = true;
- FreqEnvelope = new EnvelopeParams(0, 0);
+ FreqEnvelope = new EnvelopeParams(0, 0);
FreqEnvelope->ASRinit(64, 50, 64, 60);
- FreqLfo = new LFOParams(70, 0, 64, 0, 0, 0, 0, 0);
+ FreqLfo = new LFOParams(70, 0, 64, 0, 0, 0, 0, 0);
- AmpEnvelope = new EnvelopeParams(64, 1);
+ AmpEnvelope = new EnvelopeParams(64, 1);
AmpEnvelope->ADSRinit_dB(0, 40, 127, 25);
- AmpLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 1);
+ AmpLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 1);
GlobalFilter = new FilterParams(2, 94, 40);
FilterEnvelope = new EnvelopeParams(0, 1);
FilterEnvelope->ADSRinit_filter(64, 40, 64, 70, 60, 64);
- FilterLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 2);
+ FilterLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 2);
for(int i = 0; i < PAD_MAX_SAMPLES; ++i)
sample[i].smp = NULL;
@@ -78,13 +78,13 @@ void PADnoteParameters::defaults()
Php.freqmult = 0;
Php.modulator.par1 = 0;
Php.modulator.freq = 30;
- Php.width = 127;
- Php.amp.type = 0;
- Php.amp.mode = 0;
- Php.amp.par1 = 80;
- Php.amp.par2 = 64;
- Php.autoscale = true;
- Php.onehalf = 0;
+ Php.width = 127;
+ Php.amp.type = 0;
+ Php.amp.mode = 0;
+ Php.amp.par1 = 80;
+ Php.amp.par2 = 64;
+ Php.autoscale = true;
+ Php.onehalf = 0;
setPbandwidth(500);
Pbwscale = 0;
@@ -99,8 +99,8 @@ void PADnoteParameters::defaults()
Pquality.samplesize = 3;
Pquality.basenote = 4;
- Pquality.oct = 3;
- Pquality.smpoct = 2;
+ Pquality.oct = 3;
+ Pquality.smpoct = 2;
PStereo = 1; //stereo
/* Frequency Global Parameters */
@@ -159,23 +159,24 @@ float PADnoteParameters::getprofile(float *smp, int size)
for(int i = 0; i < size; ++i)
smp[i] = 0.0f;
const int supersample = 16;
- float basepar = powf(2.0f, (1.0f - Php.base.par1 / 127.0f) * 12.0f);
- float freqmult = floor(powf(2.0f,
- Php.freqmult / 127.0f * 5.0f) + 0.000001f);
-
- float modfreq = floor(powf(2.0f,
- Php.modulator.freq / 127.0f
- * 5.0f) + 0.000001f);
- float modpar1 = powf(Php.modulator.par1 / 127.0f, 4.0f) * 5.0f / sqrt(
+ float basepar = powf(2.0f, (1.0f - Php.base.par1 / 127.0f) * 12.0f);
+ float freqmult = floor(powf(2.0f,
+ Php.freqmult / 127.0f
+ * 5.0f) + 0.000001f);
+
+ float modfreq = floor(powf(2.0f,
+ Php.modulator.freq / 127.0f
+ * 5.0f) + 0.000001f);
+ float modpar1 = powf(Php.modulator.par1 / 127.0f, 4.0f) * 5.0f / sqrt(
modfreq);
- float amppar1 =
+ float amppar1 =
powf(2.0f, powf(Php.amp.par1 / 127.0f, 2.0f) * 10.0f) - 0.999f;
- float amppar2 = (1.0f - Php.amp.par2 / 127.0f) * 0.998f + 0.001f;
- float width = powf(150.0f / (Php.width + 22.0f), 2.0f);
+ float amppar2 = (1.0f - Php.amp.par2 / 127.0f) * 0.998f + 0.001f;
+ float width = powf(150.0f / (Php.width + 22.0f), 2.0f);
for(int i = 0; i < size * supersample; ++i) {
- bool makezero = false;
- float x = i * 1.0f / (size * (float) supersample);
+ bool makezero = false;
+ float x = i * 1.0f / (size * (float) supersample);
float origx = x;
@@ -185,21 +186,20 @@ float PADnoteParameters::getprofile(float *smp, int size)
x = 0.0f;
makezero = true;
}
- else {
- if(x > 1.0f) {
- x = 1.0f;
- makezero = true;
- }
+ else
+ if(x > 1.0f) {
+ x = 1.0f;
+ makezero = true;
}
//compute the full profile or one half
switch(Php.onehalf) {
- case 1:
- x = x * 0.5f + 0.5f;
- break;
- case 2:
- x = x * 0.5f;
- break;
+ case 1:
+ x = x * 0.5f + 0.5f;
+ break;
+ case 2:
+ x = x * 0.5f;
+ break;
}
float x_before_freq_mult = x;
@@ -215,19 +215,19 @@ float PADnoteParameters::getprofile(float *smp, int size)
//this is the base function of the profile
float f;
switch(Php.base.type) {
- case 1:
- f = expf(-(x * x) * basepar);
- if(f < 0.4f)
- f = 0.0f;
- else
- f = 1.0f;
- break;
- case 2:
- f = expf(-(fabs(x)) * sqrt(basepar));
- break;
- default:
- f = expf(-(x * x) * basepar);
- break;
+ case 1:
+ f = expf(-(x * x) * basepar);
+ if(f < 0.4f)
+ f = 0.0f;
+ else
+ f = 1.0f;
+ break;
+ case 2:
+ f = expf(-(fabs(x)) * sqrt(basepar));
+ break;
+ default:
+ f = expf(-(x * x) * basepar);
+ break;
}
if(makezero)
f = 0.0f;
@@ -237,33 +237,39 @@ float PADnoteParameters::getprofile(float *smp, int size)
//compute the amplitude multiplier
switch(Php.amp.type) {
- case 1:
- amp = expf(-(origx * origx) * 10.0f * amppar1);
- break;
- case 2:
- amp = 0.5f * (1.0f + cosf(3.1415926f * origx * sqrt(amppar1 * 4.0f + 1.0f)));
- break;
- case 3:
- amp = 1.0f / (powf(origx * (amppar1 * 2.0f + 0.8f), 14.0f) + 1.0f);
- break;
- }
-
- //apply the amplitude multiplier
- float finalsmp = f;
- if(Php.amp.type != 0)
- switch(Php.amp.mode) {
- case 0:
- finalsmp = amp * (1.0f - amppar2) + finalsmp * amppar2;
- break;
case 1:
- finalsmp *= amp * (1.0f - amppar2) + amppar2;
+ amp = expf(-(origx * origx) * 10.0f * amppar1);
break;
case 2:
- finalsmp = finalsmp / (amp + powf(amppar2, 4.0f) * 20.0f + 0.0001f);
+ amp = 0.5f
+ * (1.0f
+ + cosf(3.1415926f * origx * sqrt(amppar1 * 4.0f + 1.0f)));
break;
case 3:
- finalsmp = amp / (finalsmp + powf(amppar2, 4.0f) * 20.0f + 0.0001f);
+ amp = 1.0f
+ / (powf(origx * (amppar1 * 2.0f + 0.8f), 14.0f) + 1.0f);
break;
+ }
+
+ //apply the amplitude multiplier
+ float finalsmp = f;
+ if(Php.amp.type != 0)
+ switch(Php.amp.mode) {
+ case 0:
+ finalsmp = amp * (1.0f - amppar2) + finalsmp * amppar2;
+ break;
+ case 1:
+ finalsmp *= amp * (1.0f - amppar2) + amppar2;
+ break;
+ case 2:
+ finalsmp = finalsmp
+ / (amp + powf(amppar2, 4.0f) * 20.0f + 0.0001f);
+ break;
+ case 3:
+ finalsmp = amp
+ / (finalsmp
+ + powf(amppar2, 4.0f) * 20.0f + 0.0001f);
+ break;
}
;
@@ -288,7 +294,7 @@ float PADnoteParameters::getprofile(float *smp, int size)
//compute the estimated perceived bandwidth
float sum = 0.0f;
- int i;
+ int i;
for(i = 0; i < size / 2 - 2; ++i) {
sum += smp[i] * smp[i] + smp[size - i - 1] * smp[size - i - 1];
if(sum >= 4.0f)
@@ -322,46 +328,48 @@ float PADnoteParameters::getNhr(int n)
float n0 = n - 1.0f;
float tmp = 0.0f;
- int thresh = 0;
+ int thresh = 0;
switch(Phrpos.type) {
- case 1:
- thresh = (int)(par2 * par2 * 100.0f) + 1;
- if(n < thresh)
- result = n;
- else
- result = 1.0f + n0 + (n0 - thresh + 1.0f) * par1 * 8.0f;
- break;
- case 2:
- thresh = (int)(par2 * par2 * 100.0f) + 1;
- if(n < thresh)
+ case 1:
+ thresh = (int)(par2 * par2 * 100.0f) + 1;
+ if(n < thresh)
+ result = n;
+ else
+ result = 1.0f + n0 + (n0 - thresh + 1.0f) * par1 * 8.0f;
+ break;
+ case 2:
+ thresh = (int)(par2 * par2 * 100.0f) + 1;
+ if(n < thresh)
+ result = n;
+ else
+ result = 1.0f + n0 - (n0 - thresh + 1.0f) * par1 * 0.90f;
+ break;
+ case 3:
+ tmp = par1 * 100.0f + 1.0f;
+ result = powf(n0 / tmp, 1.0f - par2 * 0.8f) * tmp + 1.0f;
+ break;
+ case 4:
+ result = n0
+ * (1.0f
+ - par1)
+ + powf(n0 * 0.1f, par2 * 3.0f
+ + 1.0f) * par1 * 10.0f + 1.0f;
+ break;
+ case 5:
+ result = n0
+ + sinf(n0 * par2 * par2 * PI
+ * 0.999f) * sqrt(par1) * 2.0f + 1.0f;
+ break;
+ case 6:
+ tmp = powf(par2 * 2.0f, 2.0f) + 0.1f;
+ result = n0 * powf(1.0f + par1 * powf(n0 * 0.8f, tmp), tmp) + 1.0f;
+ break;
+ default:
result = n;
- else
- result = 1.0f + n0 - (n0 - thresh + 1.0f) * par1 * 0.90f;
- break;
- case 3:
- tmp = par1 * 100.0f + 1.0f;
- result = powf(n0 / tmp, 1.0f - par2 * 0.8f) * tmp + 1.0f;
- break;
- case 4:
- result = n0
- * (1.0f
- - par1)
- + powf(n0 * 0.1f, par2 * 3.0f + 1.0f) * par1 * 10.0f + 1.0f;
- break;
- case 5:
- result = n0
- + sinf(n0 * par2 * par2 * PI * 0.999f) * sqrt(par1) * 2.0f + 1.0f;
- break;
- case 6:
- tmp = powf(par2 * 2.0f, 2.0f) + 0.1f;
- result = n0 * powf(1.0f + par1 * powf(n0 * 0.8f, tmp), tmp) + 1.0f;
- break;
- default:
- result = n;
- break;
+ break;
}
- float par3 = Phrpos.par3 / 255.0f;
+ float par3 = Phrpos.par3 / 255.0f;
float iresult = floor(result + 0.5f);
float dresult = result - iresult;
@@ -411,37 +419,37 @@ void PADnoteParameters::generatespectrum_bandwidthMode(float *spectrum,
//compute the bandwidth of each harmonic
float bandwidthcents = setPbandwidth(Pbandwidth);
- float bw =
+ float bw =
(powf(2.0f, bandwidthcents / 1200.0f) - 1.0f) * basefreq / bwadjust;
float power = 1.0f;
switch(Pbwscale) {
- case 0:
- power = 1.0f;
- break;
- case 1:
- power = 0.0f;
- break;
- case 2:
- power = 0.25f;
- break;
- case 3:
- power = 0.5f;
- break;
- case 4:
- power = 0.75f;
- break;
- case 5:
- power = 1.5f;
- break;
- case 6:
- power = 2.0f;
- break;
- case 7:
- power = -0.5f;
- break;
+ case 0:
+ power = 1.0f;
+ break;
+ case 1:
+ power = 0.0f;
+ break;
+ case 2:
+ power = 0.25f;
+ break;
+ case 3:
+ power = 0.5f;
+ break;
+ case 4:
+ power = 0.75f;
+ break;
+ case 5:
+ power = 1.5f;
+ break;
+ case 6:
+ power = 2.0f;
+ break;
+ case 7:
+ power = -0.5f;
+ break;
}
bw = bw * powf(realfreq / basefreq, power);
- int ibw = (int)((bw / (SAMPLE_RATE * 0.5f) * size)) + 1;
+ int ibw = (int)((bw / (SAMPLE_RATE * 0.5f) * size)) + 1;
float amp = harmonics[nh - 1];
if(resonance->Penabled)
@@ -449,7 +457,7 @@ void PADnoteParameters::generatespectrum_bandwidthMode(float *spectrum,
if(ibw > profilesize) { //if the bandwidth is larger than the profilesize
float rap = sqrt((float)profilesize / (float)ibw);
- int cfreq =
+ int cfreq =
(int) (realfreq / (SAMPLE_RATE * 0.5f) * size) - ibw / 2;
for(int i = 0; i < ibw; ++i) {
int src = (int)(i * rap * rap);
@@ -465,15 +473,16 @@ void PADnoteParameters::generatespectrum_bandwidthMode(float *spectrum,
float rap = sqrt((float)ibw / (float)profilesize);
float ibasefreq = realfreq / (SAMPLE_RATE * 0.5f) * size;
for(int i = 0; i < profilesize; ++i) {
- float idfreq = i / (float)profilesize - 0.5f;
+ float idfreq = i / (float)profilesize - 0.5f;
idfreq *= ibw;
- int spfreq = (int) (idfreq + ibasefreq);
+ int spfreq = (int) (idfreq + ibasefreq);
float fspfreq = fmod((double)idfreq + ibasefreq, 1.0f);
if(spfreq <= 0)
continue;
if(spfreq >= size - 1)
break;
- spectrum[spfreq] += amp * profile[i] * rap * (1.0f - fspfreq);
+ spectrum[spfreq] += amp * profile[i] * rap
+ * (1.0f - fspfreq);
spectrum[spfreq + 1] += amp * profile[i] * rap * fspfreq;
}
}
@@ -521,16 +530,16 @@ void PADnoteParameters::generatespectrum_otherModes(float *spectrum,
float amp = harmonics[nh - 1];
if(resonance->Penabled)
amp *= resonance->getfreqresponse(realfreq);
- int cfreq = (int) (realfreq / (SAMPLE_RATE * 0.5f) * size);
+ int cfreq = (int) (realfreq / (SAMPLE_RATE * 0.5f) * size);
spectrum[cfreq] = amp + 1e-9;
}
if(Pmode != 1) {
int old = 0;
- for(int k = 1; k < size; ++k) {
+ for(int k = 1; k < size; ++k)
if((spectrum[k] > 1e-10) || (k == (size - 1))) {
- int delta = k - old;
+ int delta = k - old;
float val1 = spectrum[old];
float val2 = spectrum[k];
float idelta = 1.0f / delta;
@@ -540,7 +549,6 @@ void PADnoteParameters::generatespectrum_otherModes(float *spectrum,
}
old = k;
}
- }
}
}
@@ -550,10 +558,10 @@ void PADnoteParameters::generatespectrum_otherModes(float *spectrum,
void PADnoteParameters::applyparameters(bool lockmutex)
{
const int samplesize = (((int) 1) << (Pquality.samplesize + 14));
- int spectrumsize = samplesize / 2;
- float spectrum[spectrumsize];
- int profilesize = 512;
- float profile[profilesize];
+ int spectrumsize = samplesize / 2;
+ float spectrum[spectrumsize];
+ int profilesize = 512;
+ float profile[profilesize];
float bwadjust = getprofile(profile, profilesize);
@@ -598,7 +606,7 @@ void PADnoteParameters::applyparameters(bool lockmutex)
basefreq * basefreqadjust);
const int extra_samples = 5; //the last samples contains the first samples (used for linear/cubic interpolation)
- newsample.smp = new float[samplesize + extra_samples];
+ newsample.smp = new float[samplesize + extra_samples];
newsample.smp[0] = 0.0f;
for(int i = 1; i < spectrumsize; ++i) //randomize the phases
@@ -610,7 +618,7 @@ void PADnoteParameters::applyparameters(bool lockmutex)
float rms = 0.0f;
for(int i = 0; i < samplesize; ++i)
rms += newsample.smp[i] * newsample.smp[i];
- rms = sqrt(rms);
+ rms = sqrt(rms);
if(rms < 0.000001f)
rms = 1.0f;
rms *= sqrt(262144.0f / samplesize);
@@ -664,7 +672,7 @@ void PADnoteParameters::export2wav(std::string basefilename)
char tmpstr[20];
snprintf(tmpstr, 20, "_%02d", k + 1);
std::string filename = basefilename + std::string(tmpstr) + ".wav";
- WavFile wav(filename, SAMPLE_RATE, 1);
+ WavFile wav(filename, SAMPLE_RATE, 1);
if(wav.good()) {
int nsmps = sample[k].size;
short int *smps = new short int[nsmps];
@@ -782,25 +790,25 @@ void PADnoteParameters::getfromXML(XMLwrapper *xml)
Pbwscale = xml->getpar127("bandwidth_scale", Pbwscale);
if(xml->enterbranch("HARMONIC_PROFILE")) {
- Php.base.type = xml->getpar127("base_type", Php.base.type);
- Php.base.par1 = xml->getpar127("base_par1", Php.base.par1);
- Php.freqmult = xml->getpar127("frequency_multiplier",
- Php.freqmult);
+ Php.base.type = xml->getpar127("base_type", Php.base.type);
+ Php.base.par1 = xml->getpar127("base_par1", Php.base.par1);
+ Php.freqmult = xml->getpar127("frequency_multiplier",
+ Php.freqmult);
Php.modulator.par1 = xml->getpar127("modulator_par1",
Php.modulator.par1);
Php.modulator.freq = xml->getpar127("modulator_frequency",
Php.modulator.freq);
- Php.width = xml->getpar127("width", Php.width);
- Php.amp.type = xml->getpar127("amplitude_multiplier_type",
- Php.amp.type);
- Php.amp.mode = xml->getpar127("amplitude_multiplier_mode",
- Php.amp.mode);
- Php.amp.par1 = xml->getpar127("amplitude_multiplier_par1",
- Php.amp.par1);
- Php.amp.par2 = xml->getpar127("amplitude_multiplier_par2",
- Php.amp.par2);
- Php.autoscale = xml->getparbool("autoscale", Php.autoscale);
- Php.onehalf = xml->getpar127("one_half", Php.onehalf);
+ Php.width = xml->getpar127("width", Php.width);
+ Php.amp.type = xml->getpar127("amplitude_multiplier_type",
+ Php.amp.type);
+ Php.amp.mode = xml->getpar127("amplitude_multiplier_mode",
+ Php.amp.mode);
+ Php.amp.par1 = xml->getpar127("amplitude_multiplier_par1",
+ Php.amp.par1);
+ Php.amp.par2 = xml->getpar127("amplitude_multiplier_par2",
+ Php.amp.par2);
+ Php.autoscale = xml->getparbool("autoscale", Php.autoscale);
+ Php.onehalf = xml->getpar127("one_half", Php.onehalf);
xml->exitbranch();
}
@@ -825,9 +833,9 @@ void PADnoteParameters::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("SAMPLE_QUALITY")) {
Pquality.samplesize = xml->getpar127("samplesize", Pquality.samplesize);
Pquality.basenote = xml->getpar127("basenote", Pquality.basenote);
- Pquality.oct = xml->getpar127("octaves", Pquality.oct);
- Pquality.smpoct = xml->getpar127("samples_per_octave",
- Pquality.smpoct);
+ Pquality.oct = xml->getpar127("octaves", Pquality.oct);
+ Pquality.smpoct = xml->getpar127("samples_per_octave",
+ Pquality.smpoct);
xml->exitbranch();
}
@@ -891,4 +899,3 @@ void PADnoteParameters::getfromXML(XMLwrapper *xml)
xml->exitbranch();
}
}
-
diff --git a/src/Params/PADnoteParameters.h b/src/Params/PADnoteParameters.h
@@ -100,7 +100,7 @@ class PADnoteParameters:public Presets
/* Equal temperate (this is used only if the Pfixedfreq is enabled)
If this parameter is 0, the frequency is fixed (to 440 Hz);
if this parameter is 64, 1 MIDI halftone -> 1 frequency halftone */
- unsigned char PfixedfreqET;
+ unsigned char PfixedfreqET;
unsigned short int PDetune; //fine detune
unsigned short int PCoarseDetune; //coarse detune+octave
unsigned char PDetuneType; //detune type
@@ -152,7 +152,7 @@ class PADnoteParameters:public Presets
Resonance *resonance;
struct {
- int size;
+ int size;
float basefreq;
float *smp;
} sample[PAD_MAX_SAMPLES], newsample;
@@ -170,11 +170,10 @@ class PADnoteParameters:public Presets
void deletesamples();
void deletesample(int n);
- FFTwrapper *fft;
+ FFTwrapper *fft;
pthread_mutex_t *mutex;
};
#endif
-
diff --git a/src/Params/Presets.cpp b/src/Params/Presets.cpp
@@ -26,7 +26,7 @@
Presets::Presets()
{
- type[0] = 0;
+ type[0] = 0;
}
Presets::~Presets()
@@ -85,16 +85,14 @@ void Presets::paste(int npreset)
return;
}
}
- else {
- if(!presetsstore.pastepreset(xml, npreset)) {
- delete (xml);
- return;
- }
+ else
+ if(!presetsstore.pastepreset(xml, npreset)) {
+ delete (xml);
+ return;
}
- if(xml->enterbranch(type) == 0) {
+ if(xml->enterbranch(type) == 0)
return;
- }
defaults();
getfromXML(xml);
@@ -119,4 +117,3 @@ void Presets::deletepreset(int npreset)
{
presetsstore.deletepreset(npreset);
}
-
diff --git a/src/Params/Presets.h b/src/Params/Presets.h
@@ -54,4 +54,3 @@ class Presets
};
#endif
-
diff --git a/src/Params/PresetsArray.cpp b/src/Params/PresetsArray.cpp
@@ -95,12 +95,11 @@ void PresetsArray::paste(int npreset)
return;
}
}
- else {
- if(!presetsstore.pastepreset(xml, npreset)) {
- delete (xml);
- nelement = -1;
- return;
- }
+ else
+ if(!presetsstore.pastepreset(xml, npreset)) {
+ delete (xml);
+ nelement = -1;
+ return;
}
if(xml->enterbranch(type) == 0) {
@@ -135,4 +134,3 @@ void PresetsArray::setelement(int n)
{
nelement = n;
}
-
diff --git a/src/Params/PresetsArray.h b/src/Params/PresetsArray.h
@@ -28,7 +28,7 @@
#include "Presets.h"
/**PresetsArray and Clipboard management*/
-class PresetsArray : public Presets
+class PresetsArray:public Presets
{
public:
PresetsArray();
@@ -51,10 +51,9 @@ class PresetsArray : public Presets
virtual void getfromXML(XMLwrapper *xml) = 0;
virtual void defaults() = 0;
virtual void add2XMLsection(XMLwrapper *xml, int n) = 0;
- virtual void getfromXMLsection(XMLwrapper *xml, int n) = 0;
+ virtual void getfromXMLsection(XMLwrapper *xml, int n) = 0;
virtual void defaults(int n) = 0;
int nelement;
};
#endif
-
diff --git a/src/Params/PresetsStore.cpp b/src/Params/PresetsStore.cpp
@@ -101,8 +101,8 @@ void PresetsStore::rescanforpresets(const string &type)
continue;
//open directory
- string dirname = config.cfg.presetsDirList[i];
- DIR *dir = opendir(dirname.c_str());
+ string dirname = config.cfg.presetsDirList[i];
+ DIR *dir = opendir(dirname.c_str());
if(dir == NULL)
continue;
struct dirent *fn;
@@ -168,7 +168,7 @@ bool PresetsStore::pastepreset(XMLwrapper *xml, unsigned int npreset)
string filename = presets[npreset].file;
if(filename.empty())
return false;
- bool result = (xml->loadXMLfile(filename) >= 0);
+ bool result = (xml->loadXMLfile(filename) >= 0);
return result;
}
@@ -182,4 +182,3 @@ void PresetsStore::deletepreset(unsigned int npreset)
return;
remove(filename.c_str());
}
-
diff --git a/src/Params/PresetsStore.h b/src/Params/PresetsStore.h
@@ -45,7 +45,7 @@ class PresetsStore
struct presetstruct {
presetstruct(std::string _file, std::string _name)
- :file(_file),name(_name){};
+ :file(_file), name(_name) {}
bool operator<(const presetstruct &b) const;
std::string file;
std::string name;
@@ -64,4 +64,3 @@ class PresetsStore
};
extern PresetsStore presetsstore;
-
diff --git a/src/Params/SUBnoteParameters.cpp b/src/Params/SUBnoteParameters.cpp
@@ -29,7 +29,7 @@ SUBnoteParameters::SUBnoteParameters():Presets()
setpresettype("Psubsyth");
AmpEnvelope = new EnvelopeParams(64, 1);
AmpEnvelope->ADSRinit_dB(0, 40, 127, 25);
- FreqEnvelope = new EnvelopeParams(64, 0);
+ FreqEnvelope = new EnvelopeParams(64, 0);
FreqEnvelope->ASRinit(30, 50, 64, 60);
BandWidthEnvelope = new EnvelopeParams(64, 0);
BandWidthEnvelope->ASRinit_bw(100, 70, 64, 60);
@@ -48,16 +48,16 @@ void SUBnoteParameters::defaults()
PPanning = 64;
PAmpVelocityScaleFunction = 90;
- Pfixedfreq = 0;
- PfixedfreqET = 0;
- Pnumstages = 2;
- Pbandwidth = 40;
- Phmagtype = 0;
- Pbwscale = 64;
- Pstereo = 1;
- Pstart = 1;
+ Pfixedfreq = 0;
+ PfixedfreqET = 0;
+ Pnumstages = 2;
+ Pbandwidth = 40;
+ Phmagtype = 0;
+ Pbwscale = 64;
+ Pstereo = 1;
+ Pstart = 1;
- PDetune = 8192;
+ PDetune = 8192;
PCoarseDetune = 0;
PDetuneType = 1;
PFreqEnvelopeEnabled = 0;
@@ -198,15 +198,15 @@ void SUBnoteParameters::getfromXML(XMLwrapper *xml)
}
if(xml->enterbranch("FREQUENCY_PARAMETERS")) {
- Pfixedfreq = xml->getparbool("fixed_freq", Pfixedfreq);
- PfixedfreqET = xml->getpar127("fixed_freq_et", PfixedfreqET);
+ Pfixedfreq = xml->getparbool("fixed_freq", Pfixedfreq);
+ PfixedfreqET = xml->getpar127("fixed_freq_et", PfixedfreqET);
- PDetune = xml->getpar("detune", PDetune, 0, 16383);
+ PDetune = xml->getpar("detune", PDetune, 0, 16383);
PCoarseDetune = xml->getpar("coarse_detune", PCoarseDetune, 0, 16383);
PDetuneType = xml->getpar127("detune_type", PDetuneType);
- Pbandwidth = xml->getpar127("bandwidth", Pbandwidth);
- Pbwscale = xml->getpar127("bandwidth_scale", Pbwscale);
+ Pbandwidth = xml->getpar127("bandwidth", Pbandwidth);
+ Pbwscale = xml->getpar127("bandwidth_scale", Pbwscale);
PFreqEnvelopeEnabled = xml->getparbool("freq_envelope_enabled",
PFreqEnvelopeEnabled);
@@ -248,4 +248,3 @@ void SUBnoteParameters::getfromXML(XMLwrapper *xml)
xml->exitbranch();
}
}
-
diff --git a/src/Params/SUBnoteParameters.h b/src/Params/SUBnoteParameters.h
@@ -101,4 +101,3 @@ class SUBnoteParameters:public Presets
};
#endif
-
diff --git a/src/Samples/Sample.cpp b/src/Samples/Sample.cpp
@@ -19,7 +19,7 @@
Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <cmath>
-#include <cstring>//for memcpy/memset
+#include <cstring> //for memcpy/memset
#include <iostream>
#include "Sample.h"
@@ -30,7 +30,7 @@ using namespace std;
/**\TODO start using pointer math here as these will be Frequency called
* functions throughout the code*/
Sample::Sample()
- :bufferSize(1),buffer(new float[1])
+ :bufferSize(1), buffer(new float[1])
{
buffer[0] = 0.0f;
}
@@ -105,7 +105,11 @@ bool Sample::operator==(const Sample &smp) const
* @param xb X of point b
* @return estimated Y of test point
*/
-float linearEstimate(float ya, float yb, float xt, float xa = 0.0f, float xb =1.0f)
+float linearEstimate(float ya,
+ float yb,
+ float xt,
+ float xa = 0.0f,
+ float xb = 1.0f)
{
#warning TODO this could be done with a good bit less computation
//Lets make this simple by normalizing the x axis
@@ -121,7 +125,7 @@ float linearEstimate(float ya, float yb, float xt, float xa = 0.0f, float xb =1.
//Now xa=0 xb=1 0<=xt<=1
//simpily use y=mx+b
- return (yb-ya) * xt + ya;
+ return (yb - ya) * xt + ya;
}
@@ -129,7 +133,7 @@ void Sample::resample(const unsigned int rate, const unsigned int nrate)
{
if(rate == nrate)
return; //no resampling here
- else {//resampling occurs here
+ else { //resampling occurs here
float ratio = (nrate * 1.0f) / (rate * 1.0f);
int nBufferSize = (int)bufferSize * ratio;
@@ -137,11 +141,11 @@ void Sample::resample(const unsigned int rate, const unsigned int nrate)
//addition is done to avoid 0 edge case
for(int i = 0; i < nBufferSize; ++i)
- nBuffer[i] = linearEstimate(buffer[(int)floor(i/ratio)],
- buffer[(int)ceil((i+1)/ratio)],
+ nBuffer[i] = linearEstimate(buffer[(int)floor(i / ratio)],
+ buffer[(int)ceil((i + 1) / ratio)],
i,
- floor(i/ratio),
- ceil((i+1)/ratio));
+ floor(i / ratio),
+ ceil((i + 1) / ratio));
//put the new data in
delete[] buffer;
@@ -152,8 +156,8 @@ void Sample::resample(const unsigned int rate, const unsigned int nrate)
Sample &Sample::append(const Sample &smp)
{
- int nbufferSize = bufferSize + smp.bufferSize;
- float *nbuffer = new float[nbufferSize];
+ int nbufferSize = bufferSize + smp.bufferSize;
+ float *nbuffer = new float[nbufferSize];
memcpy(nbuffer, buffer, bufferSize * sizeof(float));
memcpy(nbuffer + bufferSize, smp.buffer, smp.bufferSize * sizeof(float));
@@ -166,7 +170,7 @@ Sample &Sample::append(const Sample &smp)
Sample Sample::subSample(int a, int b) const
{
- return Sample(b-a, buffer+a);
+ return Sample(b - a, buffer + a);
}
float Sample::max() const
@@ -195,4 +199,3 @@ float Sample::absMax() const
max = fabs(buffer[i]);
return max;
}
-
diff --git a/src/Samples/Sample.h b/src/Samples/Sample.h
@@ -76,4 +76,3 @@ class Sample
float *buffer;
};
#endif
-
diff --git a/src/Synth/ADnote.cpp b/src/Synth/ADnote.cpp
@@ -38,29 +38,28 @@ ADnote::ADnote(ADnoteParameters *pars,
int portamento_,
int midinote_,
bool besilent)
-:SynthNote(freq, velocity, portamento_, midinote, besilent)
+ :SynthNote(freq, velocity, portamento_, midinote, besilent)
{
-
tmpwavel = new float [SOUND_BUFFER_SIZE];
tmpwaver = new float [SOUND_BUFFER_SIZE];
bypassl = new float [SOUND_BUFFER_SIZE];
bypassr = new float [SOUND_BUFFER_SIZE];
- partparams = pars;
+ partparams = pars;
ctl = ctl_;
- portamento = portamento_;
- midinote = midinote_;
- NoteEnabled = ON;
- basefreq = freq;
+ portamento = portamento_;
+ midinote = midinote_;
+ NoteEnabled = ON;
+ basefreq = freq;
if(velocity > 1.0f)
velocity = 1.0f;
this->velocity = velocity;
time = 0.0f;
stereo = pars->GlobalPar.PStereo;
- NoteGlobalPar.Detune = getdetune(pars->GlobalPar.PDetuneType,
- pars->GlobalPar.PCoarseDetune,
- pars->GlobalPar.PDetune);
+ NoteGlobalPar.Detune = getdetune(pars->GlobalPar.PDetuneType,
+ pars->GlobalPar.PCoarseDetune,
+ pars->GlobalPar.PDetune);
bandwidthDetuneMultiplier = pars->getBandwidthDetuneMultiplier();
if(pars->GlobalPar.PPanning == 0)
@@ -77,16 +76,16 @@ ADnote::ADnote(ADnoteParameters *pars,
PFilterVelocityScaleFunction) - 1);
if(pars->GlobalPar.PPunchStrength != 0) {
- NoteGlobalPar.Punch.Enabled = 1;
+ NoteGlobalPar.Punch.Enabled = 1;
NoteGlobalPar.Punch.t = 1.0f; //start from 1.0f and to 0.0f
NoteGlobalPar.Punch.initialvalue =
((powf(10, 1.5f * pars->GlobalPar.PPunchStrength / 127.0f) - 1.0f)
* VelF(velocity,
pars->GlobalPar.PPunchVelocitySensing));
- float time =
+ float time =
powf(10, 3.0f * pars->GlobalPar.PPunchTime / 127.0f) / 10000.0f; //0.1f .. 100 ms
float stretch = powf(440.0f / freq,
- pars->GlobalPar.PPunchStretch / 64.0f);
+ pars->GlobalPar.PPunchStretch / 64.0f);
NoteGlobalPar.Punch.dt = 1.0f / (time * SAMPLE_RATE * stretch);
}
else
@@ -98,7 +97,7 @@ ADnote::ADnote(ADnoteParameters *pars,
NoteVoicePar[nvoice].FMSmp = NULL;
NoteVoicePar[nvoice].VoiceOut = NULL;
- NoteVoicePar[nvoice].FMVoice = -1;
+ NoteVoicePar[nvoice].FMVoice = -1;
unison_size[nvoice] = 1;
if(pars->VoicePar[nvoice].Enabled == 0) {
@@ -118,64 +117,66 @@ ADnote::ADnote(ADnoteParameters *pars,
unison_base_freq_rap[nvoice] = new float[unison];
unison_freq_rap[nvoice] = new float[unison];
unison_invert_phase[nvoice] = new bool[unison];
- float unison_spread = pars->getUnisonFrequencySpreadCents(
+ float unison_spread = pars->getUnisonFrequencySpreadCents(
nvoice);
float unison_real_spread = powf(2.0f, (unison_spread * 0.5f) / 1200.0f);
float unison_vibratto_a = pars->VoicePar[nvoice].Unison_vibratto
- / 127.0f; //0.0f .. 1.0f
+ / 127.0f; //0.0f .. 1.0f
switch(unison) {
- case 1:
- unison_base_freq_rap[nvoice][0] = 1.0f; //if the unison is not used, always make the only subvoice to have the default note
- break;
- case 2: { //unison for 2 subvoices
- unison_base_freq_rap[nvoice][0] = 1.0f / unison_real_spread;
- unison_base_freq_rap[nvoice][1] = unison_real_spread;
- };
- break;
- default: { //unison for more than 2 subvoices
- float unison_values[unison];
- float min = -1e-6, max = 1e-6;
- for(int k = 0; k < unison; ++k) {
- float step = (k / (float) (unison - 1)) * 2.0f - 1.0f; //this makes the unison spread more uniform
- float val = step + (RND * 2.0f - 1.0f) / (unison - 1);
- unison_values[k] = limit(val, min, max);
- }
- float diff = max - min;
- for(int k = 0; k < unison; ++k) {
- unison_values[k] =
- (unison_values[k] - (max + min) * 0.5f) / diff; //the lowest value will be -1 and the highest will be 1
- unison_base_freq_rap[nvoice][k] =
- powf(2.0f, (unison_spread * unison_values[k]) / 1200);
- }
- };
+ case 1:
+ unison_base_freq_rap[nvoice][0] = 1.0f; //if the unison is not used, always make the only subvoice to have the default note
+ break;
+ case 2: { //unison for 2 subvoices
+ unison_base_freq_rap[nvoice][0] = 1.0f / unison_real_spread;
+ unison_base_freq_rap[nvoice][1] = unison_real_spread;
+ };
+ break;
+ default: { //unison for more than 2 subvoices
+ float unison_values[unison];
+ float min = -1e-6, max = 1e-6;
+ for(int k = 0; k < unison; ++k) {
+ float step = (k / (float) (unison - 1)) * 2.0f - 1.0f; //this makes the unison spread more uniform
+ float val = step + (RND * 2.0f - 1.0f) / (unison - 1);
+ unison_values[k] = limit(val, min, max);
+ }
+ float diff = max - min;
+ for(int k = 0; k < unison; ++k) {
+ unison_values[k] =
+ (unison_values[k] - (max + min) * 0.5f) / diff; //the lowest value will be -1 and the highest will be 1
+ unison_base_freq_rap[nvoice][k] =
+ powf(2.0f, (unison_spread * unison_values[k]) / 1200);
+ }
+ };
}
//unison vibrattos
- if(unison > 1) {
+ if(unison > 1)
for(int k = 0; k < unison; ++k) //reduce the frequency difference for larger vibrattos
unison_base_freq_rap[nvoice][k] = 1.0f
- + (unison_base_freq_rap[nvoice][k] - 1.0f)
- * (1.0f - unison_vibratto_a);
- }
+ + (unison_base_freq_rap[
+ nvoice][k] - 1.0f)
+ * (1.0f - unison_vibratto_a);
unison_vibratto[nvoice].step = new float[unison];
unison_vibratto[nvoice].position = new float[unison];
unison_vibratto[nvoice].amplitude =
(unison_real_spread - 1.0f) * unison_vibratto_a;
float increments_per_second = SAMPLE_RATE
- / (float)SOUND_BUFFER_SIZE;
- float vibratto_base_period = 0.25f
- * powf(2.0f,
- (1.0f
- - pars->VoicePar[nvoice].
- Unison_vibratto_speed / 127.0f) * 4.0f);
+ / (float)SOUND_BUFFER_SIZE;
+ float vibratto_base_period = 0.25f
+ * powf(
+ 2.0f,
+ (1.0f
+ - pars->VoicePar[nvoice].
+ Unison_vibratto_speed
+ / 127.0f) * 4.0f);
for(int k = 0; k < unison; ++k) {
unison_vibratto[nvoice].position[k] = RND * 1.8f - 0.9f;
//make period to vary randomly from 50% to 200% vibratto base period
float vibratto_period = vibratto_base_period
- * powf(2.0f, RND * 2.0f - 1.0f);
+ * powf(2.0f, RND * 2.0f - 1.0f);
float m = 4.0f / (vibratto_period * increments_per_second);
if(RND < 0.5f)
@@ -194,16 +195,16 @@ ADnote::ADnote(ADnoteParameters *pars,
if(unison != 1) {
int inv = pars->VoicePar[nvoice].Unison_invert_phase;
switch(inv) {
- case 0: for(int k = 0; k < unison; ++k)
- unison_invert_phase[nvoice][k] = false;
- break;
- case 1: for(int k = 0; k < unison; ++k)
- unison_invert_phase[nvoice][k] = (RND > 0.5f);
- break;
- default: for(int k = 0; k < unison; ++k)
- unison_invert_phase[nvoice][k] =
- (k % inv == 0) ? true : false;
- break;
+ case 0: for(int k = 0; k < unison; ++k)
+ unison_invert_phase[nvoice][k] = false;
+ break;
+ case 1: for(int k = 0; k < unison; ++k)
+ unison_invert_phase[nvoice][k] = (RND > 0.5f);
+ break;
+ default: for(int k = 0; k < unison; ++k)
+ unison_invert_phase[nvoice][k] =
+ (k % inv == 0) ? true : false;
+ break;
}
}
@@ -223,7 +224,7 @@ ADnote::ADnote(ADnoteParameters *pars,
//use the Globalpars.detunetype if the detunetype is 0
if(pars->VoicePar[nvoice].PDetuneType != 0) {
- NoteVoicePar[nvoice].Detune = getdetune(
+ NoteVoicePar[nvoice].Detune = getdetune(
pars->VoicePar[nvoice].PDetuneType,
pars->VoicePar[nvoice].
PCoarseDetune,
@@ -234,7 +235,7 @@ ADnote::ADnote(ADnoteParameters *pars,
pars->VoicePar[nvoice].PDetune); //fine detune
}
else {
- NoteVoicePar[nvoice].Detune = getdetune(
+ NoteVoicePar[nvoice].Detune = getdetune(
pars->GlobalPar.PDetuneType,
pars->VoicePar[nvoice].
PCoarseDetune,
@@ -297,40 +298,40 @@ ADnote::ADnote(ADnoteParameters *pars,
oscposhi_start = (int)(RND * (OSCIL_SIZE - 1)); //put random starting point for other subvoices
}
- NoteVoicePar[nvoice].FreqLfo = NULL;
- NoteVoicePar[nvoice].FreqEnvelope = NULL;
+ NoteVoicePar[nvoice].FreqLfo = NULL;
+ NoteVoicePar[nvoice].FreqEnvelope = NULL;
- NoteVoicePar[nvoice].AmpLfo = NULL;
- NoteVoicePar[nvoice].AmpEnvelope = NULL;
+ NoteVoicePar[nvoice].AmpLfo = NULL;
+ NoteVoicePar[nvoice].AmpEnvelope = NULL;
- NoteVoicePar[nvoice].VoiceFilterL = NULL;
- NoteVoicePar[nvoice].VoiceFilterR = NULL;
- NoteVoicePar[nvoice].FilterEnvelope = NULL;
- NoteVoicePar[nvoice].FilterLfo = NULL;
+ NoteVoicePar[nvoice].VoiceFilterL = NULL;
+ NoteVoicePar[nvoice].VoiceFilterR = NULL;
+ NoteVoicePar[nvoice].FilterEnvelope = NULL;
+ NoteVoicePar[nvoice].FilterLfo = NULL;
NoteVoicePar[nvoice].FilterCenterPitch =
pars->VoicePar[nvoice].VoiceFilter->getfreq();
- NoteVoicePar[nvoice].filterbypass =
+ NoteVoicePar[nvoice].filterbypass =
pars->VoicePar[nvoice].Pfilterbypass;
switch(pars->VoicePar[nvoice].PFMEnabled) {
- case 1:
- NoteVoicePar[nvoice].FMEnabled = MORPH;
- break;
- case 2:
- NoteVoicePar[nvoice].FMEnabled = RING_MOD;
- break;
- case 3:
- NoteVoicePar[nvoice].FMEnabled = PHASE_MOD;
- break;
- case 4:
- NoteVoicePar[nvoice].FMEnabled = FREQ_MOD;
- break;
- case 5:
- NoteVoicePar[nvoice].FMEnabled = PITCH_MOD;
- break;
- default:
- NoteVoicePar[nvoice].FMEnabled = NONE;
+ case 1:
+ NoteVoicePar[nvoice].FMEnabled = MORPH;
+ break;
+ case 2:
+ NoteVoicePar[nvoice].FMEnabled = RING_MOD;
+ break;
+ case 3:
+ NoteVoicePar[nvoice].FMEnabled = PHASE_MOD;
+ break;
+ case 4:
+ NoteVoicePar[nvoice].FMEnabled = FREQ_MOD;
+ break;
+ case 5:
+ NoteVoicePar[nvoice].FMEnabled = PITCH_MOD;
+ break;
+ default:
+ NoteVoicePar[nvoice].FMEnabled = NONE;
}
NoteVoicePar[nvoice].FMVoice = pars->VoicePar[nvoice].PFMVoice;
@@ -339,30 +340,32 @@ ADnote::ADnote(ADnoteParameters *pars,
//Compute the Voice's modulator volume (incl. damping)
float fmvoldamp = powf(440.0f / getvoicebasefreq(
- nvoice),
- pars->VoicePar[nvoice].PFMVolumeDamp / 64.0f
- - 1.0f);
+ nvoice),
+ pars->VoicePar[nvoice].PFMVolumeDamp / 64.0f
+ - 1.0f);
switch(NoteVoicePar[nvoice].FMEnabled) {
- case PHASE_MOD:
- fmvoldamp =
- powf(440.0f / getvoicebasefreq(
- nvoice), pars->VoicePar[nvoice].PFMVolumeDamp / 64.0f);
- NoteVoicePar[nvoice].FMVolume =
- (expf(pars->VoicePar[nvoice].PFMVolume / 127.0f
- * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;
- break;
- case FREQ_MOD:
- NoteVoicePar[nvoice].FMVolume =
- (expf(pars->VoicePar[nvoice].PFMVolume / 127.0f
- * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;
- break;
- // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0f*8.0f)*fmvoldamp;//???????????
- // break;
- default:
- if(fmvoldamp > 1.0f)
- fmvoldamp = 1.0f;
- NoteVoicePar[nvoice].FMVolume = pars->VoicePar[nvoice].PFMVolume
- / 127.0f * fmvoldamp;
+ case PHASE_MOD:
+ fmvoldamp =
+ powf(440.0f / getvoicebasefreq(
+ nvoice), pars->VoicePar[nvoice].PFMVolumeDamp
+ / 64.0f);
+ NoteVoicePar[nvoice].FMVolume =
+ (expf(pars->VoicePar[nvoice].PFMVolume / 127.0f
+ * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;
+ break;
+ case FREQ_MOD:
+ NoteVoicePar[nvoice].FMVolume =
+ (expf(pars->VoicePar[nvoice].PFMVolume / 127.0f
+ * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;
+ break;
+ // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0f*8.0f)*fmvoldamp;//???????????
+ // break;
+ default:
+ if(fmvoldamp > 1.0f)
+ fmvoldamp = 1.0f;
+ NoteVoicePar[nvoice].FMVolume =
+ pars->VoicePar[nvoice].PFMVolume
+ / 127.0f * fmvoldamp;
}
//Voice's modulator velocity sensing
@@ -377,7 +380,7 @@ ADnote::ADnote(ADnoteParameters *pars,
firsttick[nvoice] = 1;
NoteVoicePar[nvoice].DelayTicks =
(int)((expf(pars->VoicePar[nvoice].PDelay / 127.0f
- * logf(50.0f))
+ * logf(50.0f))
- 1.0f) / SOUND_BUFFER_SIZE / 10.0f * SAMPLE_RATE);
}
@@ -417,9 +420,9 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
velocity = 1.0f;
this->velocity = velocity;
- NoteGlobalPar.Detune = getdetune(pars->GlobalPar.PDetuneType,
- pars->GlobalPar.PCoarseDetune,
- pars->GlobalPar.PDetune);
+ NoteGlobalPar.Detune = getdetune(pars->GlobalPar.PDetuneType,
+ pars->GlobalPar.PCoarseDetune,
+ pars->GlobalPar.PDetune);
bandwidthDetuneMultiplier = pars->getBandwidthDetuneMultiplier();
if(pars->GlobalPar.PPanning == 0)
@@ -429,8 +432,11 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
//center freq
NoteGlobalPar.FilterCenterPitch = pars->GlobalPar.GlobalFilter->getfreq()
- + pars->GlobalPar.PFilterVelocityScale / 127.0f * 6.0f //velocity sensing
- * (VelF(velocity, pars->GlobalPar.PFilterVelocityScaleFunction) - 1);
+ + pars->GlobalPar.PFilterVelocityScale
+ / 127.0f * 6.0f //velocity sensing
+ * (VelF(velocity,
+ pars->GlobalPar.
+ PFilterVelocityScaleFunction) - 1);
for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
@@ -442,24 +448,24 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
//use the Globalpars.detunetype if the detunetype is 0
if(pars->VoicePar[nvoice].PDetuneType != 0) {
- NoteVoicePar[nvoice].Detune = getdetune(
+ NoteVoicePar[nvoice].Detune = getdetune(
pars->VoicePar[nvoice].PDetuneType,
pars->VoicePar[nvoice].PCoarseDetune,
- 8192);//coarse detune
+ 8192); //coarse detune
NoteVoicePar[nvoice].FineDetune = getdetune(
pars->VoicePar[nvoice].PDetuneType,
0,
- pars->VoicePar[nvoice].PDetune);//fine detune
+ pars->VoicePar[nvoice].PDetune); //fine detune
}
else {
- NoteVoicePar[nvoice].Detune = getdetune(
+ NoteVoicePar[nvoice].Detune = getdetune(
pars->GlobalPar.PDetuneType,
pars->VoicePar[nvoice].PCoarseDetune,
- 8192);//coarse detune
+ 8192); //coarse detune
NoteVoicePar[nvoice].FineDetune = getdetune(
pars->GlobalPar.PDetuneType,
0,
- pars->VoicePar[nvoice].PDetune);//fine detune
+ pars->VoicePar[nvoice].PDetune); //fine detune
}
if(pars->VoicePar[nvoice].PFMDetuneType != 0)
NoteVoicePar[nvoice].FMDetune = getdetune(
@@ -482,7 +488,7 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
pars->VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,
getvoicebasefreq(nvoice),
- pars->VoicePar[nvoice].Presonance);//(gf)Modif of the above line.
+ pars->VoicePar[nvoice].Presonance); //(gf)Modif of the above line.
//I store the first elments to the last position for speedups
for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)
@@ -493,7 +499,7 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
NoteVoicePar[nvoice].FilterCenterPitch =
pars->VoicePar[nvoice].VoiceFilter->getfreq();
- NoteVoicePar[nvoice].filterbypass =
+ NoteVoicePar[nvoice].filterbypass =
pars->VoicePar[nvoice].Pfilterbypass;
@@ -501,30 +507,32 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
//Compute the Voice's modulator volume (incl. damping)
float fmvoldamp = powf(440.0f / getvoicebasefreq(nvoice),
- pars->VoicePar[nvoice].PFMVolumeDamp / 64.0f
- - 1.0f);
+ pars->VoicePar[nvoice].PFMVolumeDamp / 64.0f
+ - 1.0f);
switch(NoteVoicePar[nvoice].FMEnabled) {
- case PHASE_MOD:
- fmvoldamp =
- powf(440.0f / getvoicebasefreq(
- nvoice), pars->VoicePar[nvoice].PFMVolumeDamp / 64.0f);
- NoteVoicePar[nvoice].FMVolume =
- (expf(pars->VoicePar[nvoice].PFMVolume / 127.0f
- * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;
- break;
- case FREQ_MOD:
- NoteVoicePar[nvoice].FMVolume =
- (expf(pars->VoicePar[nvoice].PFMVolume / 127.0f
- * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;
- break;
- // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0f*8.0f)*fmvoldamp;//???????????
- // break;
- default:
- if(fmvoldamp > 1.0f)
- fmvoldamp = 1.0f;
- NoteVoicePar[nvoice].FMVolume = pars->VoicePar[nvoice].PFMVolume
- / 127.0f * fmvoldamp;
+ case PHASE_MOD:
+ fmvoldamp =
+ powf(440.0f / getvoicebasefreq(
+ nvoice), pars->VoicePar[nvoice].PFMVolumeDamp
+ / 64.0f);
+ NoteVoicePar[nvoice].FMVolume =
+ (expf(pars->VoicePar[nvoice].PFMVolume / 127.0f
+ * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;
+ break;
+ case FREQ_MOD:
+ NoteVoicePar[nvoice].FMVolume =
+ (expf(pars->VoicePar[nvoice].PFMVolume / 127.0f
+ * FM_AMP_MULTIPLIER) - 1.0f) * fmvoldamp * 4.0f;
+ break;
+ // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0f*8.0f)*fmvoldamp;//???????????
+ // break;
+ default:
+ if(fmvoldamp > 1.0f)
+ fmvoldamp = 1.0f;
+ NoteVoicePar[nvoice].FMVolume =
+ pars->VoicePar[nvoice].PFMVolume
+ / 127.0f * fmvoldamp;
}
//Voice's modulator velocity sensing
@@ -534,7 +542,7 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
NoteVoicePar[nvoice].DelayTicks =
(int)((expf(pars->VoicePar[nvoice].PDelay / 127.0f
- * logf(50.0f))
+ * logf(50.0f))
- 1.0f) / SOUND_BUFFER_SIZE / 10.0f * SAMPLE_RATE);
}
@@ -547,7 +555,8 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
NoteGlobalPar.Volume = 4.0f
* powf(0.1f, 3.0f
- * (1.0f - partparams->GlobalPar.PVolume / 96.0f)) //-60 dB .. 0 dB
+ * (1.0f - partparams->GlobalPar.PVolume
+ / 96.0f)) //-60 dB .. 0 dB
* VelF(
velocity,
partparams->GlobalPar.
@@ -573,10 +582,11 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
NoteVoicePar[nvoice].noisetype = partparams->VoicePar[nvoice].Type;
/* Voice Amplitude Parameters Init */
- NoteVoicePar[nvoice].Volume =
- powf(0.1f, 3.0f * (1.0f - partparams->VoicePar[nvoice].PVolume / 127.0f)) // -60 dB .. 0 dB
+ NoteVoicePar[nvoice].Volume =
+ powf(0.1f, 3.0f
+ * (1.0f - partparams->VoicePar[nvoice].PVolume / 127.0f)) // -60 dB .. 0 dB
* VelF(velocity,
- partparams->VoicePar[nvoice].PAmpVelocityScaleFunction);//velocity
+ partparams->VoicePar[nvoice].PAmpVelocityScaleFunction); //velocity
if(partparams->VoicePar[nvoice].PVolumeminus != 0)
NoteVoicePar[nvoice].Volume = -NoteVoicePar[nvoice].Volume;
@@ -617,7 +627,7 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
partparams->VoicePar[vc].FMSmp->newrandseed(prng());
for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)
- NoteVoicePar[nvoice].FMSmp[OSCIL_SIZE + i] =
+ NoteVoicePar[nvoice].FMSmp[OSCIL_SIZE + i] =
NoteVoicePar[nvoice].FMSmp[i];
}
@@ -628,7 +638,6 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
&& (NoteVoicePar[nvoice].FMAmpEnvelope != NULL))
FMnewamplitude[nvoice] *=
NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();
-
}
for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
@@ -637,7 +646,6 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
for(unsigned i = nvoice + 1; i < NUM_VOICES; ++i)
if((NoteVoicePar[i].FMVoice == nvoice) && (tmp[i] == 0))
tmp[i] = 1;
-
}
}
@@ -730,8 +738,8 @@ void ADnote::initparameters()
vce.noisetype = param.Type;
/* Voice Amplitude Parameters Init */
- vce.Volume = powf(0.1f, 3.0f * (1.0f - param.PVolume / 127.0f))// -60dB..0dB
- * VelF(velocity, param.PAmpVelocityScaleFunction);
+ vce.Volume = powf(0.1f, 3.0f * (1.0f - param.PVolume / 127.0f)) // -60dB..0dB
+ * VelF(velocity, param.PAmpVelocityScaleFunction);
if(param.PVolumeminus)
vce.Volume = -vce.Volume;
@@ -797,8 +805,9 @@ void ADnote::initparameters()
for(int k = 0; k < unison_size[nvoice]; ++k)
oscposhiFM[nvoice][k] = (oscposhi[nvoice][k]
- + partparams->VoicePar[vc].FMSmp->get(vce.FMSmp, tmp))
- % OSCIL_SIZE;
+ + partparams->VoicePar[vc].FMSmp->get(
+ vce.FMSmp, tmp))
+ % OSCIL_SIZE;
for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)
vce.FMSmp[OSCIL_SIZE + i] = vce.FMSmp[i];
@@ -817,7 +826,8 @@ void ADnote::initparameters()
FMnewamplitude[nvoice] = vce.FMVolume * ctl->fmamp.relamp;
if(param.PFMAmpEnvelopeEnabled != 0) {
- vce.FMAmpEnvelope = new Envelope(param.FMAmpEnvelope, basefreq);
+ vce.FMAmpEnvelope = new Envelope(param.FMAmpEnvelope,
+ basefreq);
FMnewamplitude[nvoice] *= vce.FMAmpEnvelope->envout_dB();
}
}
@@ -832,7 +842,8 @@ void ADnote::initparameters()
}
if(NoteVoicePar[nvoice].VoiceOut)
- memset(NoteVoicePar[nvoice].VoiceOut, 0, SOUND_BUFFER_SIZE * sizeof(float));
+ memset(NoteVoicePar[nvoice].VoiceOut, 0, SOUND_BUFFER_SIZE
+ * sizeof(float));
}
}
@@ -910,19 +921,20 @@ void ADnote::setfreqFM(int nvoice, float in_freq)
float ADnote::getvoicebasefreq(int nvoice) const
{
float detune = NoteVoicePar[nvoice].Detune / 100.0f
- + NoteVoicePar[nvoice].FineDetune / 100.0f
- * ctl->bandwidth.relbw * bandwidthDetuneMultiplier
- + NoteGlobalPar.Detune / 100.0f;
+ + NoteVoicePar[nvoice].FineDetune / 100.0f
+ * ctl->bandwidth.relbw * bandwidthDetuneMultiplier
+ + NoteGlobalPar.Detune / 100.0f;
if(NoteVoicePar[nvoice].fixedfreq == 0)
- return this->basefreq * powf(2, detune / 12.0f);
+ return this->basefreq *powf(2, detune / 12.0f);
else { //the fixed freq is enabled
float fixedfreq = 440.0f;
- int fixedfreqET = NoteVoicePar[nvoice].fixedfreqET;
+ int fixedfreqET = NoteVoicePar[nvoice].fixedfreqET;
if(fixedfreqET != 0) { //if the frequency varies according the keyboard note
float tmp =
(midinote
- - 69.0f) / 12.0f * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);
+ - 69.0f) / 12.0f
+ * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);
if(fixedfreqET <= 64)
fixedfreq *= powf(2.0f, tmp);
else
@@ -946,12 +958,12 @@ float ADnote::getFMvoicebasefreq(int nvoice) const
*/
void ADnote::computecurrentparameters()
{
- int nvoice;
+ int nvoice;
float voicefreq, voicepitch, filterpitch, filterfreq, FMfreq,
- FMrelativepitch, globalpitch, globalfilterpitch;
+ FMrelativepitch, globalpitch, globalfilterpitch;
globalpitch = 0.01f * (NoteGlobalPar.FreqEnvelope->envout()
- + NoteGlobalPar.FreqLfo->lfoout()
- * ctl->modwheel.relmod);
+ + NoteGlobalPar.FreqLfo->lfoout()
+ * ctl->modwheel.relmod);
globaloldamplitude = globalnewamplitude;
globalnewamplitude = NoteGlobalPar.Volume
* NoteGlobalPar.AmpEnvelope->envout_dB()
@@ -962,7 +974,7 @@ void ADnote::computecurrentparameters()
+ NoteGlobalPar.FilterCenterPitch;
float tmpfilterfreq = globalfilterpitch + ctl->filtercutoff.relfreq
- + NoteGlobalPar.FilterFreqTracking;
+ + NoteGlobalPar.FilterFreqTracking;
tmpfilterfreq = Filter::getrealfreq(tmpfilterfreq);
@@ -977,7 +989,6 @@ void ADnote::computecurrentparameters()
portamentofreqrap = ctl->portamento.freqrap;
if(ctl->portamento.used == 0) //the portamento has finished
portamento = 0; //this note is no longer "portamented"
-
}
//compute parameters for all voices
@@ -1034,8 +1045,8 @@ void ADnote::computecurrentparameters()
if(NoteVoicePar[nvoice].FreqEnvelope != NULL)
voicepitch += NoteVoicePar[nvoice].FreqEnvelope->envout()
/ 100.0f;
- voicefreq = getvoicebasefreq(nvoice)
- * powf(2, (voicepitch + globalpitch) / 12.0f); //Hz frequency
+ voicefreq = getvoicebasefreq(nvoice)
+ * powf(2, (voicepitch + globalpitch) / 12.0f); //Hz frequency
voicefreq *= ctl->pitchwheel.relfreq; //change the frequency by the controller
setfreq(nvoice, voicefreq * portamentofreqrap);
@@ -1049,7 +1060,7 @@ void ADnote::computecurrentparameters()
NoteVoicePar[nvoice].FMFreqEnvelope->envout() / 100;
FMfreq =
powf(2.0f, FMrelativepitch
- / 12.0f) * voicefreq * portamentofreqrap;
+ / 12.0f) * voicefreq * portamentofreqrap;
setfreqFM(nvoice, FMfreq);
FMoldamplitude[nvoice] = FMnewamplitude[nvoice];
@@ -1094,13 +1105,13 @@ inline void ADnote::fadein(float *smps) const
*/
inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice)
{
- int i, poshi;
+ int i, poshi;
float poslo;
for(int k = 0; k < unison_size[nvoice]; ++k) {
poshi = oscposhi[nvoice][k];
poslo = oscposlo[nvoice][k];
- int freqhi = oscfreqhi[nvoice][k];
+ int freqhi = oscfreqhi[nvoice][k];
float freqlo = oscfreqlo[nvoice][k];
float *smps = NoteVoicePar[nvoice].OscilSmp;
float *tw = tmpwave_unison[k];
@@ -1161,7 +1172,7 @@ inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){
*/
inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)
{
- int i;
+ int i;
float amp;
ComputeVoiceOscillator_LinearInterpolation(nvoice);
if(FMnewamplitude[nvoice] > 1.0f)
@@ -1175,28 +1186,29 @@ inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)
for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
- FMnewamplitude[nvoice],
- i,
- SOUND_BUFFER_SIZE);
+ amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
+ FMnewamplitude[nvoice],
+ i,
+ SOUND_BUFFER_SIZE);
tw[i] = tw[i]
- * (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];
+ * (1.0f
+ - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];
}
}
}
- else {
+ else
for(int k = 0; k < unison_size[nvoice]; ++k) {
- int poshiFM = oscposhiFM[nvoice][k];
+ int poshiFM = oscposhiFM[nvoice][k];
float posloFM = oscposloFM[nvoice][k];
- int freqhiFM = oscfreqhiFM[nvoice][k];
+ int freqhiFM = oscfreqhiFM[nvoice][k];
float freqloFM = oscfreqloFM[nvoice][k];
float *tw = tmpwave_unison[k];
for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
- FMnewamplitude[nvoice],
- i,
- SOUND_BUFFER_SIZE);
+ amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
+ FMnewamplitude[nvoice],
+ i,
+ SOUND_BUFFER_SIZE);
tw[i] = tw[i] * (1.0f - amp) + amp
* (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1 - posloFM)
+ NoteVoicePar[nvoice].FMSmp[poshiFM + 1] * posloFM);
@@ -1211,7 +1223,6 @@ inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)
oscposhiFM[nvoice][k] = poshiFM;
oscposloFM[nvoice][k] = posloFM;
}
- }
}
/*
@@ -1219,14 +1230,14 @@ inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)
*/
inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)
{
- int i;
+ int i;
float amp;
ComputeVoiceOscillator_LinearInterpolation(nvoice);
if(FMnewamplitude[nvoice] > 1.0f)
FMnewamplitude[nvoice] = 1.0f;
if(FMoldamplitude[nvoice] > 1.0f)
FMoldamplitude[nvoice] = 1.0f;
- if(NoteVoicePar[nvoice].FMVoice >= 0) {
+ if(NoteVoicePar[nvoice].FMVoice >= 0)
// if I use VoiceOut[] as modullator
for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
@@ -1239,20 +1250,19 @@ inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)
tw[i] *= (1.0f - amp) + amp * NoteVoicePar[FMVoice].VoiceOut[i];
}
}
- }
- else {
+ else
for(int k = 0; k < unison_size[nvoice]; ++k) {
- int poshiFM = oscposhiFM[nvoice][k];
+ int poshiFM = oscposhiFM[nvoice][k];
float posloFM = oscposloFM[nvoice][k];
- int freqhiFM = oscfreqhiFM[nvoice][k];
+ int freqhiFM = oscfreqhiFM[nvoice][k];
float freqloFM = oscfreqloFM[nvoice][k];
float *tw = tmpwave_unison[k];
for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
- FMnewamplitude[nvoice],
- i,
- SOUND_BUFFER_SIZE);
+ amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
+ FMnewamplitude[nvoice],
+ i,
+ SOUND_BUFFER_SIZE);
tw[i] *= (NoteVoicePar[nvoice].FMSmp[poshiFM] * (1.0f - posloFM)
+ NoteVoicePar[nvoice].FMSmp[poshiFM
+ 1] * posloFM) * amp
@@ -1268,7 +1278,6 @@ inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)
oscposhiFM[nvoice][k] = poshiFM;
oscposloFM[nvoice][k] = posloFM;
}
- }
}
@@ -1279,24 +1288,23 @@ inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)
inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
int FMmode)
{
- int carposhi = 0;
- int i, FMmodfreqhi = 0;
+ int carposhi = 0;
+ int i, FMmodfreqhi = 0;
float FMmodfreqlo = 0, carposlo = 0;
- if(NoteVoicePar[nvoice].FMVoice >= 0) {
+ if(NoteVoicePar[nvoice].FMVoice >= 0)
//if I use VoiceOut[] as modulator
for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
memcpy(tw, NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut,
- SOUND_BUFFER_SIZE * sizeof(float));
+ SOUND_BUFFER_SIZE * sizeof(float));
}
- }
- else {
+ else
//Compute the modulator and store it in tmpwave_unison[][]
for(int k = 0; k < unison_size[nvoice]; ++k) {
- int poshiFM = oscposhiFM[nvoice][k];
+ int poshiFM = oscposhiFM[nvoice][k];
float posloFM = oscposloFM[nvoice][k];
- int freqhiFM = oscfreqhiFM[nvoice][k];
+ int freqhiFM = oscfreqhiFM[nvoice][k];
float freqloFM = oscfreqloFM[nvoice][k];
float *tw = tmpwave_unison[k];
@@ -1315,10 +1323,9 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
oscposhiFM[nvoice][k] = poshiFM;
oscposloFM[nvoice][k] = posloFM;
}
- }
// Amplitude interpolation
if(ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],
- FMnewamplitude[nvoice])) {
+ FMnewamplitude[nvoice]))
for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
for(i = 0; i < SOUND_BUFFER_SIZE; ++i)
@@ -1326,22 +1333,19 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
FMnewamplitude[nvoice],
i,
SOUND_BUFFER_SIZE);
-
}
- }
- else {
+ else
for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
for(i = 0; i < SOUND_BUFFER_SIZE; ++i)
tw[i] *= FMnewamplitude[nvoice];
}
- }
//normalize: makes all sample-rates, oscil_sizes to produce same sound
if(FMmode != 0) { //Frequency modulation
float normalize = OSCIL_SIZE / 262144.0f * 44100.0f
- / (float)SAMPLE_RATE;
+ / (float)SAMPLE_RATE;
for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
float fmold = FMoldsmp[nvoice][k];
@@ -1364,10 +1368,10 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
//do the modulation
for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
- int poshi = oscposhi[nvoice][k];
- float poslo = oscposlo[nvoice][k];
- int freqhi = oscfreqhi[nvoice][k];
- float freqlo = oscfreqlo[nvoice][k];
+ int poshi = oscposhi[nvoice][k];
+ float poslo = oscposlo[nvoice][k];
+ int freqhi = oscfreqhi[nvoice][k];
+ float freqlo = oscfreqlo[nvoice][k];
for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
F2I(tw[i], FMmodfreqhi);
@@ -1385,10 +1389,10 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
}
carposhi &= (OSCIL_SIZE - 1);
- tw[i] = NoteVoicePar[nvoice].OscilSmp[carposhi]
- * (1.0f - carposlo)
- + NoteVoicePar[nvoice].OscilSmp[carposhi
- + 1] * carposlo;
+ tw[i] = NoteVoicePar[nvoice].OscilSmp[carposhi]
+ * (1.0f - carposlo)
+ + NoteVoicePar[nvoice].OscilSmp[carposhi
+ + 1] * carposlo;
poslo += freqlo;
if(poslo >= 1.0f) {
@@ -1447,22 +1451,22 @@ int ADnote::noteout(float *outl, float *outr)
continue;
if(NoteVoicePar[nvoice].noisetype == 0) //voice mode=sound
switch(NoteVoicePar[nvoice].FMEnabled) {
- case MORPH:
- ComputeVoiceOscillatorMorph(nvoice);
- break;
- case RING_MOD:
- ComputeVoiceOscillatorRingModulation(nvoice);
- break;
- case PHASE_MOD:
- ComputeVoiceOscillatorFrequencyModulation(nvoice, 0);
- break;
- case FREQ_MOD:
- ComputeVoiceOscillatorFrequencyModulation(nvoice, 1);
- break;
- //case PITCH_MOD:ComputeVoiceOscillatorPitchModulation(nvoice);break;
- default:
- ComputeVoiceOscillator_LinearInterpolation(nvoice);
- //if (config.cfg.Interpolation) ComputeVoiceOscillator_CubicInterpolation(nvoice);
+ case MORPH:
+ ComputeVoiceOscillatorMorph(nvoice);
+ break;
+ case RING_MOD:
+ ComputeVoiceOscillatorRingModulation(nvoice);
+ break;
+ case PHASE_MOD:
+ ComputeVoiceOscillatorFrequencyModulation(nvoice, 0);
+ break;
+ case FREQ_MOD:
+ ComputeVoiceOscillatorFrequencyModulation(nvoice, 1);
+ break;
+ //case PITCH_MOD:ComputeVoiceOscillatorPitchModulation(nvoice);break;
+ default:
+ ComputeVoiceOscillator_LinearInterpolation(nvoice);
+ //if (config.cfg.Interpolation) ComputeVoiceOscillator_CubicInterpolation(nvoice);
}
else
ComputeVoiceNoise(nvoice);
@@ -1476,11 +1480,11 @@ int ADnote::noteout(float *outl, float *outr)
for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
if(stereo) {
- float stereo_pos = 0;
+ float stereo_pos = 0;
if(unison_size[nvoice] > 1)
stereo_pos = k
/ (float)(unison_size[nvoice]
- - 1) * 2.0f - 1.0f;
+ - 1) * 2.0f - 1.0f;
float stereo_spread = unison_stereo_spread[nvoice] * 2.0f; //between 0 and 2.0f
if(stereo_spread > 1.0f) {
float stereo_pos_1 = (stereo_pos >= 0.0f) ? 1.0f : -1.0f;
@@ -1518,7 +1522,6 @@ int ADnote::noteout(float *outl, float *outr)
else
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwavel[i] += tw[i];
-
}
@@ -1572,7 +1575,7 @@ int ADnote::noteout(float *outl, float *outr)
NoteVoicePar[nvoice].VoiceFilterR->filterout(&tmpwaver[0]);
//check if the amplitude envelope is finished, if yes, the voice will be fadeout
- if(NoteVoicePar[nvoice].AmpEnvelope != NULL) {
+ if(NoteVoicePar[nvoice].AmpEnvelope != NULL)
if(NoteVoicePar[nvoice].AmpEnvelope->finished() != 0) {
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwavel[i] *= 1.0f - (float)i
@@ -1582,8 +1585,7 @@ int ADnote::noteout(float *outl, float *outr)
tmpwaver[i] *= 1.0f - (float)i
/ (float)SOUND_BUFFER_SIZE;
}
- //the voice is killed later
- }
+ //the voice is killed later
// Put the ADnote samples in VoiceOut (without appling Global volume, because I wish to use this voice as a modullator)
@@ -1595,44 +1597,39 @@ int ADnote::noteout(float *outl, float *outr)
else //mono
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i];
-
}
// Add the voice that do not bypass the filter to out
if(NoteVoicePar[nvoice].filterbypass == 0) { //no bypass
- if(stereo) {
+ if(stereo)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //stereo
outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume
* NoteVoicePar[nvoice].Panning * 2.0f;
outr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume
* (1.0f - NoteVoicePar[nvoice].Panning) * 2.0f;
}
- }
else
- for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)//mono
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) //mono
outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;
-
}
else { //bypass the filter
- if(stereo) {
+ if(stereo)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //stereo
bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume
* NoteVoicePar[nvoice].Panning * 2.0f;
bypassr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume
- * (1.0f - NoteVoicePar[nvoice].Panning) * 2.0f;
+ * (1.0f
+ - NoteVoicePar[nvoice].Panning) * 2.0f;
}
- }
else
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) //mono
bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;
-
}
// chech if there is necesary to proces the voice longer (if the Amplitude envelope isn't finished)
if(NoteVoicePar[nvoice].AmpEnvelope != NULL)
if(NoteVoicePar[nvoice].AmpEnvelope->finished() != 0)
KillVoice(nvoice);
-
}
@@ -1651,29 +1648,27 @@ int ADnote::noteout(float *outl, float *outr)
outr[i] += bypassr[i];
}
- if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude)) {
+ if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude))
// Amplitude Interpolation
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,
- globalnewamplitude,
- i,
- SOUND_BUFFER_SIZE);
+ globalnewamplitude,
+ i,
+ SOUND_BUFFER_SIZE);
outl[i] *= tmpvol * NoteGlobalPar.Panning;
outr[i] *= tmpvol * (1.0f - NoteGlobalPar.Panning);
}
- }
- else {
+ else
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;
outr[i] *= globalnewamplitude * (1.0f - NoteGlobalPar.Panning);
}
- }
//Apply the punch
- if(NoteGlobalPar.Punch.Enabled != 0) {
+ if(NoteGlobalPar.Punch.Enabled != 0)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float punchamp = NoteGlobalPar.Punch.initialvalue
- * NoteGlobalPar.Punch.t + 1.0f;
+ * NoteGlobalPar.Punch.t + 1.0f;
outl[i] *= punchamp;
outr[i] *= punchamp;
NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;
@@ -1682,11 +1677,10 @@ int ADnote::noteout(float *outl, float *outr)
break;
}
}
- }
// Apply legato-specific sound signal modifications
- legato.apply(*this,outl,outr);
+ legato.apply(*this, outl, outr);
// Check if the global amplitude is finished.
@@ -1743,9 +1737,9 @@ void ADnote::Voice::releasekey()
}
template<class T>
-static inline void nullify(T &t) {delete t; t=NULL;}
+static inline void nullify(T &t) {delete t; t = NULL;}
template<class T>
-static inline void arrayNullify(T &t) {delete [] t; t=NULL;}
+static inline void arrayNullify(T &t) {delete [] t; t = NULL;}
void ADnote::Voice::kill()
{
@@ -1792,18 +1786,18 @@ void ADnote::Global::initparameters(const ADnoteGlobalParam ¶m,
FreqEnvelope = new Envelope(param.FreqEnvelope, basefreq);
FreqLfo = new LFO(param.FreqLfo, basefreq);
- AmpEnvelope = new Envelope(param.AmpEnvelope, basefreq);
- AmpLfo = new LFO(param.AmpLfo, basefreq);
+ AmpEnvelope = new Envelope(param.AmpEnvelope, basefreq);
+ AmpLfo = new LFO(param.AmpLfo, basefreq);
Volume = 4.0f * powf(0.1f, 3.0f * (1.0f - param.PVolume / 96.0f)) //-60 dB .. 0 dB
- * VelF(velocity, param.PAmpVelocityScaleFunction); //sensing
+ * VelF(velocity, param.PAmpVelocityScaleFunction); //sensing
GlobalFilterL = Filter::generate(param.GlobalFilter);
if(stereo)
GlobalFilterR = Filter::generate(param.GlobalFilter);
- FilterEnvelope = new Envelope(param.FilterEnvelope, basefreq);
- FilterLfo = new LFO(param.FilterLfo, basefreq);
+ FilterEnvelope = new Envelope(param.FilterEnvelope, basefreq);
+ FilterLfo = new LFO(param.FilterLfo, basefreq);
FilterQ = param.GlobalFilter->getq();
FilterFreqTracking = param.GlobalFilter->getfreqtracking(basefreq);
}
diff --git a/src/Synth/ADnote.h b/src/Synth/ADnote.h
@@ -37,7 +37,7 @@
#define OSCIL_SMP_EXTRA_SAMPLES 5
/**The "additive" synthesizer*/
-class ADnote :public SynthNote
+class ADnote:public SynthNote
{
public:
/**Constructor.
@@ -114,7 +114,7 @@ class ADnote :public SynthNote
//GLOBALS
ADnoteParameters *partparams;
unsigned char stereo; //if the note is stereo (allows note Panning)
- int midinote;
+ int midinote;
float velocity, basefreq;
ONOFFTYPE NoteEnabled;
@@ -132,7 +132,7 @@ class ADnote :public SynthNote
/******************************************
* FREQUENCY GLOBAL PARAMETERS *
******************************************/
- float Detune; //cents
+ float Detune; //cents
Envelope *FreqEnvelope;
LFO *FreqLfo;
@@ -140,30 +140,30 @@ class ADnote :public SynthNote
/********************************************
* AMPLITUDE GLOBAL PARAMETERS *
********************************************/
- float Volume; // [ 0 .. 1 ]
+ float Volume; // [ 0 .. 1 ]
- float Panning; // [ 0 .. 1 ]
+ float Panning; // [ 0 .. 1 ]
Envelope *AmpEnvelope;
LFO *AmpLfo;
struct {
- int Enabled;
+ int Enabled;
float initialvalue, dt, t;
} Punch;
/******************************************
* FILTER GLOBAL PARAMETERS *
******************************************/
- class Filter *GlobalFilterL, *GlobalFilterR;
+ class Filter *GlobalFilterL, *GlobalFilterR;
- float FilterCenterPitch; //octaves
- float FilterQ;
- float FilterFreqTracking;
+ float FilterCenterPitch; //octaves
+ float FilterQ;
+ float FilterFreqTracking;
Envelope *FilterEnvelope;
- LFO *FilterLfo;
+ LFO *FilterLfo;
} NoteGlobalPar;
@@ -171,7 +171,7 @@ class ADnote :public SynthNote
/***********************************************************/
/* VOICE PARAMETERS */
/***********************************************************/
- struct Voice{
+ struct Voice {
void releasekey();
void kill();
/* If the voice is enabled */
@@ -196,7 +196,7 @@ class ADnote :public SynthNote
int fixedfreqET; //if the "fixed" frequency varies according to the note (ET)
// cents = basefreq*VoiceDetune
- float Detune, FineDetune;
+ float Detune, FineDetune;
Envelope *FreqEnvelope;
LFO *FreqLfo;
@@ -207,8 +207,8 @@ class ADnote :public SynthNote
***************************/
/* Panning 0.0f=left, 0.5f - center, 1.0f = right */
- float Panning;
- float Volume; // [-1.0f .. 1.0f]
+ float Panning;
+ float Volume; // [-1.0f .. 1.0f]
Envelope *AmpEnvelope;
LFO *AmpLfo;
@@ -217,11 +217,11 @@ class ADnote :public SynthNote
* FILTER PARAMETERS *
*************************/
- class Filter *VoiceFilterL;
- class Filter *VoiceFilterR;
+ class Filter *VoiceFilterL;
+ class Filter *VoiceFilterR;
- float FilterCenterPitch; /* Filter center Pitch*/
- float FilterFreqTracking;
+ float FilterCenterPitch; /* Filter center Pitch*/
+ float FilterFreqTracking;
Envelope *FilterEnvelope;
LFO *FilterLfo;
@@ -233,7 +233,7 @@ class ADnote :public SynthNote
FMTYPE FMEnabled;
- int FMVoice;
+ int FMVoice;
// Voice Output used by other voices if use this as modullator
float *VoiceOut;
@@ -241,8 +241,8 @@ class ADnote :public SynthNote
/* Wave of the Voice */
float *FMSmp;
- float FMVolume;
- float FMDetune; //in cents
+ float FMVolume;
+ float FMDetune; //in cents
Envelope *FMFreqEnvelope;
Envelope *FMAmpEnvelope;
@@ -293,17 +293,17 @@ class ADnote :public SynthNote
//used to compute and interpolate the amplitudes of voices and modullators
float oldamplitude[NUM_VOICES],
- newamplitude[NUM_VOICES],
- FMoldamplitude[NUM_VOICES],
- FMnewamplitude[NUM_VOICES];
+ newamplitude[NUM_VOICES],
+ FMoldamplitude[NUM_VOICES],
+ FMnewamplitude[NUM_VOICES];
//used by Frequency Modulation (for integration)
float *FMoldsmp[NUM_VOICES];
//temporary buffer
- float *tmpwavel;
- float *tmpwaver;
- int max_unison;
+ float *tmpwavel;
+ float *tmpwaver;
+ int max_unison;
float **tmpwave_unison;
//Filter bypass samples
@@ -323,4 +323,3 @@ class ADnote :public SynthNote
};
#endif
-
diff --git a/src/Synth/Envelope.cpp b/src/Synth/Envelope.cpp
@@ -27,7 +27,7 @@
Envelope::Envelope(EnvelopeParams *envpars, float basefreq)
{
int i;
- envpoints = envpars->Penvpoints;
+ envpoints = envpars->Penvpoints;
if(envpoints > MAX_ENVELOPE_POINTS)
envpoints = MAX_ENVELOPE_POINTS;
envsustain = (envpars->Penvsustain == 0) ? -1 : envpars->Penvsustain;
@@ -56,35 +56,36 @@ Envelope::Envelope(EnvelopeParams *envpars, float basefreq)
envdt[i] = 2.0f; //any value larger than 1
switch(mode) {
- case 2:
- envval[i] = (1.0f - envpars->Penvval[i] / 127.0f) * -40;
- break;
- case 3:
- envval[i] =
- (powf(2, 6.0f
- * fabs(envpars->Penvval[i] - 64.0f) / 64.0f) - 1.0f) * 100.0f;
- if(envpars->Penvval[i] < 64)
- envval[i] = -envval[i];
- break;
- case 4:
- envval[i] = (envpars->Penvval[i] - 64.0f) / 64.0f * 6.0f; //6 octaves (filtru)
- break;
- case 5:
- envval[i] = (envpars->Penvval[i] - 64.0f) / 64.0f * 10;
- break;
- default:
- envval[i] = envpars->Penvval[i] / 127.0f;
+ case 2:
+ envval[i] = (1.0f - envpars->Penvval[i] / 127.0f) * -40;
+ break;
+ case 3:
+ envval[i] =
+ (powf(2, 6.0f
+ * fabs(envpars->Penvval[i]
+ - 64.0f) / 64.0f) - 1.0f) * 100.0f;
+ if(envpars->Penvval[i] < 64)
+ envval[i] = -envval[i];
+ break;
+ case 4:
+ envval[i] = (envpars->Penvval[i] - 64.0f) / 64.0f * 6.0f; //6 octaves (filtru)
+ break;
+ case 5:
+ envval[i] = (envpars->Penvval[i] - 64.0f) / 64.0f * 10;
+ break;
+ default:
+ envval[i] = envpars->Penvval[i] / 127.0f;
}
}
- envdt[0] = 1.0f;
+ envdt[0] = 1.0f;
currentpoint = 1; //the envelope starts from 1
keyreleased = false;
t = 0.0f;
- envfinish = false;
- inct = envdt[1];
- envoutval = 0.0f;
+ envfinish = false;
+ inct = envdt[1];
+ envoutval = 0.0f;
}
Envelope::~Envelope()
@@ -141,8 +142,8 @@ float Envelope::envout()
if(inct >= 1.0f)
out = envval[currentpoint];
else
- out = envval[currentpoint - 1] +
- (envval[currentpoint] - envval[currentpoint - 1]) * t;
+ out = envval[currentpoint - 1]
+ + (envval[currentpoint] - envval[currentpoint - 1]) * t;
t += inct;
if(t >= 1.0f) {
@@ -172,12 +173,12 @@ float Envelope::envout_dB()
float v2 = dB2rap(envval[1]);
out = v1 + (v2 - v1) * t;
- t += inct;
+ t += inct;
if(t >= 1.0f) {
t = 0.0f;
inct = envdt[2];
currentpoint++;
- out = v2;
+ out = v2;
}
if(out > 0.001f)
@@ -195,4 +196,3 @@ bool Envelope::finished() const
{
return envfinish;
}
-
diff --git a/src/Synth/Envelope.h b/src/Synth/Envelope.h
@@ -42,17 +42,17 @@ class Envelope
* @return returns 1 if the envelope is finished*/
bool finished() const;
private:
- int envpoints;
- int envsustain; //"-1" means disabled
+ int envpoints;
+ int envsustain; //"-1" means disabled
float envdt[MAX_ENVELOPE_POINTS]; //millisecons
float envval[MAX_ENVELOPE_POINTS]; // [0.0f .. 1.0f]
float envstretch;
- int linearenvelope;
+ int linearenvelope;
- int currentpoint; //current envelope point (starts from 1)
- int forcedrelase;
- bool keyreleased; //if the key was released
- bool envfinish;
+ int currentpoint; //current envelope point (starts from 1)
+ int forcedrelase;
+ bool keyreleased; //if the key was released
+ bool envfinish;
float t; // the time from the last point
float inct; // the time increment
float envoutval; //used to do the forced release
@@ -60,4 +60,3 @@ class Envelope
#endif
-
diff --git a/src/Synth/LFO.cpp b/src/Synth/LFO.cpp
@@ -32,9 +32,9 @@ LFO::LFO(LFOParams *lfopars, float basefreq)
if(lfopars->Pstretch == 0)
lfopars->Pstretch = 1;
float lfostretch = powf(basefreq / 440.0f,
- (lfopars->Pstretch - 64.0f) / 63.0f); //max 2x/octave
+ (lfopars->Pstretch - 64.0f) / 63.0f); //max 2x/octave
- float lfofreq =
+ float lfofreq =
(powf(2, lfopars->Pfreq * 10.0f) - 1.0f) / 12.0f * lfostretch;
incx = fabs(lfofreq) * (float)SOUND_BUFFER_SIZE / (float)SAMPLE_RATE;
@@ -65,16 +65,16 @@ LFO::LFO(LFOParams *lfopars, float basefreq)
lfofreqrnd = powf(lfopars->Pfreqrand / 127.0f, 2.0f) * 4.0f;
switch(lfopars->fel) {
- case 1:
- lfointensity = lfopars->Pintensity / 127.0f;
- break;
- case 2:
- lfointensity = lfopars->Pintensity / 127.0f * 4.0f;
- break; //in octave
- default:
- lfointensity = powf(2, lfopars->Pintensity / 127.0f * 11.0f) - 1.0f; //in centi
- x -= 0.25f; //chance the starting phase
- break;
+ case 1:
+ lfointensity = lfopars->Pintensity / 127.0f;
+ break;
+ case 2:
+ lfointensity = lfopars->Pintensity / 127.0f * 4.0f;
+ break; //in octave
+ default:
+ lfointensity = powf(2, lfopars->Pintensity / 127.0f * 11.0f) - 1.0f; //in centi
+ x -= 0.25f; //chance the starting phase
+ break;
}
amp1 = (1 - lfornd) + lfornd * RND;
@@ -97,35 +97,35 @@ float LFO::lfoout()
{
float out;
switch(lfotype) {
- case 1: //LFO_TRIANGLE
- if((x >= 0.0f) && (x < 0.25f))
- out = 4.0f * x;
- else
- if((x > 0.25f) && (x < 0.75f))
- out = 2 - 4 * x;
- else
- out = 4.0f * x - 4.0f;
- break;
- case 2: //LFO_SQUARE
- if(x < 0.5f)
- out = -1;
- else
- out = 1;
- break;
- case 3: //LFO_RAMPUP
- out = (x - 0.5f) * 2.0f;
- break;
- case 4: //LFO_RAMPDOWN
- out = (0.5f - x) * 2.0f;
- break;
- case 5: //LFO_EXP_DOWN 1
- out = powf(0.05f, x) * 2.0f - 1.0f;
- break;
- case 6: //LFO_EXP_DOWN 2
- out = powf(0.001f, x) * 2.0f - 1.0f;
- break;
- default:
- out = cosf(x * 2.0f * PI); //LFO_SINE
+ case 1: //LFO_TRIANGLE
+ if((x >= 0.0f) && (x < 0.25f))
+ out = 4.0f * x;
+ else
+ if((x > 0.25f) && (x < 0.75f))
+ out = 2 - 4 * x;
+ else
+ out = 4.0f * x - 4.0f;
+ break;
+ case 2: //LFO_SQUARE
+ if(x < 0.5f)
+ out = -1;
+ else
+ out = 1;
+ break;
+ case 3: //LFO_RAMPUP
+ out = (x - 0.5f) * 2.0f;
+ break;
+ case 4: //LFO_RAMPDOWN
+ out = (0.5f - x) * 2.0f;
+ break;
+ case 5: //LFO_EXP_DOWN 1
+ out = powf(0.05f, x) * 2.0f - 1.0f;
+ break;
+ case 6: //LFO_EXP_DOWN 2
+ out = powf(0.001f, x) * 2.0f - 1.0f;
+ break;
+ default:
+ out = cosf(x * 2.0f * PI); //LFO_SINE
}
@@ -181,4 +181,3 @@ void LFO::computenextincrnd()
incrnd = nextincrnd;
nextincrnd = powf(0.5f, lfofreqrnd) + RND * (powf(2.0f, lfofreqrnd) - 1.0f);
}
-
diff --git a/src/Synth/LFO.h b/src/Synth/LFO.h
@@ -56,4 +56,3 @@ class LFO
};
#endif
-
diff --git a/src/Synth/OscilGen.cpp b/src/Synth/OscilGen.cpp
@@ -77,7 +77,7 @@ void normalize(fft_t *freqs)
if(max < 1e-8) //data is all ~zero, do not amplify noise
return;
- for(int i=0; i < OSCIL_SIZE / 2; ++i)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
freqs[i] /= max;
}
@@ -97,15 +97,15 @@ void rmsNormalize(fft_t *freqs)
freqs[i] *= gain;
}
-#define DIFF(par) (old##par != P##par)
+#define DIFF(par) (old ## par != P ## par)
OscilGen::OscilGen(FFTwrapper *fft_, Resonance *res_):Presets()
{
assert(fft_);
setpresettype("Poscilgen");
- fft = fft_;
- res = res_;
+ fft = fft_;
+ res = res_;
tmpsmps = new float[OSCIL_SIZE];
@@ -133,17 +133,17 @@ void OscilGen::defaults()
oldbasefunc = 0;
oldbasepar = 64;
oldhmagtype = 0;
- oldwaveshapingfunction = 0;
- oldwaveshaping = 64;
+ oldwaveshapingfunction = 0;
+ oldwaveshaping = 64;
oldbasefuncmodulation = 0;
oldharmonicshift = 0;
oldbasefuncmodulationpar1 = 0;
oldbasefuncmodulationpar2 = 0;
oldbasefuncmodulationpar3 = 0;
- oldmodulation = 0;
- oldmodulationpar1 = 0;
- oldmodulationpar2 = 0;
- oldmodulationpar3 = 0;
+ oldmodulation = 0;
+ oldmodulationpar1 = 0;
+ oldmodulationpar2 = 0;
+ oldmodulationpar3 = 0;
for(int i = 0; i < MAX_AD_HARMONICS; ++i) {
hmag[i] = 0.0f;
@@ -166,27 +166,27 @@ void OscilGen::defaults()
Pbasefuncmodulationpar2 = 64;
Pbasefuncmodulationpar3 = 32;
- Pmodulation = 0;
- Pmodulationpar1 = 64;
- Pmodulationpar2 = 64;
- Pmodulationpar3 = 32;
+ Pmodulation = 0;
+ Pmodulationpar1 = 64;
+ Pmodulationpar2 = 64;
+ Pmodulationpar3 = 32;
Pwaveshapingfunction = 0;
- Pwaveshaping = 64;
- Pfiltertype = 0;
- Pfilterpar1 = 64;
- Pfilterpar2 = 64;
- Pfilterbeforews = 0;
+ Pwaveshaping = 64;
+ Pfiltertype = 0;
+ Pfilterpar1 = 64;
+ Pfilterpar2 = 64;
+ Pfilterbeforews = 0;
Psatype = 0;
Psapar = 64;
Pamprandpower = 64;
- Pamprandtype = 0;
+ Pamprandtype = 0;
Pharmonicshift = 0;
Pharmonicshiftfirst = 0;
- Padaptiveharmonics = 0;
+ Padaptiveharmonics = 0;
Padaptiveharmonicspower = 100;
Padaptiveharmonicsbasefreq = 128;
Padaptiveharmonicspar = 50;
@@ -201,8 +201,8 @@ void OscilGen::defaults()
void OscilGen::convert2sine()
{
- float mag[MAX_AD_HARMONICS], phase[MAX_AD_HARMONICS];
- float oscil[OSCIL_SIZE];
+ float mag[MAX_AD_HARMONICS], phase[MAX_AD_HARMONICS];
+ float oscil[OSCIL_SIZE];
fft_t *freqs = new fft_t[OSCIL_SIZE / 2];
get(oscil, -1.0f);
@@ -225,7 +225,7 @@ void OscilGen::convert2sine()
float newmag = mag[i];
float newphase = phase[i];
- Phmag[i] = (int) ((newmag) * 64.0f) + 64;
+ Phmag[i] = (int) ((newmag) * 64.0f) + 64;
Phphase[i] = 64 - (int) (64.0f * newphase / PI);
if(Phphase[i] > 127)
@@ -243,38 +243,38 @@ void OscilGen::convert2sine()
*/
void OscilGen::getbasefunction(float *smps)
{
- int i;
+ int i;
float par = (Pbasefuncpar + 0.5f) / 128.0f;
if(Pbasefuncpar == 64)
par = 0.5f;
float basefuncmodulationpar1 = Pbasefuncmodulationpar1 / 127.0f,
- basefuncmodulationpar2 = Pbasefuncmodulationpar2 / 127.0f,
- basefuncmodulationpar3 = Pbasefuncmodulationpar3 / 127.0f;
+ basefuncmodulationpar2 = Pbasefuncmodulationpar2 / 127.0f,
+ basefuncmodulationpar3 = Pbasefuncmodulationpar3 / 127.0f;
switch(Pbasefuncmodulation) {
- case 1:
- basefuncmodulationpar1 =
- (powf(2, basefuncmodulationpar1 * 5.0f) - 1.0f) / 10.0f;
- basefuncmodulationpar3 =
- floor((powf(2, basefuncmodulationpar3 * 5.0f) - 1.0f));
- if(basefuncmodulationpar3 < 0.9999f)
- basefuncmodulationpar3 = -1.0f;
- break;
- case 2:
- basefuncmodulationpar1 =
- (powf(2, basefuncmodulationpar1 * 5.0f) - 1.0f) / 10.0f;
- basefuncmodulationpar3 = 1.0f
- + floor((powf(2, basefuncmodulationpar3
- * 5.0f) - 1.0f));
- break;
- case 3:
- basefuncmodulationpar1 =
- (powf(2, basefuncmodulationpar1 * 7.0f) - 1.0f) / 10.0f;
- basefuncmodulationpar3 = 0.01f
- + (powf(2, basefuncmodulationpar3
- * 16.0f) - 1.0f) / 10.0f;
- break;
+ case 1:
+ basefuncmodulationpar1 =
+ (powf(2, basefuncmodulationpar1 * 5.0f) - 1.0f) / 10.0f;
+ basefuncmodulationpar3 =
+ floor((powf(2, basefuncmodulationpar3 * 5.0f) - 1.0f));
+ if(basefuncmodulationpar3 < 0.9999f)
+ basefuncmodulationpar3 = -1.0f;
+ break;
+ case 2:
+ basefuncmodulationpar1 =
+ (powf(2, basefuncmodulationpar1 * 5.0f) - 1.0f) / 10.0f;
+ basefuncmodulationpar3 = 1.0f
+ + floor((powf(2, basefuncmodulationpar3
+ * 5.0f) - 1.0f));
+ break;
+ case 3:
+ basefuncmodulationpar1 =
+ (powf(2, basefuncmodulationpar1 * 7.0f) - 1.0f) / 10.0f;
+ basefuncmodulationpar3 = 0.01f
+ + (powf(2, basefuncmodulationpar3
+ * 16.0f) - 1.0f) / 10.0f;
+ break;
}
base_func func = getBaseFunction(Pcurrentbasefunc);
@@ -283,21 +283,25 @@ void OscilGen::getbasefunction(float *smps)
float t = i * 1.0f / OSCIL_SIZE;
switch(Pbasefuncmodulation) {
- case 1:
- t = t * basefuncmodulationpar3 + sinf(
- (t
- + basefuncmodulationpar2) * 2.0f * PI) * basefuncmodulationpar1;//rev
- break;
- case 2:
- t = t + sinf(
- (t * basefuncmodulationpar3
- + basefuncmodulationpar2) * 2.0f * PI) * basefuncmodulationpar1;//sine
- break;
- case 3:
- t = t + powf((1.0f - cosf(
- (t + basefuncmodulationpar2) * 2.0f * PI)) * 0.5f,
- basefuncmodulationpar3) * basefuncmodulationpar1;//power
- break;
+ case 1:
+ t = t * basefuncmodulationpar3 + sinf(
+ (t
+ + basefuncmodulationpar2) * 2.0f
+ * PI) * basefuncmodulationpar1; //rev
+ break;
+ case 2:
+ t = t + sinf(
+ (t * basefuncmodulationpar3
+ + basefuncmodulationpar2) * 2.0f
+ * PI) * basefuncmodulationpar1; //sine
+ break;
+ case 3:
+ t = t + powf((1.0f - cosf(
+ (t
+ + basefuncmodulationpar2) * 2.0f
+ * PI)) * 0.5f,
+ basefuncmodulationpar3) * basefuncmodulationpar1; //power
+ break;
}
t = t - floor(t);
@@ -318,12 +322,12 @@ void OscilGen::oscilfilter()
if(Pfiltertype == 0)
return;
- const float par = 1.0f - Pfilterpar1 / 128.0f;
- const float par2 = Pfilterpar2 / 127.0f;
+ const float par = 1.0f - Pfilterpar1 / 128.0f;
+ const float par2 = Pfilterpar2 / 127.0f;
filter_func filter = getFilter(Pfiltertype);
for(int i = 1; i < OSCIL_SIZE / 2; ++i)
- oscilFFTfreqs[i] *= filter(i,par,par2);
+ oscilFFTfreqs[i] *= filter(i, par, par2);
normalize(oscilFFTfreqs);
}
@@ -409,24 +413,26 @@ void OscilGen::modulation()
float modulationpar1 = Pmodulationpar1 / 127.0f,
- modulationpar2 = 0.5f - Pmodulationpar2 / 127.0f,
- modulationpar3 = Pmodulationpar3 / 127.0f;
+ modulationpar2 = 0.5f - Pmodulationpar2 / 127.0f,
+ modulationpar3 = Pmodulationpar3 / 127.0f;
switch(Pmodulation) {
- case 1:
- modulationpar1 = (powf(2, modulationpar1 * 7.0f) - 1.0f) / 100.0f;
- modulationpar3 = floor((powf(2, modulationpar3 * 5.0f) - 1.0f));
- if(modulationpar3 < 0.9999f)
- modulationpar3 = -1.0f;
- break;
- case 2:
- modulationpar1 = (powf(2, modulationpar1 * 7.0f) - 1.0f) / 100.0f;
- modulationpar3 = 1.0f + floor((powf(2, modulationpar3 * 5.0f) - 1.0f));
- break;
- case 3:
- modulationpar1 = (powf(2, modulationpar1 * 9.0f) - 1.0f) / 100.0f;
- modulationpar3 = 0.01f + (powf(2, modulationpar3 * 16.0f) - 1.0f) / 10.0f;
- break;
+ case 1:
+ modulationpar1 = (powf(2, modulationpar1 * 7.0f) - 1.0f) / 100.0f;
+ modulationpar3 = floor((powf(2, modulationpar3 * 5.0f) - 1.0f));
+ if(modulationpar3 < 0.9999f)
+ modulationpar3 = -1.0f;
+ break;
+ case 2:
+ modulationpar1 = (powf(2, modulationpar1 * 7.0f) - 1.0f) / 100.0f;
+ modulationpar3 = 1.0f
+ + floor((powf(2, modulationpar3 * 5.0f) - 1.0f));
+ break;
+ case 3:
+ modulationpar1 = (powf(2, modulationpar1 * 9.0f) - 1.0f) / 100.0f;
+ modulationpar3 = 0.01f
+ + (powf(2, modulationpar3 * 16.0f) - 1.0f) / 10.0f;
+ break;
}
clearDC(oscilFFTfreqs); //remove the DC
@@ -436,8 +442,8 @@ void OscilGen::modulation()
oscilFFTfreqs[OSCIL_SIZE / 2 - i] *= tmp;
}
fft->freqs2smps(oscilFFTfreqs, tmpsmps);
- int extra_points = 2;
- float *in = new float[OSCIL_SIZE + extra_points];
+ int extra_points = 2;
+ float *in = new float[OSCIL_SIZE + extra_points];
//Normalize
normalize(tmpsmps, OSCIL_SIZE);
@@ -452,25 +458,25 @@ void OscilGen::modulation()
float t = i * 1.0f / OSCIL_SIZE;
switch(Pmodulation) {
- case 1:
- t = t * modulationpar3
- + sinf((t + modulationpar2) * 2.0f * PI) * modulationpar1; //rev
- break;
- case 2:
- t = t
- + sinf((t * modulationpar3
- + modulationpar2) * 2.0f * PI) * modulationpar1; //sine
- break;
- case 3:
- t = t + powf((1.0f - cosf(
- (t + modulationpar2) * 2.0f * PI)) * 0.5f,
- modulationpar3) * modulationpar1; //power
- break;
+ case 1:
+ t = t * modulationpar3
+ + sinf((t + modulationpar2) * 2.0f * PI) * modulationpar1; //rev
+ break;
+ case 2:
+ t = t
+ + sinf((t * modulationpar3
+ + modulationpar2) * 2.0f * PI) * modulationpar1; //sine
+ break;
+ case 3:
+ t = t + powf((1.0f - cosf(
+ (t + modulationpar2) * 2.0f * PI)) * 0.5f,
+ modulationpar3) * modulationpar1; //power
+ break;
}
t = (t - floor(t)) * OSCIL_SIZE;
- int poshi = (int) t;
+ int poshi = (int) t;
float poslo = t - floor(t);
tmpsmps[i] = in[poshi] * (1.0f - poslo) + in[poshi + 1] * poslo;
@@ -490,19 +496,19 @@ void OscilGen::spectrumadjust()
return;
float par = Psapar / 127.0f;
switch(Psatype) {
- case 1:
- par = 1.0f - par * 2.0f;
- if(par >= 0.0f)
- par = powf(5.0f, par);
- else
- par = powf(8.0f, par);
- break;
- case 2:
- par = powf(10.0f, (1.0f - par) * 3.0f) * 0.25f;
- break;
- case 3:
- par = powf(10.0f, (1.0f - par) * 3.0f) * 0.25f;
- break;
+ case 1:
+ par = 1.0f - par * 2.0f;
+ if(par >= 0.0f)
+ par = powf(5.0f, par);
+ else
+ par = powf(8.0f, par);
+ break;
+ case 2:
+ par = powf(10.0f, (1.0f - par) * 3.0f) * 0.25f;
+ break;
+ case 3:
+ par = powf(10.0f, (1.0f - par) * 3.0f) * 0.25f;
+ break;
}
@@ -513,18 +519,18 @@ void OscilGen::spectrumadjust()
float phase = arg(oscilFFTfreqs, i);
switch(Psatype) {
- case 1:
- mag = powf(mag, par);
- break;
- case 2:
- if(mag < par)
- mag = 0.0f;
- break;
- case 3:
- mag /= par;
- if(mag > 1.0f)
- mag = 1.0f;
- break;
+ case 1:
+ mag = powf(mag, par);
+ break;
+ case 2:
+ if(mag < par)
+ mag = 0.0f;
+ break;
+ case 3:
+ mag /= par;
+ if(mag > 1.0f)
+ mag = 1.0f;
+ break;
}
oscilFFTfreqs[i] = std::polar<fftw_real>(mag, phase);
}
@@ -538,7 +544,7 @@ void OscilGen::shiftharmonics()
int harmonicshift = -Pharmonicshift;
fft_t h;
- if(harmonicshift > 0) {
+ if(harmonicshift > 0)
for(int i = OSCIL_SIZE / 2 - 2; i >= 0; i--) {
int oldh = i - harmonicshift;
if(oldh < 0)
@@ -547,8 +553,7 @@ void OscilGen::shiftharmonics()
h = oscilFFTfreqs[oldh + 1];
oscilFFTfreqs[i + 1] = h;
}
- }
- else {
+ else
for(int i = 0; i < OSCIL_SIZE / 2 - 1; ++i) {
int oldh = i + abs(harmonicshift);
if(oldh >= (OSCIL_SIZE / 2 - 1))
@@ -561,7 +566,6 @@ void OscilGen::shiftharmonics()
oscilFFTfreqs[i + 1] = h;
}
- }
clearDC(oscilFFTfreqs);
}
@@ -582,21 +586,21 @@ void OscilGen::prepare()
for(int i = 0; i < MAX_AD_HARMONICS; ++i) {
const float hmagnew = 1.0f - fabs(Phmag[i] / 64.0f - 1.0f);
switch(Phmagtype) {
- case 1:
- hmag[i] = expf(hmagnew * logf(0.01f));
- break;
- case 2:
- hmag[i] = expf(hmagnew * logf(0.001f));
- break;
- case 3:
- hmag[i] = expf(hmagnew * logf(0.0001f));
- break;
- case 4:
- hmag[i] = expf(hmagnew * logf(0.00001f));
- break;
- default:
- hmag[i] = 1.0f - hmagnew;
- break;
+ case 1:
+ hmag[i] = expf(hmagnew * logf(0.01f));
+ break;
+ case 2:
+ hmag[i] = expf(hmagnew * logf(0.001f));
+ break;
+ case 3:
+ hmag[i] = expf(hmagnew * logf(0.0001f));
+ break;
+ case 4:
+ hmag[i] = expf(hmagnew * logf(0.00001f));
+ break;
+ default:
+ hmag[i] = 1.0f - hmagnew;
+ break;
}
if(Phmag[i] < 64)
@@ -610,13 +614,16 @@ void OscilGen::prepare()
clearAll(oscilFFTfreqs);
- if(Pcurrentbasefunc == 0) { //the sine case
+ if(Pcurrentbasefunc == 0) //the sine case
for(int i = 0; i < MAX_AD_HARMONICS; ++i) {
- oscilFFTfreqs[i + 1].real() = -hmag[i] * sinf(hphase[i] * (i + 1)) / 2.0f;
- oscilFFTfreqs[i + 1].imag() = hmag[i] * cosf(hphase[i] * (i + 1)) / 2.0f;
+ oscilFFTfreqs[i
+ + 1].real() = -hmag[i]
+ * sinf(hphase[i] * (i + 1)) / 2.0f;
+ oscilFFTfreqs[i
+ + 1].imag() = hmag[i]
+ * cosf(hphase[i] * (i + 1)) / 2.0f;
}
- }
- else {
+ else
for(int j = 0; j < MAX_AD_HARMONICS; ++j) {
if(Phmag[j] == 64)
continue;
@@ -624,10 +631,11 @@ void OscilGen::prepare()
int k = i * (j + 1);
if(k >= OSCIL_SIZE / 2)
break;
- oscilFFTfreqs[k] += basefuncFFTfreqs[i] * std::polar<fftw_real>(hmag[j], hphase[j] * k);
+ oscilFFTfreqs[k] += basefuncFFTfreqs[i] * std::polar<fftw_real>(
+ hmag[j],
+ hphase[j] * k);
}
}
- }
if(Pharmonicshiftfirst != 0)
shiftharmonics();
@@ -651,7 +659,7 @@ void OscilGen::prepare()
oldhmagtype = Phmagtype;
oldharmonicshift = Pharmonicshift + Pharmonicshiftfirst * 256;
- oscilprepared = 1;
+ oscilprepared = 1;
}
void OscilGen::adaptiveharmonic(fft_t *f, float freq)
@@ -671,7 +679,7 @@ void OscilGen::adaptiveharmonic(fft_t *f, float freq)
float basefreq = 30.0f * powf(10.0f, Padaptiveharmonicsbasefreq / 128.0f);
float power = (Padaptiveharmonicspower + 1.0f) / 101.0f;
- float rap = freq / basefreq;
+ float rap = freq / basefreq;
rap = powf(rap, power);
@@ -683,7 +691,7 @@ void OscilGen::adaptiveharmonic(fft_t *f, float freq)
for(int i = 0; i < OSCIL_SIZE / 2 - 2; ++i) {
float h = i * rap;
- int high = (int)(i * rap);
+ int high = (int)(i * rap);
float low = fmod(h, 1.0f);
if(high >= (OSCIL_SIZE / 2 - 2))
@@ -696,8 +704,10 @@ void OscilGen::adaptiveharmonic(fft_t *f, float freq)
f[high + 1].imag() += inf[i].imag() * low;
}
else {
- hc = inf[high].real() * (1.0f - low) + inf[high + 1].real() * low;
- hs = inf[high].imag() * (1.0f - low) + inf[high + 1].imag() * low;
+ hc = inf[high].real()
+ * (1.0f - low) + inf[high + 1].real() * low;
+ hs = inf[high].imag()
+ * (1.0f - low) + inf[high + 1].imag() * low;
}
if(fabs(hc) < 0.000001f)
hc = 0.0f;
@@ -729,7 +739,7 @@ void OscilGen::adaptiveharmonicpostprocess(fft_t *f, int size)
for(int i = 0; i < size; ++i) {
inf[i] = f[i] * double(par);
- f[i] *= (1.0f - par);
+ f[i] *= (1.0f - par);
}
@@ -742,15 +752,13 @@ void OscilGen::adaptiveharmonicpostprocess(fft_t *f, int size)
int nh = (Padaptiveharmonics - 3) / 2 + 2;
int sub_vs_add = (Padaptiveharmonics - 3) % 2;
if(sub_vs_add == 0) {
- for(int i = 0; i < size; ++i) {
+ for(int i = 0; i < size; ++i)
if(((i + 1) % nh) == 0)
f[i] += inf[i];
- }
}
- else {
+ else
for(int i = 0; i < size / nh - 1; ++i)
f[(i + 1) * nh - 1] += inf[i];
- }
}
delete [] inf;
@@ -786,12 +794,12 @@ bool OscilGen::needPrepare(void)
//Check function modulation
if(DIFF(basefuncmodulation) || DIFF(basefuncmodulationpar1)
- || DIFF(basefuncmodulationpar2) || DIFF(basefuncmodulationpar3))
+ || DIFF(basefuncmodulationpar2) || DIFF(basefuncmodulationpar3))
outdated = true;
//Check overall modulation
if(DIFF(modulation) || DIFF(modulationpar1)
- || DIFF(modulationpar2) || DIFF(modulationpar3))
+ || DIFF(modulationpar2) || DIFF(modulationpar3))
outdated = true;
//Check harmonic shifts
@@ -810,7 +818,8 @@ short int OscilGen::get(float *smps, float freqHz, int resonance)
prepare();
int outpos =
- (int)((RND * 2.0f - 1.0f) * (float) OSCIL_SIZE * (Prand - 64.0f) / 64.0f);
+ (int)((RND * 2.0f
+ - 1.0f) * (float) OSCIL_SIZE * (Prand - 64.0f) / 64.0f);
outpos = (outpos + 2 * OSCIL_SIZE) % OSCIL_SIZE;
@@ -837,17 +846,17 @@ short int OscilGen::get(float *smps, float freqHz, int resonance)
nyquist = realnyquist;
}
- if(Padaptiveharmonics) { //do the antialiasing in the case of adaptive harmonics
+ if(Padaptiveharmonics) //do the antialiasing in the case of adaptive harmonics
for(int i = nyquist; i < OSCIL_SIZE / 2; ++i)
outoscilFFTfreqs[i] = fft_t(0.0f, 0.0f);
- }
// Randomness (each harmonic), the block type is computed
// in ADnote by setting start position according to this setting
if((Prand > 64) && (freqHz >= 0.0f) && (!ADvsPAD)) {
const float rnd = PI * powf((Prand - 64.0f) / 64.0f, 2.0f);
for(int i = 1; i < nyquist - 1; ++i) //to Nyquist only for AntiAliasing
- outoscilFFTfreqs[i] *= std::polar<fftw_real>(1.0f, (float)(rnd * i * RND));
+ outoscilFFTfreqs[i] *=
+ std::polar<fftw_real>(1.0f, (float)(rnd * i * RND));
}
//Harmonic Amplitude Randomness
@@ -857,20 +866,20 @@ short int OscilGen::get(float *smps, float freqHz, int resonance)
float power = Pamprandpower / 127.0f;
float normalize = 1.0f / (1.2f - power);
switch(Pamprandtype) {
- case 1:
- power = power * 2.0f - 0.5f;
- power = powf(15.0f, power);
- for(int i = 1; i < nyquist - 1; ++i)
- outoscilFFTfreqs[i] *= powf(RND, power) * normalize;
- break;
- case 2:
- power = power * 2.0f - 0.5f;
- power = powf(15.0f, power) * 2.0f;
- float rndfreq = 2 * PI * RND;
- for(int i = 1; i < nyquist - 1; ++i)
- outoscilFFTfreqs[i] *= powf(fabs(sinf(i * rndfreq)), power)
- * normalize;
- break;
+ case 1:
+ power = power * 2.0f - 0.5f;
+ power = powf(15.0f, power);
+ for(int i = 1; i < nyquist - 1; ++i)
+ outoscilFFTfreqs[i] *= powf(RND, power) * normalize;
+ break;
+ case 2:
+ power = power * 2.0f - 0.5f;
+ power = powf(15.0f, power) * 2.0f;
+ float rndfreq = 2 * PI * RND;
+ for(int i = 1; i < nyquist - 1; ++i)
+ outoscilFFTfreqs[i] *= powf(fabs(sinf(i * rndfreq)), power)
+ * normalize;
+ break;
}
sprng(realrnd + 1);
}
@@ -918,7 +927,7 @@ void OscilGen::getspectrum(int n, float *spc, int what)
if(what == 0) {
for(int i = 0; i < n; ++i)
outoscilFFTfreqs[i] = fft_t(spc[i], spc[i]);
- memset(outoscilFFTfreqs+n, 0, (OSCIL_SIZE / 2 - n) * sizeof(fft_t));
+ memset(outoscilFFTfreqs + n, 0, (OSCIL_SIZE / 2 - n) * sizeof(fft_t));
adaptiveharmonic(outoscilFFTfreqs, 0.0f);
adaptiveharmonicpostprocess(outoscilFFTfreqs, n - 1);
for(int i = 0; i < n; ++i)
@@ -1022,63 +1031,63 @@ void OscilGen::getfromXML(XMLwrapper *xml)
{
Phmagtype = xml->getpar127("harmonic_mag_type", Phmagtype);
- Pcurrentbasefunc = xml->getpar127("base_function", Pcurrentbasefunc);
- Pbasefuncpar = xml->getpar127("base_function_par", Pbasefuncpar);
+ Pcurrentbasefunc = xml->getpar127("base_function", Pcurrentbasefunc);
+ Pbasefuncpar = xml->getpar127("base_function_par", Pbasefuncpar);
Pbasefuncmodulation = xml->getpar127("base_function_modulation",
Pbasefuncmodulation);
- Pbasefuncmodulationpar1 = xml->getpar127("base_function_modulation_par1",
- Pbasefuncmodulationpar1);
- Pbasefuncmodulationpar2 = xml->getpar127("base_function_modulation_par2",
- Pbasefuncmodulationpar2);
- Pbasefuncmodulationpar3 = xml->getpar127("base_function_modulation_par3",
- Pbasefuncmodulationpar3);
-
- Pmodulation = xml->getpar127("modulation", Pmodulation);
- Pmodulationpar1 = xml->getpar127("modulation_par1",
- Pmodulationpar1);
- Pmodulationpar2 = xml->getpar127("modulation_par2",
- Pmodulationpar2);
- Pmodulationpar3 = xml->getpar127("modulation_par3",
- Pmodulationpar3);
-
- Pwaveshaping = xml->getpar127("wave_shaping", Pwaveshaping);
- Pwaveshapingfunction = xml->getpar127("wave_shaping_function",
- Pwaveshapingfunction);
-
- Pfiltertype = xml->getpar127("filter_type", Pfiltertype);
- Pfilterpar1 = xml->getpar127("filter_par1", Pfilterpar1);
- Pfilterpar2 = xml->getpar127("filter_par2", Pfilterpar2);
- Pfilterbeforews = xml->getpar127("filter_before_wave_shaping",
- Pfilterbeforews);
-
- Psatype = xml->getpar127("spectrum_adjust_type", Psatype);
- Psapar = xml->getpar127("spectrum_adjust_par", Psapar);
-
- Prand = xml->getpar127("rand", Prand);
- Pamprandtype = xml->getpar127("amp_rand_type", Pamprandtype);
- Pamprandpower = xml->getpar127("amp_rand_power", Pamprandpower);
-
- Pharmonicshift = xml->getpar("harmonic_shift",
- Pharmonicshift,
- -64,
- 64);
- Pharmonicshiftfirst = xml->getparbool("harmonic_shift_first",
- Pharmonicshiftfirst);
-
- Padaptiveharmonics = xml->getpar("adaptive_harmonics",
- Padaptiveharmonics,
- 0,
- 127);
+ Pbasefuncmodulationpar1 = xml->getpar127("base_function_modulation_par1",
+ Pbasefuncmodulationpar1);
+ Pbasefuncmodulationpar2 = xml->getpar127("base_function_modulation_par2",
+ Pbasefuncmodulationpar2);
+ Pbasefuncmodulationpar3 = xml->getpar127("base_function_modulation_par3",
+ Pbasefuncmodulationpar3);
+
+ Pmodulation = xml->getpar127("modulation", Pmodulation);
+ Pmodulationpar1 = xml->getpar127("modulation_par1",
+ Pmodulationpar1);
+ Pmodulationpar2 = xml->getpar127("modulation_par2",
+ Pmodulationpar2);
+ Pmodulationpar3 = xml->getpar127("modulation_par3",
+ Pmodulationpar3);
+
+ Pwaveshaping = xml->getpar127("wave_shaping", Pwaveshaping);
+ Pwaveshapingfunction = xml->getpar127("wave_shaping_function",
+ Pwaveshapingfunction);
+
+ Pfiltertype = xml->getpar127("filter_type", Pfiltertype);
+ Pfilterpar1 = xml->getpar127("filter_par1", Pfilterpar1);
+ Pfilterpar2 = xml->getpar127("filter_par2", Pfilterpar2);
+ Pfilterbeforews = xml->getpar127("filter_before_wave_shaping",
+ Pfilterbeforews);
+
+ Psatype = xml->getpar127("spectrum_adjust_type", Psatype);
+ Psapar = xml->getpar127("spectrum_adjust_par", Psapar);
+
+ Prand = xml->getpar127("rand", Prand);
+ Pamprandtype = xml->getpar127("amp_rand_type", Pamprandtype);
+ Pamprandpower = xml->getpar127("amp_rand_power", Pamprandpower);
+
+ Pharmonicshift = xml->getpar("harmonic_shift",
+ Pharmonicshift,
+ -64,
+ 64);
+ Pharmonicshiftfirst = xml->getparbool("harmonic_shift_first",
+ Pharmonicshiftfirst);
+
+ Padaptiveharmonics = xml->getpar("adaptive_harmonics",
+ Padaptiveharmonics,
+ 0,
+ 127);
Padaptiveharmonicsbasefreq = xml->getpar(
"adaptive_harmonics_base_frequency",
Padaptiveharmonicsbasefreq,
0,
255);
- Padaptiveharmonicspower = xml->getpar("adaptive_harmonics_power",
- Padaptiveharmonicspower,
- 0,
- 200);
+ Padaptiveharmonicspower = xml->getpar("adaptive_harmonics_power",
+ Padaptiveharmonicspower,
+ 0,
+ 200);
if(xml->enterbranch("HARMONICS")) {
@@ -1099,13 +1108,12 @@ void OscilGen::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("BASE_FUNCTION")) {
- for(int i = 1; i < OSCIL_SIZE / 2; ++i) {
+ for(int i = 1; i < OSCIL_SIZE / 2; ++i)
if(xml->enterbranch("BF_HARMONIC", i)) {
basefuncFFTfreqs[i].real() = xml->getparreal("cos", 0.0f);
basefuncFFTfreqs[i].imag() = xml->getparreal("sin", 0.0f);
xml->exitbranch();
}
- }
xml->exitbranch();
clearDC(basefuncFFTfreqs);
@@ -1115,7 +1123,7 @@ void OscilGen::getfromXML(XMLwrapper *xml)
//Define basic functions
-#define FUNC(b) float basefunc_##b(float x, float a)
+#define FUNC(b) float basefunc_ ## b(float x, float a)
FUNC(pulse)
{
@@ -1258,7 +1266,7 @@ FUNC(sqr)
return -atanf(sinf(x * 2.0f * PI) * a);
}
-typedef float(*base_func)(float,float);
+typedef float (*base_func)(float, float);
base_func getBaseFunction(unsigned char func)
{
if(!func)
@@ -1282,13 +1290,14 @@ base_func getBaseFunction(unsigned char func)
basefunc_chirp,
basefunc_absstretchsine,
basefunc_chebyshev,
- basefunc_sqr,};
+ basefunc_sqr,
+ };
return functions[func];
}
//And filters
-#define FILTER(x) float osc_##x(unsigned int i, float par, float par2)
+#define FILTER(x) float osc_ ## x(unsigned int i, float par, float par2)
FILTER(lp)
{
float gain = powf(1.0f - par * par * par * 0.99f, i);
@@ -1308,8 +1317,9 @@ FILTER(hp1b)
{
if(par < 0.2f)
par = par * 0.25f + 0.15f;
- float gain = 1.0f - powf(1.0f - par * par * 0.999f + 0.001f, i * 0.05f * i + 1.0f);
- float tmp = powf(5.0f, par2 * 2.0f);
+ float gain = 1.0f - powf(1.0f - par * par * 0.999f + 0.001f,
+ i * 0.05f * i + 1.0f);
+ float tmp = powf(5.0f, par2 * 2.0f);
return powf(gain, tmp);
}
@@ -1317,7 +1327,7 @@ FILTER(bp1)
{
float gain = i + 1 - powf(2, (1.0f - par) * 7.5f);
gain = 1.0f / (1.0f + gain * gain / (i + 1.0f));
- float tmp = powf(5.0f, par2 * 2.0f);
+ float tmp = powf(5.0f, par2 * 2.0f);
gain = powf(gain, tmp);
if(gain < 1e-5)
gain = 1e-5;
@@ -1333,24 +1343,34 @@ FILTER(bs1)
FILTER(lp2)
{
- return (i + 1 > powf(2, (1.0f - par) * 10) ? 0.0f : 1.0f) * par2 + (1.0f - par2);
+ return (i + 1 >
+ powf(2, (1.0f - par) * 10) ? 0.0f : 1.0f) * par2 + (1.0f - par2);
}
FILTER(hp2)
{
if(par == 1)
return 1.0f;
- return (i + 1 > powf(2, (1.0f - par) * 7) ? 1.0f : 0.0f) * par2 + (1.0f - par2);
+ return (i + 1 >
+ powf(2, (1.0f - par) * 7) ? 1.0f : 0.0f) * par2 + (1.0f - par2);
}
FILTER(bp2)
{
- return (fabs(powf(2, (1.0f - par) * 7) - i) > i / 2 + 1 ? 0.0f : 1.0f) * par2 + (1.0f - par2);
+ return (fabs(powf(2,
+ (1.0f
+ - par)
+ * 7)
+ - i) > i / 2 + 1 ? 0.0f : 1.0f) * par2 + (1.0f - par2);
}
FILTER(bs2)
{
- return (fabs(powf(2, (1.0f - par) * 7) - i) < i / 2 + 1 ? 0.0f : 1.0f) * par2 + (1.0f - par2);
+ return (fabs(powf(2,
+ (1.0f
+ - par)
+ * 7)
+ - i) < i / 2 + 1 ? 0.0f : 1.0f) * par2 + (1.0f - par2);
}
bool floatEq(float a, float b)
@@ -1365,7 +1385,7 @@ FILTER(cos)
tmp = powf(i / 32.0f, tmp) * 32.0f;
if(floatEq(par2 * 127.0f, 64.0f))
tmp = i;
- float gain = cosf(par * par * PI / 2.0f * tmp);
+ float gain = cosf(par * par * PI / 2.0f * tmp);
gain *= gain;
return gain;
}
@@ -1388,15 +1408,15 @@ FILTER(low_shelf)
if(x < 0.0f)
x = 0.0f;
else
- if(x > 1.0f)
- x = 1.0f;
- float tmp = powf(1.0f - par2, 2.0f);
+ if(x > 1.0f)
+ x = 1.0f;
+ float tmp = powf(1.0f - par2, 2.0f);
return cosf(x * PI) * (1.0f - tmp) + 1.01f + tmp;
}
FILTER(s)
{
- unsigned int tmp = (int) (powf(2.0f, (1.0f - par) * 7.2f));
+ unsigned int tmp = (int) (powf(2.0f, (1.0f - par) * 7.2f));
float gain = 1.0f;
if(i == tmp)
gain = powf(2.0f, par2 * par2 * 8.0f);
@@ -1404,7 +1424,7 @@ FILTER(s)
}
#undef FILTER
-typedef float(*filter_func)(unsigned int, float, float);
+typedef float (*filter_func)(unsigned int, float, float);
filter_func getFilter(unsigned char func)
{
if(!func)
@@ -1425,7 +1445,7 @@ filter_func getFilter(unsigned char func)
osc_cos,
osc_sin,
osc_low_shelf,
- osc_s};
+ osc_s
+ };
return functions[func];
}
-
diff --git a/src/Synth/OscilGen.h b/src/Synth/OscilGen.h
@@ -40,7 +40,7 @@ class OscilGen:public Presets
/**do the antialiasing(cut off higher freqs.),apply randomness and do a IFFT*/
//returns where should I start getting samples, used in block type randomness
- short get(float *smps, float freqHz, int resonance=0);
+ short get(float *smps, float freqHz, int resonance = 0);
//if freqHz is smaller than 0, return the "un-randomized" sample for UI
void getbasefunction(float *smps);
@@ -79,7 +79,7 @@ class OscilGen:public Presets
unsigned char Pbasefuncmodulation; //what modulation is applied to the basefunc
unsigned char Pbasefuncmodulationpar1, Pbasefuncmodulationpar2,
- Pbasefuncmodulationpar3;//the parameter of the base function modulation
+ Pbasefuncmodulationpar3; //the parameter of the base function modulation
/*the Randomness:
64=no randomness
@@ -159,18 +159,17 @@ class OscilGen:public Presets
fft_t *basefuncFFTfreqs; //Base Function Frequencies
fft_t *oscilFFTfreqs; //Oscillator Frequencies - this is different than the hamonics set-up by the user, it may contains time-domain data if the antialiasing is turned off
- int oscilprepared; //1 if the oscil is prepared, 0 if it is not prepared and is need to call ::prepare() before ::get()
+ int oscilprepared; //1 if the oscil is prepared, 0 if it is not prepared and is need to call ::prepare() before ::get()
Resonance *res;
unsigned int randseed;
};
-typedef float(*filter_func)(unsigned int, float, float);
+typedef float (*filter_func)(unsigned int, float, float);
filter_func getFilter(unsigned char func);
-typedef float(*base_func)(float,float);
+typedef float (*base_func)(float, float);
base_func getBaseFunction(unsigned char func);
#endif
-
diff --git a/src/Synth/PADnote.cpp b/src/Synth/PADnote.cpp
@@ -30,17 +30,21 @@ PADnote::PADnote(PADnoteParameters *parameters,
int portamento_,
int midinote,
bool besilent)
-:SynthNote(freq, velocity, portamento_, midinote, besilent)
+ :SynthNote(freq, velocity, portamento_, midinote, besilent)
{
pars = parameters;
- ctl = ctl_;
+ ctl = ctl_;
firsttime = true;
setup(freq, velocity, portamento_, midinote);
}
-void PADnote::setup(float freq, float velocity,int portamento_, int midinote, bool legato)
+void PADnote::setup(float freq,
+ float velocity,
+ int portamento_,
+ int midinote,
+ bool legato)
{
portamento = portamento_;
this->velocity = velocity;
@@ -55,7 +59,8 @@ void PADnote::setup(float freq, float velocity,int portamento_, int midinote, bo
if(fixedfreqET != 0) { //if the frequency varies according the keyboard note
float tmp =
(midinote
- - 69.0f) / 12.0f * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);
+ - 69.0f) / 12.0f
+ * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);
if(fixedfreqET <= 64)
basefreq *= powf(2.0f, tmp);
else
@@ -91,13 +96,13 @@ void PADnote::setup(float freq, float velocity,int portamento_, int midinote, bo
size = 1;
- if(!legato) {//not sure
+ if(!legato) { //not sure
poshi_l = (int)(RND * (size - 1));
if(pars->PStereo != 0)
poshi_r = (poshi_l + size / 2) % size;
else
poshi_r = poshi_l;
- poslo = 0.0f;
+ poslo = 0.0f;
}
@@ -115,13 +120,14 @@ void PADnote::setup(float freq, float velocity,int portamento_, int midinote, bo
if(!legato) {
if(pars->PPunchStrength != 0) {
- NoteGlobalPar.Punch.Enabled = 1;
+ NoteGlobalPar.Punch.Enabled = 1;
NoteGlobalPar.Punch.t = 1.0f; //start from 1.0f and to 0.0f
NoteGlobalPar.Punch.initialvalue =
((powf(10, 1.5f * pars->PPunchStrength / 127.0f) - 1.0f)
* VelF(velocity,
- pars->PPunchVelocitySensing));
- float time = powf(10, 3.0f * pars->PPunchTime / 127.0f) / 10000.0f; //0.1f .. 100 ms
+ pars->PPunchVelocitySensing));
+ float time =
+ powf(10, 3.0f * pars->PPunchTime / 127.0f) / 10000.0f; //0.1f .. 100 ms
float stretch = powf(440.0f / freq, pars->PPunchStretch / 64.0f);
NoteGlobalPar.Punch.dt = 1.0f / (time * SAMPLE_RATE * stretch);
}
@@ -131,11 +137,12 @@ void PADnote::setup(float freq, float velocity,int portamento_, int midinote, bo
NoteGlobalPar.FreqEnvelope = new Envelope(pars->FreqEnvelope, basefreq);
NoteGlobalPar.FreqLfo = new LFO(pars->FreqLfo, basefreq);
- NoteGlobalPar.AmpEnvelope = new Envelope(pars->AmpEnvelope, basefreq);
- NoteGlobalPar.AmpLfo = new LFO(pars->AmpLfo, basefreq);
+ NoteGlobalPar.AmpEnvelope = new Envelope(pars->AmpEnvelope, basefreq);
+ NoteGlobalPar.AmpLfo = new LFO(pars->AmpLfo, basefreq);
}
- NoteGlobalPar.Volume = 4.0f * powf(0.1f, 3.0f * (1.0f - pars->PVolume / 96.0f)) //-60 dB .. 0 dB
+ NoteGlobalPar.Volume = 4.0f
+ * powf(0.1f, 3.0f * (1.0f - pars->PVolume / 96.0f)) //-60 dB .. 0 dB
* VelF(velocity, pars->PAmpVelocityScaleFunction); //velocity sensing
NoteGlobalPar.AmpEnvelope->envout_dB(); //discard the first envelope output
@@ -145,12 +152,12 @@ void PADnote::setup(float freq, float velocity,int portamento_, int midinote, bo
* NoteGlobalPar.AmpLfo->amplfoout();
if(!legato) {
- NoteGlobalPar.GlobalFilterL = Filter::generate(pars->GlobalFilter);
- NoteGlobalPar.GlobalFilterR = Filter::generate(pars->GlobalFilter);
+ NoteGlobalPar.GlobalFilterL = Filter::generate(pars->GlobalFilter);
+ NoteGlobalPar.GlobalFilterR = Filter::generate(pars->GlobalFilter);
NoteGlobalPar.FilterEnvelope = new Envelope(pars->FilterEnvelope,
- basefreq);
- NoteGlobalPar.FilterLfo = new LFO(pars->FilterLfo, basefreq);
+ basefreq);
+ NoteGlobalPar.FilterLfo = new LFO(pars->FilterLfo, basefreq);
}
NoteGlobalPar.FilterQ = pars->GlobalFilter->getq();
NoteGlobalPar.FilterFreqTracking = pars->GlobalFilter->getfreqtracking(
@@ -163,10 +170,10 @@ void PADnote::setup(float freq, float velocity,int portamento_, int midinote, bo
}
void PADnote::legatonote(float freq,
- float velocity,
- int portamento_,
- int midinote,
- bool externcall)
+ float velocity,
+ int portamento_,
+ int midinote,
+ bool externcall)
{
// Manage legato stuff
if(legato.update(freq, velocity, portamento_, midinote, externcall))
@@ -215,8 +222,8 @@ void PADnote::computecurrentparameters()
{
float globalpitch, globalfilterpitch;
globalpitch = 0.01f * (NoteGlobalPar.FreqEnvelope->envout()
- + NoteGlobalPar.FreqLfo->lfoout()
- * ctl->modwheel.relmod + NoteGlobalPar.Detune);
+ + NoteGlobalPar.FreqLfo->lfoout()
+ * ctl->modwheel.relmod + NoteGlobalPar.Detune);
globaloldamplitude = globalnewamplitude;
globalnewamplitude = NoteGlobalPar.Volume
* NoteGlobalPar.AmpEnvelope->envout_dB()
@@ -227,7 +234,7 @@ void PADnote::computecurrentparameters()
+ NoteGlobalPar.FilterCenterPitch;
float tmpfilterfreq = globalfilterpitch + ctl->filtercutoff.relfreq
- + NoteGlobalPar.FilterFreqTracking;
+ + NoteGlobalPar.FilterFreqTracking;
tmpfilterfreq = Filter::getrealfreq(tmpfilterfreq);
@@ -289,7 +296,7 @@ int PADnote::Compute_Cubic(float *outl,
finished_ = true;
return 1;
}
- int size = pars->sample[nsample].size;
+ int size = pars->sample[nsample].size;
float xm1, x0, x1, x2, a, b, c;
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
poshi_l += freqhi;
@@ -344,7 +351,7 @@ int PADnote::noteout(float *outl, float *outr)
float freqrap = realfreq / smpfreq;
- int freqhi = (int) (floor(freqrap));
+ int freqhi = (int) (floor(freqrap));
float freqlo = freqrap - floor(freqrap);
@@ -364,10 +371,10 @@ int PADnote::noteout(float *outl, float *outr)
NoteGlobalPar.GlobalFilterR->filterout(outr);
//Apply the punch
- if(NoteGlobalPar.Punch.Enabled != 0) {
+ if(NoteGlobalPar.Punch.Enabled != 0)
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float punchamp = NoteGlobalPar.Punch.initialvalue
- * NoteGlobalPar.Punch.t + 1.0f;
+ * NoteGlobalPar.Punch.t + 1.0f;
outl[i] *= punchamp;
outr[i] *= punchamp;
NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;
@@ -376,29 +383,26 @@ int PADnote::noteout(float *outl, float *outr)
break;
}
}
- }
- if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude)) {
+ if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude))
// Amplitude Interpolation
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,
- globalnewamplitude,
- i,
- SOUND_BUFFER_SIZE);
+ globalnewamplitude,
+ i,
+ SOUND_BUFFER_SIZE);
outl[i] *= tmpvol * NoteGlobalPar.Panning;
outr[i] *= tmpvol * (1.0f - NoteGlobalPar.Panning);
}
- }
- else {
+ else
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;
outr[i] *= globalnewamplitude * (1.0f - NoteGlobalPar.Panning);
}
- }
// Apply legato-specific sound signal modifications
- legato.apply(*this,outl,outr);
+ legato.apply(*this, outl, outr);
// Check if the global amplitude is finished.
// If it does, disable the note
@@ -425,4 +429,3 @@ void PADnote::relasekey()
NoteGlobalPar.FilterEnvelope->relasekey();
NoteGlobalPar.AmpEnvelope->relasekey();
}
-
diff --git a/src/Synth/PADnote.h b/src/Synth/PADnote.h
@@ -30,7 +30,7 @@
#include "../Params/Controller.h"
/**The "pad" synthesizer*/
-class PADnote :public SynthNote
+class PADnote:public SynthNote
{
public:
PADnote(PADnoteParameters *parameters,
@@ -49,18 +49,18 @@ class PADnote :public SynthNote
int finished() const;
void relasekey();
private:
- void setup(float freq, float velocity,int portamento_,
- int midinote, bool legato=false);
+ void setup(float freq, float velocity, int portamento_,
+ int midinote, bool legato = false);
void fadein(float *smps);
void computecurrentparameters();
bool finished_;
PADnoteParameters *pars;
- int poshi_l, poshi_r;
+ int poshi_l, poshi_r;
float poslo;
float basefreq;
- bool firsttime, released;
+ bool firsttime, released;
int nsample, portamento;
@@ -78,7 +78,7 @@ class PADnote :public SynthNote
/******************************************
* FREQUENCY GLOBAL PARAMETERS *
******************************************/
- float Detune; //cents
+ float Detune; //cents
Envelope *FreqEnvelope;
LFO *FreqLfo;
@@ -86,37 +86,36 @@ class PADnote :public SynthNote
/********************************************
* AMPLITUDE GLOBAL PARAMETERS *
********************************************/
- float Volume; // [ 0 .. 1 ]
+ float Volume; // [ 0 .. 1 ]
- float Panning; // [ 0 .. 1 ]
+ float Panning; // [ 0 .. 1 ]
Envelope *AmpEnvelope;
LFO *AmpLfo;
struct {
- int Enabled;
+ int Enabled;
float initialvalue, dt, t;
} Punch;
/******************************************
* FILTER GLOBAL PARAMETERS *
******************************************/
- class Filter *GlobalFilterL, *GlobalFilterR;
+ class Filter *GlobalFilterL, *GlobalFilterR;
- float FilterCenterPitch; //octaves
- float FilterQ;
- float FilterFreqTracking;
+ float FilterCenterPitch; //octaves
+ float FilterQ;
+ float FilterFreqTracking;
Envelope *FilterEnvelope;
- LFO *FilterLfo;
+ LFO *FilterLfo;
} NoteGlobalPar;
- float globaloldamplitude, globalnewamplitude, velocity, realfreq;
+ float globaloldamplitude, globalnewamplitude, velocity, realfreq;
Controller *ctl;
};
#endif
-
diff --git a/src/Synth/Resonance.cpp b/src/Synth/Resonance.cpp
@@ -40,8 +40,8 @@ void Resonance::defaults()
Pcenterfreq = 64; //1 kHz
Poctavesfreq = 64;
Pprotectthefundamental = 0;
- ctlcenter = 1.0f;
- ctlbw = 1.0f;
+ ctlcenter = 1.0f;
+ ctlbw = 1.0f;
for(int i = 0; i < N_RES_POINTS; ++i)
Prespoints[i] = 64;
}
@@ -64,8 +64,8 @@ void Resonance::applyres(int n, fft_t *fftdata, float freq)
if(Penabled == 0)
return; //if the resonance is disabled
float sum = 0.0f,
- l1 = logf(getfreqx(0.0f) * ctlcenter),
- l2 = logf(2.0f) * getoctavesfreq() * ctlbw;
+ l1 = logf(getfreqx(0.0f) * ctlcenter),
+ l2 = logf(2.0f) * getoctavesfreq() * ctlbw;
for(int i = 0; i < N_RES_POINTS; ++i)
if(sum < Prespoints[i])
@@ -80,14 +80,14 @@ void Resonance::applyres(int n, fft_t *fftdata, float freq)
x *= N_RES_POINTS;
float dx = x - floor(x);
- x = floor(x);
- int kx1 = (int)x;
+ x = floor(x);
+ int kx1 = (int)x;
if(kx1 >= N_RES_POINTS)
kx1 = N_RES_POINTS - 1;
- int kx2 = kx1 + 1;
+ int kx2 = kx1 + 1;
if(kx2 >= N_RES_POINTS)
kx2 = N_RES_POINTS - 1;
- float y =
+ float y =
(Prespoints[kx1]
* (1.0f - dx) + Prespoints[kx2] * dx) / 127.0f - sum / 127.0f;
@@ -107,7 +107,7 @@ void Resonance::applyres(int n, fft_t *fftdata, float freq)
float Resonance::getfreqresponse(float freq)
{
float l1 = logf(getfreqx(0.0f) * ctlcenter),
- l2 = logf(2.0f) * getoctavesfreq() * ctlbw, sum = 0.0f;
+ l2 = logf(2.0f) * getoctavesfreq() * ctlbw, sum = 0.0f;
for(int i = 0; i < N_RES_POINTS; ++i)
if(sum < Prespoints[i])
@@ -119,12 +119,12 @@ float Resonance::getfreqresponse(float freq)
if(x < 0.0f)
x = 0.0f;
x *= N_RES_POINTS;
- float dx = x - floor(x);
- x = floor(x);
- int kx1 = (int)x;
+ float dx = x - floor(x);
+ x = floor(x);
+ int kx1 = (int)x;
if(kx1 >= N_RES_POINTS)
kx1 = N_RES_POINTS - 1;
- int kx2 = kx1 + 1;
+ int kx2 = kx1 + 1;
if(kx2 >= N_RES_POINTS)
kx2 = N_RES_POINTS - 1;
float result =
@@ -178,7 +178,7 @@ void Resonance::randomize(int type)
void Resonance::interpolatepeaks(int type)
{
int x1 = 0, y1 = Prespoints[0];
- for(int i = 1; i < N_RES_POINTS; ++i) {
+ for(int i = 1; i < N_RES_POINTS; ++i)
if((Prespoints[i] != 64) || (i + 1 == N_RES_POINTS)) {
int y2 = Prespoints[i];
for(int k = 0; k < i - x1; ++k) {
@@ -190,7 +190,6 @@ void Resonance::interpolatepeaks(int type)
x1 = i;
y1 = y2;
}
- }
}
/*
@@ -261,7 +260,7 @@ void Resonance::add2XML(XMLwrapper *xml)
void Resonance::getfromXML(XMLwrapper *xml)
{
- Penabled = xml->getparbool("enabled", Penabled);
+ Penabled = xml->getparbool("enabled", Penabled);
PmaxdB = xml->getpar127("max_db", PmaxdB);
Pcenterfreq = xml->getpar127("center_freq", Pcenterfreq);
@@ -275,4 +274,3 @@ void Resonance::getfromXML(XMLwrapper *xml)
xml->exitbranch();
}
}
-
diff --git a/src/Synth/Resonance.h b/src/Synth/Resonance.h
@@ -68,4 +68,3 @@ class Resonance:public Presets
};
#endif
-
diff --git a/src/Synth/SUBnote.cpp b/src/Synth/SUBnote.cpp
@@ -35,15 +35,19 @@ SUBnote::SUBnote(SUBnoteParameters *parameters,
int portamento_,
int midinote,
bool besilent)
-:SynthNote(freq, velocity, portamento_, midinote, besilent)
+ :SynthNote(freq, velocity, portamento_, midinote, besilent)
{
pars = parameters;
ctl = ctl_;
- NoteEnabled= ON;
+ NoteEnabled = ON;
setup(freq, velocity, portamento_, midinote);
}
-void SUBnote::setup(float freq, float velocity, int portamento_, int midinote, bool legato)
+void SUBnote::setup(float freq,
+ float velocity,
+ int portamento_,
+ int midinote,
+ bool legato)
{
portamento = portamento_;
NoteEnabled = ON;
@@ -69,7 +73,8 @@ void SUBnote::setup(float freq, float velocity, int portamento_, int midinote, b
if(fixedfreqET != 0) { //if the frequency varies according the keyboard note
float tmp =
(midinote
- - 69.0f) / 12.0f * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);
+ - 69.0f) / 12.0f
+ * (powf(2.0f, (fixedfreqET - 1) / 63.0f) - 1.0f);
if(fixedfreqET <= 64)
basefreq *= powf(2.0f, tmp);
else
@@ -77,14 +82,15 @@ void SUBnote::setup(float freq, float velocity, int portamento_, int midinote, b
}
}
float detune = getdetune(pars->PDetuneType,
- pars->PCoarseDetune,
- pars->PDetune);
+ pars->PCoarseDetune,
+ pars->PDetune);
basefreq *= powf(2.0f, detune / 1200.0f); //detune
// basefreq*=ctl->pitchwheel.relfreq;//pitch wheel
//global filter
GlobalFilterCenterPitch = pars->GlobalFilter->getfreq() //center freq
- + (pars->PGlobalFilterVelocityScale / 127.0f * 6.0f) //velocity sensing
+ + (pars->PGlobalFilterVelocityScale / 127.0f
+ * 6.0f) //velocity sensing
* (VelF(velocity,
pars->PGlobalFilterVelocityScaleFunction)
- 1);
@@ -146,26 +152,26 @@ void SUBnote::setup(float freq, float velocity, int portamento_, int midinote, b
bw = 25.0f;
//try to keep same amplitude on all freqs and bw. (empirically)
- float gain = sqrt(1500.0f / (bw * freq));
+ float gain = sqrt(1500.0f / (bw * freq));
float hmagnew = 1.0f - pars->Phmag[pos[n]] / 127.0f;
float hgain;
switch(pars->Phmagtype) {
- case 1:
- hgain = expf(hmagnew * logf(0.01f));
- break;
- case 2:
- hgain = expf(hmagnew * logf(0.001f));
- break;
- case 3:
- hgain = expf(hmagnew * logf(0.0001f));
- break;
- case 4:
- hgain = expf(hmagnew * logf(0.00001f));
- break;
- default:
- hgain = 1.0f - hmagnew;
+ case 1:
+ hgain = expf(hmagnew * logf(0.01f));
+ break;
+ case 2:
+ hgain = expf(hmagnew * logf(0.001f));
+ break;
+ case 3:
+ hgain = expf(hmagnew * logf(0.0001f));
+ break;
+ case 4:
+ hgain = expf(hmagnew * logf(0.00001f));
+ break;
+ default:
+ hgain = 1.0f - hmagnew;
}
gain *= hgain;
reduceamp += hgain;
@@ -200,7 +206,8 @@ void SUBnote::setup(float freq, float velocity, int portamento_, int midinote, b
if(pars->PGlobalFilterEnabled != 0) {
globalfiltercenterq = pars->GlobalFilter->getq();
- GlobalFilterFreqTracking = pars->GlobalFilter->getfreqtracking(basefreq);
+ GlobalFilterFreqTracking = pars->GlobalFilter->getfreqtracking(
+ basefreq);
}
}
@@ -330,14 +337,14 @@ void SUBnote::filter(bpfilter &filter, float *smps)
{
assert(SOUND_BUFFER_SIZE % 8 == 0);
for(int i = 0; i < SOUND_BUFFER_SIZE; i += 8) {
- smps[i] = SubFilter(filter, smps[i]);
- smps[i+1] = SubFilter(filter, smps[i+1]);
- smps[i+2] = SubFilter(filter, smps[i+2]);
- smps[i+3] = SubFilter(filter, smps[i+3]);
- smps[i+4] = SubFilter(filter, smps[i+4]);
- smps[i+5] = SubFilter(filter, smps[i+5]);
- smps[i+6] = SubFilter(filter, smps[i+6]);
- smps[i+7] = SubFilter(filter, smps[i+7]);
+ smps[i] = SubFilter(filter, smps[i]);
+ smps[i + 1] = SubFilter(filter, smps[i + 1]);
+ smps[i + 2] = SubFilter(filter, smps[i + 2]);
+ smps[i + 3] = SubFilter(filter, smps[i + 3]);
+ smps[i + 4] = SubFilter(filter, smps[i + 4]);
+ smps[i + 5] = SubFilter(filter, smps[i + 5]);
+ smps[i + 6] = SubFilter(filter, smps[i + 6]);
+ smps[i + 7] = SubFilter(filter, smps[i + 7]);
}
}
@@ -360,8 +367,8 @@ void SUBnote::initparameters(float freq)
GlobalFilterL = Filter::generate(pars->GlobalFilter);
if(stereo)
GlobalFilterR = Filter::generate(pars->GlobalFilter);
- GlobalFilterEnvelope = new Envelope(pars->GlobalFilterEnvelope,
- freq);
+ GlobalFilterEnvelope = new Envelope(pars->GlobalFilterEnvelope,
+ freq);
GlobalFilterFreqTracking = pars->GlobalFilter->getfreqtracking(basefreq);
}
computecurrentparameters();
@@ -401,7 +408,7 @@ void SUBnote::computecurrentparameters()
float tmpgain = 1.0f / sqrt(envbw * envfreq);
- for(int n = 0; n < numharmonics; ++n) {
+ for(int n = 0; n < numharmonics; ++n)
for(int nph = 0; nph < numstages; ++nph) {
if(nph == 0)
gain = tmpgain;
@@ -412,9 +419,8 @@ void SUBnote::computecurrentparameters()
lfilter[nph + n * numstages].bw * envbw,
gain);
}
- }
if(stereo != 0)
- for(int n = 0; n < numharmonics; ++n) {
+ for(int n = 0; n < numharmonics; ++n)
for(int nph = 0; nph < numstages; ++nph) {
if(nph == 0)
gain = tmpgain;
@@ -427,7 +433,7 @@ void SUBnote::computecurrentparameters()
rfilter[nph + n * numstages].bw * envbw,
gain);
}
- }
+
oldbandwidth = ctl->bandwidth.data;
oldpitchwheel = ctl->pitchwheel.data;
@@ -437,9 +443,9 @@ void SUBnote::computecurrentparameters()
//Filter
if(GlobalFilterL != NULL) {
float globalfilterpitch = GlobalFilterCenterPitch
- + GlobalFilterEnvelope->envout();
+ + GlobalFilterEnvelope->envout();
float filterfreq = globalfilterpitch + ctl->filtercutoff.relfreq
- + GlobalFilterFreqTracking;
+ + GlobalFilterFreqTracking;
filterfreq = Filter::getrealfreq(filterfreq);
GlobalFilterL->setfreq_and_q(filterfreq,
@@ -511,29 +517,27 @@ int SUBnote::noteout(float *outl, float *outr)
firsttick = 0;
}
- if(ABOVE_AMPLITUDE_THRESHOLD(oldamplitude, newamplitude)) {
+ if(ABOVE_AMPLITUDE_THRESHOLD(oldamplitude, newamplitude))
// Amplitude interpolation
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmpvol = INTERPOLATE_AMPLITUDE(oldamplitude,
- newamplitude,
- i,
- SOUND_BUFFER_SIZE);
+ newamplitude,
+ i,
+ SOUND_BUFFER_SIZE);
outl[i] *= tmpvol * panning;
outr[i] *= tmpvol * (1.0f - panning);
}
- }
- else {
+ else
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] *= newamplitude * panning;
outr[i] *= newamplitude * (1.0f - panning);
}
- }
oldamplitude = newamplitude;
computecurrentparameters();
// Apply legato-specific sound signal modifications
- legato.apply(*this,outl,outr);
+ legato.apply(*this, outl, outr);
// Check if the note needs to be computed more
if(AmpEnvelope->finished() != 0) {
@@ -571,4 +575,3 @@ int SUBnote::finished() const
else
return 0;
}
-
diff --git a/src/Synth/SUBnote.h b/src/Synth/SUBnote.h
@@ -30,7 +30,7 @@
#include "Envelope.h"
#include "../DSP/Filter.h"
-class SUBnote :public SynthNote
+class SUBnote:public SynthNote
{
public:
SUBnote(SUBnoteParameters *parameters, Controller *ctl_, float freq,
@@ -45,7 +45,11 @@ class SUBnote :public SynthNote
int finished() const;
private:
- void setup(float freq, float velocity, int portamento_, int midinote, bool legato=false);
+ void setup(float freq,
+ float velocity,
+ int portamento_,
+ int midinote,
+ bool legato = false);
void computecurrentparameters();
void initparameters(float freq);
void KillNote();
@@ -53,13 +57,13 @@ class SUBnote :public SynthNote
SUBnoteParameters *pars;
//parameters
- int stereo;
- int numstages; //number of stages of filters
- int numharmonics; //number of harmonics (after the too higher hamonics are removed)
- int firstnumharmonics; //To keep track of the first note's numharmonics value, useful in legato mode.
- int start; //how the harmonics start
- float basefreq;
- float panning;
+ int stereo;
+ int numstages; //number of stages of filters
+ int numharmonics; //number of harmonics (after the too higher hamonics are removed)
+ int firstnumharmonics; //To keep track of the first note's numharmonics value, useful in legato mode.
+ int start; //how the harmonics start
+ float basefreq;
+ float panning;
Envelope *AmpEnvelope;
Envelope *FreqEnvelope;
Envelope *BandWidthEnvelope;
@@ -70,8 +74,8 @@ class SUBnote :public SynthNote
//internal values
ONOFFTYPE NoteEnabled;
- int firsttick, portamento;
- float volume, oldamplitude, newamplitude;
+ int firsttick, portamento;
+ float volume, oldamplitude, newamplitude;
float GlobalFilterCenterPitch; //octaves
float GlobalFilterFreqTracking;
@@ -97,9 +101,8 @@ class SUBnote :public SynthNote
bpfilter *lfilter, *rfilter;
Controller *ctl;
- int oldpitchwheel, oldbandwidth;
+ int oldpitchwheel, oldbandwidth;
float globalfiltercenterq;
};
#endif
-
diff --git a/src/Synth/SynthNote.cpp b/src/Synth/SynthNote.cpp
@@ -3,7 +3,7 @@
#include <cstring>
SynthNote::SynthNote(float freq, float vel, int port, int note, bool quiet)
- :legato(freq,vel,port,note,quiet)
+ :legato(freq, vel, port, note, quiet)
{}
SynthNote::Legato::Legato(float freq, float vel, int port,
@@ -11,13 +11,13 @@ SynthNote::Legato::Legato(float freq, float vel, int port,
{
// Initialise some legato-specific vars
msg = LM_Norm;
- fade.length = (int)(SAMPLE_RATE * 0.005f); // 0.005f seems ok.
+ fade.length = (int)(SAMPLE_RATE * 0.005f); // 0.005f seems ok.
if(fade.length < 1)
fade.length = 1; // (if something's fishy)
- fade.step = (1.0f / fade.length);
- decounter = -10;
- param.freq = freq;
- param.vel = vel;
+ fade.step = (1.0f / fade.length);
+ decounter = -10;
+ param.freq = freq;
+ param.vel = vel;
param.portamento = port;
param.midinote = note;
lastfreq = 0.0f;
@@ -25,7 +25,7 @@ SynthNote::Legato::Legato(float freq, float vel, int port,
}
int SynthNote::Legato::update(float freq, float velocity, int portamento_,
- int midinote_, bool externcall)
+ int midinote_, bool externcall)
{
if(externcall)
msg = LM_Norm;
@@ -60,69 +60,68 @@ void SynthNote::Legato::apply(SynthNote ¬e, float *outl, float *outr)
memset(outr, 0, SOUND_BUFFER_SIZE * sizeof(float));
}
switch(msg) {
- case LM_CatchUp: // Continue the catch-up...
- if(decounter == -10)
- decounter = fade.length;
- //Yea, could be done without the loop...
- for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- decounter--;
- if(decounter < 1) {
- // Catching-up done, we can finally set
- // the note to the actual parameters.
- decounter = -10;
- msg = LM_ToNorm;
- note.legatonote(param.freq, param.vel, param.portamento,
- param.midinote, false);
- break;
- }
- }
- break;
- case LM_FadeIn: // Fade-in
- if(decounter == -10)
- decounter = fade.length;
- silent = false;
- for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- decounter--;
- if(decounter < 1) {
- decounter = -10;
- msg = LM_Norm;
- break;
+ case LM_CatchUp: // Continue the catch-up...
+ if(decounter == -10)
+ decounter = fade.length;
+ //Yea, could be done without the loop...
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
+ decounter--;
+ if(decounter < 1) {
+ // Catching-up done, we can finally set
+ // the note to the actual parameters.
+ decounter = -10;
+ msg = LM_ToNorm;
+ note.legatonote(param.freq, param.vel, param.portamento,
+ param.midinote, false);
+ break;
+ }
}
- fade.m += fade.step;
- outl[i] *= fade.m;
- outr[i] *= fade.m;
- }
- break;
- case LM_FadeOut: // Fade-out, then set the catch-up
- if(decounter == -10)
- decounter = fade.length;
- for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
- decounter--;
- if(decounter < 1) {
- for(int j = i; j < SOUND_BUFFER_SIZE; ++j) {
- outl[j] = 0.0f;
- outr[j] = 0.0f;
+ break;
+ case LM_FadeIn: // Fade-in
+ if(decounter == -10)
+ decounter = fade.length;
+ silent = false;
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
+ decounter--;
+ if(decounter < 1) {
+ decounter = -10;
+ msg = LM_Norm;
+ break;
}
- decounter = -10;
- silent = true;
- // Fading-out done, now set the catch-up :
+ fade.m += fade.step;
+ outl[i] *= fade.m;
+ outr[i] *= fade.m;
+ }
+ break;
+ case LM_FadeOut: // Fade-out, then set the catch-up
+ if(decounter == -10)
decounter = fade.length;
- msg = LM_CatchUp;
- //This freq should make this now silent note to catch-up/resync
- //with the heard note for the same length it stayed at the
- //previous freq during the fadeout.
- float catchupfreq = param.freq * (param.freq / lastfreq);
- note.legatonote(catchupfreq, param.vel, param.portamento,
- param.midinote, false);
- break;
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
+ decounter--;
+ if(decounter < 1) {
+ for(int j = i; j < SOUND_BUFFER_SIZE; ++j) {
+ outl[j] = 0.0f;
+ outr[j] = 0.0f;
+ }
+ decounter = -10;
+ silent = true;
+ // Fading-out done, now set the catch-up :
+ decounter = fade.length;
+ msg = LM_CatchUp;
+ //This freq should make this now silent note to catch-up/resync
+ //with the heard note for the same length it stayed at the
+ //previous freq during the fadeout.
+ float catchupfreq = param.freq * (param.freq / lastfreq);
+ note.legatonote(catchupfreq, param.vel, param.portamento,
+ param.midinote, false);
+ break;
+ }
+ fade.m -= fade.step;
+ outl[i] *= fade.m;
+ outr[i] *= fade.m;
}
- fade.m -= fade.step;
- outl[i] *= fade.m;
- outr[i] *= fade.m;
- }
- break;
- default:
- break;
+ break;
+ default:
+ break;
}
}
-
diff --git a/src/Synth/SynthNote.h b/src/Synth/SynthNote.h
@@ -43,33 +43,34 @@ class SynthNote
virtual int finished() const = 0;
virtual void legatonote(float freq, float velocity,
- int portamento_, int midinote_, bool externcall) = 0;
+ int portamento_, int midinote_,
+ bool externcall) = 0;
protected:
// Legato transitions
- class Legato{
+ class Legato
+ {
public:
Legato(float freq, float vel, int port,
- int note, bool quiet);
+ int note, bool quiet);
void apply(SynthNote ¬e, float *outl, float *outr);
int update(float freq, float velocity, int portamento_,
- int midinote_, bool externalcall);
+ int midinote_, bool externalcall);
private:
bool silent;
- float lastfreq;
+ float lastfreq;
LegatoMsg msg;
- int decounter;
+ int decounter;
struct { // Fade In/Out vars
- int length;
+ int length;
float m, step;
} fade;
struct { // Note parameters
float freq, vel;
- int portamento, midinote;
+ int portamento, midinote;
} param;
} legato;
};
#endif
-
diff --git a/src/Tests/AdNoteTest.h b/src/Tests/AdNoteTest.h
@@ -15,9 +15,9 @@ class AdNoteTest:public CxxTest::TestSuite
{
public:
- ADnote *note;
- Master *master;
- FFTwrapper *fft;
+ ADnote *note;
+ Master *master;
+ FFTwrapper *fft;
Controller *controller;
unsigned char testnote;
@@ -170,4 +170,3 @@ class AdNoteTest:public CxxTest::TestSuite
}
#endif
};
-
diff --git a/src/Tests/ControllerTest.h b/src/Tests/ControllerTest.h
@@ -69,4 +69,3 @@ class ControllerTest:public CxxTest::TestSuite
private:
Controller *testCtl;
};
-
diff --git a/src/Tests/EchoTest.h b/src/Tests/EchoTest.h
@@ -38,7 +38,8 @@ class EchoTest:public CxxTest::TestSuite
outR = new float[SOUND_BUFFER_SIZE];
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
outR[i] = 0.0f;
- input = new Stereo<float *>(new float[SOUND_BUFFER_SIZE],new float[SOUND_BUFFER_SIZE]);
+ input = new Stereo<float *>(new float[SOUND_BUFFER_SIZE],
+ new float[SOUND_BUFFER_SIZE]);
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
input->l[i] = input->r[i] = 0.0f;
testFX = new Echo(true, outL, outR);
@@ -91,7 +92,6 @@ class EchoTest:public CxxTest::TestSuite
}
//Insures that the proper decay occurs with high feedback
void testDecaywFb() {
-
//flood with high input
for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
input->r[i] = input->l[i] = 1.0f;
@@ -120,4 +120,3 @@ class EchoTest:public CxxTest::TestSuite
float *outR, *outL;
Echo *testFX;
};
-
diff --git a/src/Tests/MicrotonalTest.h b/src/Tests/MicrotonalTest.h
@@ -87,7 +87,7 @@ class MicrotonalTest:public CxxTest::TestSuite
xml.endbranch();
xml.endbranch();
- char *tmp = xml.getXMLdata();
+ char *tmp = xml.getXMLdata();
Microtonal other;
other.Penabled = 1;
@@ -129,7 +129,7 @@ class MicrotonalTest:public CxxTest::TestSuite
//the tuning is from old documentation for "Intense Diatonic" scale
const char *tuning[7] =
{"9/8", "5/4", "4/3", "3/2", "5/3", "15/8", "2/1"};
- const int numTunings = 7;
+ const int numTunings = 7;
//for(int i=0;i<20;++i)
// cout << i << ':' << testMicro->getnotefreq(i,0) << endl;
// go to middle key and verify the proportions
@@ -140,4 +140,3 @@ class MicrotonalTest:public CxxTest::TestSuite
private:
Microtonal *testMicro;
};
-
diff --git a/src/Tests/OscilGenTest.h b/src/Tests/OscilGenTest.h
@@ -7,15 +7,15 @@ using namespace std;
class OscilGenTest:public CxxTest::TestSuite
{
public:
- float freq;
+ float freq;
float *outR, *outL;
FFTwrapper *fft;
- OscilGen *oscil;
+ OscilGen *oscil;
void setUp() {
//First the sensible settings and variables that have to be set:
SOUND_BUFFER_SIZE = 256;
- OSCIL_SIZE = 1024;
+ OSCIL_SIZE = 1024;
outL = new float[OSCIL_SIZE];
outR = new float[OSCIL_SIZE];
@@ -55,7 +55,7 @@ class OscilGenTest:public CxxTest::TestSuite
const char testnote = 50;
freq = 440.0f * powf(2.0f, (testnote - 69.0f) / 12.0f);
}
-
+
void tearDown() {
delete oscil;
delete fft;
@@ -74,7 +74,7 @@ class OscilGenTest:public CxxTest::TestSuite
void testOutput(void)
{
oscil->get(outL, freq);
- TS_ASSERT_DELTA(outL[23], -0.044547f, 0.0001f);
+ TS_ASSERT_DELTA(outL[23], -0.044547f, 0.0001f);
TS_ASSERT_DELTA(outL[129], -0.018169f, 0.0001f);
TS_ASSERT_DELTA(outL[586], 0.045647f, 0.0001f);
TS_ASSERT_DELTA(outL[1023], -0.038334f, 0.0001f);
@@ -104,7 +104,7 @@ class OscilGenTest:public CxxTest::TestSuite
printf("OscilGenTest: %f seconds for %d prepares.\n",
(static_cast<float>(t_off - t_on)) / CLOCKS_PER_SEC, samps);
-
+
t_on = clock(); // timer before calling func
for(int i = 0; i < samps; ++i)
oscil->get(outL, freq);
diff --git a/src/Tests/RandTest.h b/src/Tests/RandTest.h
@@ -38,5 +38,3 @@ class RandTest:public CxxTest::TestSuite
TS_ASSERT_DELTA(RND, 0.511766, 0.00001);
}
};
-
-
diff --git a/src/Tests/SampleTest.h b/src/Tests/SampleTest.h
@@ -41,7 +41,7 @@ class SampleTest:public CxxTest::TestSuite
smp[1] = 1;
smp[2] = 2;
Sample nsmp(40);
- nsmp = smp;
+ nsmp = smp;
TS_ASSERT_EQUALS(smp.size(), nsmp.size());
for(int i = 0; i < 29; ++i)
TS_ASSERT_EQUALS(smp[i], nsmp[i]);
@@ -82,21 +82,20 @@ class SampleTest:public CxxTest::TestSuite
}
void testResample() {
- Sample orig(32,2);
+ Sample orig(32, 2);
Sample cpy(orig);
//test for no resampleing
- orig.resample(128,128);
- TS_ASSERT_EQUALS(cpy,orig);
+ orig.resample(128, 128);
+ TS_ASSERT_EQUALS(cpy, orig);
//test for no bad distortions
- orig.resample(128,256);
- orig.resample(256,128);
- TS_ASSERT_EQUALS(cpy,orig);
+ orig.resample(128, 256);
+ orig.resample(256, 128);
+ TS_ASSERT_EQUALS(cpy, orig);
//test for downsample
- orig.resample(256,128);
- TS_ASSERT_EQUALS(orig.size(),cpy.size()/2);
+ orig.resample(256, 128);
+ TS_ASSERT_EQUALS(orig.size(), cpy.size() / 2);
}
};
-
diff --git a/src/Tests/SubNoteTest.h b/src/Tests/SubNoteTest.h
@@ -156,4 +156,3 @@ class SubNoteTest:public CxxTest::TestSuite
}
#endif
};
-
diff --git a/src/Tests/XMLwrapperTest.h b/src/Tests/XMLwrapperTest.h
@@ -40,13 +40,15 @@ class XMLwrapperTest:public CxxTest::TestSuite
//here to verify that no leaks occur
void testLoad() {
- string location = string(SOURCE_DIR) + string("/Tests/guitar-adnote.xmz");
+ string location = string(SOURCE_DIR) + string(
+ "/Tests/guitar-adnote.xmz");
xmla->loadXMLfile(location);
}
void testAnotherLoad()
{
- string dat = "\n<?xml version=\"1.0f\" encoding=\"UTF-8\"?>\n\
+ string dat =
+ "\n<?xml version=\"1.0f\" encoding=\"UTF-8\"?>\n\
<!DOCTYPE ZynAddSubFX-data>\n\
<ZynAddSubFX-data version-major=\"2\" version-minor=\"4\"\n\
version-revision=\"1\" ZynAddSubFX-author=\"Nasca Octavian Paul\">\n\
@@ -64,4 +66,3 @@ version-revision=\"1\" ZynAddSubFX-author=\"Nasca Octavian Paul\">\n\
XMLwrapper *xmla;
XMLwrapper *xmlb;
};
-
diff --git a/src/UI/NioUI.cpp b/src/UI/NioUI.cpp
@@ -14,25 +14,25 @@
using namespace std;
NioUI::NioUI()
- :Fl_Window(200,100,400,400,"New IO Controls")
+ :Fl_Window(200, 100, 400, 400, "New IO Controls")
{
//hm, I appear to be leaking memory
- Fl_Tabs *wintabs = new Fl_Tabs(0,0,400,400-15);
+ Fl_Tabs *wintabs = new Fl_Tabs(0, 0, 400, 400 - 15);
{
- Fl_Group *gen = new Fl_Group(0,20,400,400-35,"General");
+ Fl_Group *gen = new Fl_Group(0, 20, 400, 400 - 35, "General");
{
- Fl_Text_Buffer *buff = new Fl_Text_Buffer();
- Fl_Text_Display *intro = new Fl_Text_Display(20,40,350,300);
+ Fl_Text_Buffer *buff = new Fl_Text_Buffer();
+ Fl_Text_Display *intro = new Fl_Text_Display(20, 40, 350, 300);
intro->buffer(buff);
buff->text("Thanks For Testing Out the New"
- " Input/Output system. "
- "Beware of bugs that may exist and"
- " enjoy the new system.");
+ " Input/Output system. "
+ "Beware of bugs that may exist and"
+ " enjoy the new system.");
intro->wrap_mode(4, 40);
}
gen->end();
- Fl_Group *settings = new Fl_Group(0,20,400,400-35,"Settings");
+ Fl_Group *settings = new Fl_Group(0, 20, 400, 400 - 35, "Settings");
{
audio = new Fl_Choice(60, 80, 100, 25, "Audio");
audio->callback(audioCallback);
@@ -46,10 +46,10 @@ NioUI::NioUI()
//initialize midi list
{
set<string> midiList = nio.getSources();
- string source = nio.getSource();
+ string source = nio.getSource();
int midival = 0;
for(set<string>::iterator itr = midiList.begin();
- itr != midiList.end(); ++itr) {
+ itr != midiList.end(); ++itr) {
midi->add(itr->c_str());
if(*itr == source)
midival = midi->size() - 2;
@@ -60,10 +60,10 @@ NioUI::NioUI()
//initialize audio list
{
set<string> audioList = nio.getSinks();
- string sink = nio.getInstance().getSink();
+ string sink = nio.getInstance().getSink();
int audioval = 0;
for(set<string>::iterator itr = audioList.begin();
- itr != audioList.end(); ++itr) {
+ itr != audioList.end(); ++itr) {
audio->add(itr->c_str());
if(*itr == sink)
audioval = audio->size() - 2;
@@ -74,26 +74,23 @@ NioUI::NioUI()
wintabs->end();
resizable(this);
- size_range(400,300);
+ size_range(400, 300);
}
NioUI::~NioUI()
-{
-}
+{}
void NioUI::refresh()
-{
-}
+{}
void NioUI::midiCallback(Fl_Widget *c)
{
bool good = Nio::getInstance().setSource(static_cast<Fl_Choice *>(c)->text());
- static_cast<Fl_Choice *>(c)->textcolor(fl_rgb_color(255*!good,0,0));
+ static_cast<Fl_Choice *>(c)->textcolor(fl_rgb_color(255 * !good, 0, 0));
}
void NioUI::audioCallback(Fl_Widget *c)
{
bool good = Nio::getInstance().setSink(static_cast<Fl_Choice *>(c)->text());
- static_cast<Fl_Choice *>(c)->textcolor(fl_rgb_color(255*!good,0,0));
+ static_cast<Fl_Choice *>(c)->textcolor(fl_rgb_color(255 * !good, 0, 0));
}
-
diff --git a/src/UI/NioUI.h b/src/UI/NioUI.h
@@ -4,18 +4,17 @@
#include <FL/Fl.H>
#include <FL/Fl_Window.H>
-class NioUI : public Fl_Window
+class NioUI:public Fl_Window
{
public:
NioUI();
~NioUI();
void refresh();
private:
- class Fl_Choice *midi;
- class Fl_Choice *audio;
+ class Fl_Choice * midi;
+ class Fl_Choice * audio;
static void midiCallback(Fl_Widget *c);
static void audioCallback(Fl_Widget *c);
};
#endif
-
diff --git a/src/UI/WidgetPDial.cpp b/src/UI/WidgetPDial.cpp
@@ -15,22 +15,23 @@
using namespace std;
-class TipWin : public Fl_Menu_Window {
+class TipWin:public Fl_Menu_Window
+{
public:
TipWin();
void draw();
void showValue(float f);
- void setText(const char * c);
+ void setText(const char *c);
void showText();
private:
void redraw();
const char *getStr() const;
string tip;
string text;
- bool textmode;
+ bool textmode;
};
-TipWin::TipWin():Fl_Menu_Window(1,1)
+TipWin::TipWin():Fl_Menu_Window(1, 1)
{
set_override();
end();
@@ -44,7 +45,8 @@ void TipWin::draw()
fl_font(labelfont(), labelsize());
//Draw the current string
- fl_draw(getStr(), 3, 3, w()-6, h()-6, Fl_Align(FL_ALIGN_LEFT|FL_ALIGN_WRAP));
+ fl_draw(getStr(), 3, 3, w() - 6, h() - 6,
+ Fl_Align(FL_ALIGN_LEFT | FL_ALIGN_WRAP));
}
void TipWin::showValue(float f)
@@ -54,22 +56,22 @@ void TipWin::showValue(float f)
snprintf(tmp, 9, "%.2f", f);
tip = tmp;
- textmode=false;
+ textmode = false;
redraw();
show();
}
-void TipWin::setText(const char * c)
+void TipWin::setText(const char *c)
{
- text = c;
- textmode=true;
+ text = c;
+ textmode = true;
redraw();
}
void TipWin::showText()
{
if(!text.empty()) {
- textmode=true;
+ textmode = true;
redraw();
show();
}
@@ -95,8 +97,8 @@ const char *TipWin::getStr() const
//static int numobj = 0;
-WidgetPDial::WidgetPDial(int x,int y, int w, int h, const char *label)
- :Fl_Dial(x,y,w,h,label),oldvalue(0.0f),pos(false),textset(false)
+WidgetPDial::WidgetPDial(int x, int y, int w, int h, const char *label)
+ :Fl_Dial(x, y, w, h, label), oldvalue(0.0f), pos(false), textset(false)
{
//cout << "[" << label << "] There are now " << ++numobj << endl;
Fl_Group *save = Fl_Group::current();
@@ -113,24 +115,25 @@ WidgetPDial::~WidgetPDial()
int WidgetPDial::handle(int event)
{
- double dragsize, min=minimum(),max=maximum();
- int my;
+ double dragsize, min = minimum(), max = maximum();
+ int my;
- switch (event){
+ switch(event) {
case FL_PUSH:
- oldvalue=value();
+ oldvalue = value();
case FL_DRAG:
getPos();
tipwin->showValue(value());
- my=-(Fl::event_y()-y()-h()/2);
+ my = -(Fl::event_y() - y() - h() / 2);
- dragsize=200.0f;
- if (Fl::event_state(FL_BUTTON1)==0)
- dragsize*=10;
+ dragsize = 200.0f;
+ if(Fl::event_state(FL_BUTTON1) == 0)
+ dragsize *= 10;
- value(limit(oldvalue+my/dragsize*(max-min),min,max));
+ value(limit(oldvalue + my / dragsize * (max - min), min, max));
value_damage();
- if (this->when()!=0) do_callback();
+ if(this->when() != 0)
+ do_callback();
return 1;
case FL_ENTER:
getPos();
@@ -144,7 +147,7 @@ int WidgetPDial::handle(int event)
case FL_RELEASE:
tipwin->hide();
resetPos();
- if (this->when()==0)
+ if(this->when() == 0)
do_callback();
return 1;
break;
@@ -152,87 +155,92 @@ int WidgetPDial::handle(int event)
return 0;
}
-void WidgetPDial::drawgradient(int cx,int cy,int sx,double m1,double m2)
+void WidgetPDial::drawgradient(int cx, int cy, int sx, double m1, double m2)
{
- for (int i=(int)(m1*sx);i<(int)(m2*sx);i++){
- double tmp=1.0f-powf(i*1.0f/sx,2.0f);
- pdialcolor(140+(int) (tmp*90),140+(int)(tmp*90),140+(int) (tmp*100));
- fl_arc(cx+sx/2-i/2,cy+sx/2-i/2,i,i,0,360);
+ for(int i = (int)(m1 * sx); i < (int)(m2 * sx); i++) {
+ double tmp = 1.0f - powf(i * 1.0f / sx, 2.0f);
+ pdialcolor(140
+ + (int) (tmp
+ * 90), 140
+ + (int)(tmp * 90), 140 + (int) (tmp * 100));
+ fl_arc(cx + sx / 2 - i / 2, cy + sx / 2 - i / 2, i, i, 0, 360);
}
}
void WidgetPDial::draw()
{
- int cx=x(),cy=y(),sx=w(),sy=h();
+ int cx = x(), cy = y(), sx = w(), sy = h();
//clears the button face
- pdialcolor(190,190,200);
- fl_pie(cx-1,cy-1,sx+2,sy+2,0,360);
+ pdialcolor(190, 190, 200);
+ fl_pie(cx - 1, cy - 1, sx + 2, sy + 2, 0, 360);
//Draws the button face (gradinet)
- drawgradient(cx,cy,sx,0.5f,1.0f);
+ drawgradient(cx, cy, sx, 0.5f, 1.0f);
- double val=(value()-minimum())/(maximum()-minimum());
+ double val = (value() - minimum()) / (maximum() - minimum());
//draws the scale
- pdialcolor(220,220,250);
- double a1=angle1(),a2=angle2();
- for (int i=0;i<12;i++){
- double a=-i/12.0f*360.0f-val*(a2-a1)-a1;
- fl_pie(cx,cy,sx,sy,a+270-3,a+3+270);
- };
+ pdialcolor(220, 220, 250);
+ double a1 = angle1(), a2 = angle2();
+ for(int i = 0; i < 12; i++) {
+ double a = -i / 12.0f * 360.0f - val * (a2 - a1) - a1;
+ fl_pie(cx, cy, sx, sy, a + 270 - 3, a + 3 + 270);
+ }
- drawgradient(cx,cy,sx,0.0f,0.75f);
+ drawgradient(cx, cy, sx, 0.0f, 0.75f);
//draws the value
- double a=-(a2-a1)*val-a1;
+ double a = -(a2 - a1) * val - a1;
//draws the max and min points
- pdialcolor(0,100,200);
- int xp=(int)(cx+sx/2.0f+sx/2.0f*sinf(angle1()/180.0f*3.141592f));
- int yp=(int)(cy+sy/2.0f+sy/2.0f*cosf(angle1()/180.0f*3.141592f));
- fl_pie(xp-2,yp-2,4,4,0,360);
+ pdialcolor(0, 100, 200);
+ int xp =
+ (int)(cx + sx / 2.0f + sx / 2.0f * sinf(angle1() / 180.0f * 3.141592f));
+ int yp =
+ (int)(cy + sy / 2.0f + sy / 2.0f * cosf(angle1() / 180.0f * 3.141592f));
+ fl_pie(xp - 2, yp - 2, 4, 4, 0, 360);
- xp=(int)(cx+sx/2.0f+sx/2.0f*sinf(angle2()/180.0f*3.141592f));
- yp=(int)(cy+sy/2.0f+sy/2.0f*cosf(angle2()/180.0f*3.141592f));
- fl_pie(xp-2,yp-2,4,4,0,360);
+ xp = (int)(cx + sx / 2.0f + sx / 2.0f * sinf(angle2() / 180.0f * 3.141592f));
+ yp = (int)(cy + sy / 2.0f + sy / 2.0f * cosf(angle2() / 180.0f * 3.141592f));
+ fl_pie(xp - 2, yp - 2, 4, 4, 0, 360);
fl_push_matrix();
- fl_translate(cx+sx/2,cy+sy/2);
- fl_rotate(a-90.0f);
+ fl_translate(cx + sx / 2, cy + sy / 2);
+ fl_rotate(a - 90.0f);
- fl_translate(sx/2,0);
+ fl_translate(sx / 2, 0);
fl_begin_polygon();
- pdialcolor(0,0,0);
- fl_vertex(-10,-4);
- fl_vertex(-10,4);
- fl_vertex(0,0);
+ pdialcolor(0, 0, 0);
+ fl_vertex(-10, -4);
+ fl_vertex(-10, 4);
+ fl_vertex(0, 0);
fl_end_polygon();
fl_pop_matrix();
}
-void WidgetPDial::pdialcolor(int r,int g,int b)
+void WidgetPDial::pdialcolor(int r, int g, int b)
{
- if (active_r())
- fl_color(r,g,b);
+ if(active_r())
+ fl_color(r, g, b);
else
- fl_color(160-(160-r)/3,160-(160-b)/3,160-(160-b)/3);
+ fl_color(160 - (160 - r) / 3, 160 - (160 - b) / 3, 160 - (160 - b) / 3);
}
-void WidgetPDial::tooltip(const char * c)
+void WidgetPDial::tooltip(const char *c)
{
tipwin->setText(c);
- textset=true;
+ textset = true;
}
void WidgetPDial::getPos()
{
if(!pos) {
- tipwin->position(Fl::event_x_root(), Fl::event_y_root()+20);
- pos=true;
+ tipwin->position(Fl::event_x_root(), Fl::event_y_root() + 20);
+ pos = true;
}
}
diff --git a/src/UI/WidgetPDial.h b/src/UI/WidgetPDial.h
@@ -5,21 +5,22 @@
#include <FL/Fl_Dial.H>
-class WidgetPDial : public Fl_Dial {
+class WidgetPDial:public Fl_Dial
+{
public:
- WidgetPDial(int x,int y, int w, int h, const char *label=0);
+ WidgetPDial(int x, int y, int w, int h, const char *label = 0);
~WidgetPDial();
int handle(int event);
- void drawgradient(int cx,int cy,int sx,double m1,double m2);
+ void drawgradient(int cx, int cy, int sx, double m1, double m2);
void draw();
- void pdialcolor(int r,int g,int b);
- void tooltip(const char * c);
+ void pdialcolor(int r, int g, int b);
+ void tooltip(const char *c);
private:
void getPos();
void resetPos();
double oldvalue;
- bool pos;
- bool textset;
- class TipWin *tipwin;
+ bool pos;
+ bool textset;
+ class TipWin * tipwin;
};
#endif
diff --git a/src/globals.h b/src/globals.h
@@ -154,7 +154,8 @@ extern int OSCIL_SIZE;
* How the amplitude threshold is computed
*/
#define ABOVE_AMPLITUDE_THRESHOLD(a, b) ((2.0f * fabs((b) - (a)) \
- / (fabs((b) + (a) + 0.0000000001f))) > \
+ / (fabs((b) + (a) \
+ + 0.0000000001f))) > \
AMPLITUDE_INTERPOLATION_THRESHOLD)
/*
@@ -177,10 +178,10 @@ extern int OSCIL_SIZE;
i++) \
data_[i] = 0;}
#define ZERO_float(data, size) {float *data_ = (float *) data; \
- for(int i = 0; \
- i < size; \
- i++) \
- data_[i] = 0.0f;}
+ for(int i = 0; \
+ i < size; \
+ i++) \
+ data_[i] = 0.0f;}
enum ONOFFTYPE {
OFF = 0, ON = 1
@@ -205,7 +206,8 @@ enum LegatoMsg {
#ifdef ASM_F2I_YES
#define F2I(f, \
i) __asm__ __volatile__ ("fistpl %0" : "=m" (i) : "t" (f \
- - 0.49999999f) \
+ - \
+ 0.49999999f) \
: "st");
#else
#define F2I(f, i) (i) = ((f > 0) ? ((int)(f)) : ((int)(f - 1.0f)));
@@ -218,4 +220,3 @@ enum LegatoMsg {
#endif
#endif
-
diff --git a/src/main.cpp b/src/main.cpp
@@ -59,14 +59,14 @@ using namespace std;
pthread_t thr3, thr4;
Master *master;
-int swaplr = 0; //1 for left-right swapping
+int swaplr = 0; //1 for left-right swapping
#if LASH
#include "Misc/LASHClient.h"
LASHClient *lash = NULL;
#endif
-int Pexitprogram = 0; //if the UI set this to 1, the program will exit
+int Pexitprogram = 0; //if the UI set this to 1, the program will exit
/*
* User Interface thread
@@ -175,7 +175,6 @@ void exitprogram()
int main(int argc, char *argv[])
{
-
config.init();
dump.startnow();
int noui = 0;
@@ -185,14 +184,14 @@ int main(int argc, char *argv[])
cerr << "Compiled: " << __DATE__ << " " << __TIME__ << endl;
cerr << "This program is free software (GNU GPL v.2 or later) and \n";
cerr << "it comes with ABSOLUTELY NO WARRANTY.\n" << endl;
- if(argc == 1)
- cerr << "Try 'zynaddsubfx --help' for command-line options." << endl;
+ if(argc == 1)
+ cerr << "Try 'zynaddsubfx --help' for command-line options." << endl;
/* Get the settings from the Config*/
SAMPLE_RATE = config.cfg.SampleRate;
SOUND_BUFFER_SIZE = config.cfg.SoundBufferSize;
- OSCIL_SIZE = config.cfg.OscilSize;
- swaplr = config.cfg.SwapStereo;
+ OSCIL_SIZE = config.cfg.OscilSize;
+ swaplr = config.cfg.SwapStereo;
sprng(time(NULL));
//produce denormal buf
@@ -202,21 +201,51 @@ int main(int argc, char *argv[])
/* Parse command-line options */
struct option opts[] = {
- {"load", 2, NULL, 'l'},
- {"load-instrument", 2, NULL, 'L'},
- {"sample-rate", 2, NULL, 'r'},
- {"buffer-size", 2, NULL, 'b'},
- {"oscil-size", 2, NULL, 'o'},
- {"dump", 2, NULL, 'D'},
- {"swap", 2, NULL, 'S'},
- {"no-gui", 2, NULL, 'U'},
- {"dummy", 2, NULL, 'Y'},
- {"help", 2, NULL, 'h'},
- {"named", 1, NULL, 'N'},
- {"auto-connect", 0, NULL, 'a'},
- {"output", 1, NULL, 'O'},
- {"input", 1, NULL, 'I'},
- {0, 0, 0, 0}
+ {
+ "load", 2, NULL, 'l'
+ },
+ {
+ "load-instrument", 2, NULL, 'L'
+ },
+ {
+ "sample-rate", 2, NULL, 'r'
+ },
+ {
+ "buffer-size", 2, NULL, 'b'
+ },
+ {
+ "oscil-size", 2, NULL, 'o'
+ },
+ {
+ "dump", 2, NULL, 'D'
+ },
+ {
+ "swap", 2, NULL, 'S'
+ },
+ {
+ "no-gui", 2, NULL, 'U'
+ },
+ {
+ "dummy", 2, NULL, 'Y'
+ },
+ {
+ "help", 2, NULL, 'h'
+ },
+ {
+ "named", 1, NULL, 'N'
+ },
+ {
+ "auto-connect", 0, NULL, 'a'
+ },
+ {
+ "output", 1, NULL, 'O'
+ },
+ {
+ "input", 1, NULL, 'I'
+ },
+ {
+ 0, 0, 0, 0
+ }
};
opterr = 0;
int option_index = 0, opt, exitwithhelp = 0;
@@ -227,89 +256,97 @@ int main(int argc, char *argv[])
int tmp = 0;
/**\todo check this process for a small memory leak*/
- opt = getopt_long(argc, argv, "l:L:r:b:o:I:O:N:haSDUY", opts, &option_index);
+ opt = getopt_long(argc,
+ argv,
+ "l:L:r:b:o:I:O:N:haSDUY",
+ opts,
+ &option_index);
char *optarguments = optarg;
-#define GETOP(x) if(optarguments) x = optarguments
-#define GETOPNUM(x) if(optarguments) x = atoi(optarguments)
+#define GETOP(x) if(optarguments) \
+ x = optarguments
+#define GETOPNUM(x) if(optarguments) \
+ x = atoi(optarguments)
if(opt == -1)
break;
switch(opt) {
- case 'h':
- exitwithhelp = 1;
- break;
- case 'Y':/* this command a dummy command (has NO effect)
+ case 'h':
+ exitwithhelp = 1;
+ break;
+ case 'Y':/* this command a dummy command (has NO effect)
and is used because I need for NSIS installer
(NSIS sometimes forces a command line for a
program, even if I don't need that; eg. when
I want to add a icon to a shortcut.
*/
- break;
- case 'U':
- noui = 1;
- break;
- case 'l':
- GETOP(loadfile);
- break;
- case 'L':
- GETOP(loadinstrument);
- break;
- case 'r':
- GETOPNUM(SAMPLE_RATE);
- if(SAMPLE_RATE < 4000) {
- cerr << "ERROR:Incorrect sample rate: " << optarguments << endl;
- exit(1);
- }
- break;
- case 'b':
- GETOPNUM(SOUND_BUFFER_SIZE);
- if(SOUND_BUFFER_SIZE < 2) {
- cerr << "ERROR:Incorrect buffer size: " << optarguments << endl;
- exit(1);
- }
- break;
- case 'o':
- if(optarguments)
- OSCIL_SIZE = tmp = atoi(optarguments);
- if(OSCIL_SIZE < MAX_AD_HARMONICS * 2)
- OSCIL_SIZE = MAX_AD_HARMONICS * 2;
- OSCIL_SIZE = (int) powf(2, ceil(logf(OSCIL_SIZE - 1.0f) / logf(2.0f)));
- if(tmp != OSCIL_SIZE) {
- cerr << "OSCIL_SIZE is wrong (must be 2^n) or too small. Adjusting to "
- << OSCIL_SIZE << "." << endl;
- }
- break;
- case 'S':
- swaplr = 1;
- break;
- case 'D':
- dump.startnow();
- break;
- case 'N':
- Nio::getInstance().setPostfix(optarguments);
- break;
- case 'I':
- if(optarguments) {
- if(Nio::getInstance().setDefaultSource(optarguments))
+ break;
+ case 'U':
+ noui = 1;
+ break;
+ case 'l':
+ GETOP(loadfile);
+ break;
+ case 'L':
+ GETOP(loadinstrument);
+ break;
+ case 'r':
+ GETOPNUM(SAMPLE_RATE);
+ if(SAMPLE_RATE < 4000) {
+ cerr << "ERROR:Incorrect sample rate: " << optarguments
+ << endl;
exit(1);
- }
- break;
- case 'O':
- if(optarguments) {
- if(Nio::getInstance().setDefaultSink(optarguments))
+ }
+ break;
+ case 'b':
+ GETOPNUM(SOUND_BUFFER_SIZE);
+ if(SOUND_BUFFER_SIZE < 2) {
+ cerr << "ERROR:Incorrect buffer size: " << optarguments
+ << endl;
exit(1);
- }
- break;
- case 'a':
- Nio::getInstance().autoConnect = true;
- break;
- case '?':
- cerr << "ERROR:Bad option or parameter.\n" << endl;
- exitwithhelp = 1;
- break;
+ }
+ break;
+ case 'o':
+ if(optarguments)
+ OSCIL_SIZE = tmp = atoi(optarguments);
+ if(OSCIL_SIZE < MAX_AD_HARMONICS * 2)
+ OSCIL_SIZE = MAX_AD_HARMONICS * 2;
+ OSCIL_SIZE =
+ (int) powf(2, ceil(logf(OSCIL_SIZE - 1.0f) / logf(2.0f)));
+ if(tmp != OSCIL_SIZE)
+ cerr
+ <<
+ "OSCIL_SIZE is wrong (must be 2^n) or too small. Adjusting to "
+ << OSCIL_SIZE << "." << endl;
+ break;
+ case 'S':
+ swaplr = 1;
+ break;
+ case 'D':
+ dump.startnow();
+ break;
+ case 'N':
+ Nio::getInstance().setPostfix(optarguments);
+ break;
+ case 'I':
+ if(optarguments)
+ if(Nio::getInstance().setDefaultSource(optarguments))
+ exit(1);
+ break;
+ case 'O':
+ if(optarguments)
+ if(Nio::getInstance().setDefaultSink(optarguments))
+ exit(1);
+ break;
+ case 'a':
+ Nio::getInstance().autoConnect = true;
+ break;
+ case '?':
+ cerr << "ERROR:Bad option or parameter.\n" << endl;
+ exitwithhelp = 1;
+ break;
}
}
@@ -319,11 +356,13 @@ int main(int argc, char *argv[])
<< " -l file, --load=FILE\t\t\t Loads a .xmz file\n"
<< " -L file, --load-instrument=FILE\t Loads a .xiz file\n"
<< " -r SR, --sample-rate=SR\t\t Set the sample rate SR\n"
- << " -b BS, --buffer-size=SR\t\t Set the buffer size (granularity)\n"
+ <<
+ " -b BS, --buffer-size=SR\t\t Set the buffer size (granularity)\n"
<< " -o OS, --oscil-size=OS\t\t Set the ADsynth oscil. size\n"
<< " -S , --swap\t\t\t\t Swap Left <--> Right\n"
<< " -D , --dump\t\t\t\t Dumps midi note ON/OFF commands\n"
- << " -U , --no-gui\t\t\t\t Run ZynAddSubFX without user interface\n"
+ <<
+ " -U , --no-gui\t\t\t\t Run ZynAddSubFX without user interface\n"
<< " -N , --named\t\t\t\t Postfix IO Name when possible\n"
<< " -a , --auto-connect\t\t\t AutoConnect when using JACK\n"
<< " -O , --output\t\t\t\t Set Output Engine\n"
@@ -332,7 +371,7 @@ int main(int argc, char *argv[])
return 0;
}
- initprogram(argc,argv);
+ initprogram(argc, argv);
#if 0 //TODO update this code
#ifdef USE_LASH
@@ -363,7 +402,8 @@ int main(int argc, char *argv[])
if(!loadinstrument.empty()) {
int loadtopart = 0;
- int tmp = master->part[loadtopart]->loadXMLinstrument(loadinstrument.c_str());
+ int tmp = master->part[loadtopart]->loadXMLinstrument(
+ loadinstrument.c_str());
if(tmp < 0) {
cerr << "ERROR: Could not load instrument file "
<< loadinstrument << '.' << endl;
@@ -380,7 +420,7 @@ int main(int argc, char *argv[])
#ifndef DISABLE_GUI
if(noui == 0)
- pthread_create(&thr3, NULL, thread3, (void*)!ioGood);
+ pthread_create(&thr3, NULL, thread3, (void *)!ioGood);
#endif
//TODO look into a conditional variable here, it seems to match usage
diff --git a/style.cfg b/style.cfg
@@ -0,0 +1,1218 @@
+# Uncrustify 0.53
+
+#
+# General options
+#
+
+# The type of line endings
+newlines = lf # auto/lf/crlf/cr
+
+# The original size of tabs in the input
+input_tab_size = 4 # number
+
+# The size of tabs in the output (only used if align_with_tabs=true)
+output_tab_size = 4 # number
+
+# The ascii value of the string escape char, usually 92 (\) or 94 (^). (Pawn)
+string_escape_char = 92 # number
+
+# Alternate string escape char for Pawn. Only works right before the quote char.
+string_escape_char2 = 0 # number
+
+#
+# Indenting
+#
+
+# The number of columns to indent per level.
+# Usually 2, 3, 4, or 8.
+indent_columns = 4 # number
+
+# How to use tabs when indenting code
+# 0=spaces only
+# 1=indent with tabs, align with spaces
+# 2=indent and align with tabs
+indent_with_tabs = 0 # number
+
+# Whether to indent strings broken by '\' so that they line up
+indent_align_string = true # false/true
+
+# The number of spaces to indent multi-line XML strings.
+# Requires indent_align_string=True
+indent_xml_string = 0 # number
+
+# Spaces to indent '{' from level
+indent_brace = 0 # number
+
+# Whether braces are indented to the body level
+indent_braces = false # false/true
+
+# Disabled indenting function braces if indent_braces is true
+indent_braces_no_func = false # false/true
+
+# Indent based on the size of the brace parent, ie 'if' => 3 spaces, 'for' => 4 spaces, etc.
+indent_brace_parent = false # false/true
+
+# Whether the 'namespace' body is indented
+indent_namespace = true # false/true
+
+# Whether the 'extern "C"' body is indented
+indent_extern = true # false/true
+
+# Whether the 'class' body is indented
+indent_class = true # false/true
+
+# Whether to indent the stuff after a leading class colon
+indent_class_colon = true # false/true
+
+# False=treat 'else\nif' as 'else if' for indenting purposes
+# True=indent the 'if' one level
+indent_else_if = false # false/true
+
+# Amount to indent variable declarations after a open brace. neg=relative, pos=absolute
+indent_var_def_blk = 0 # number
+
+# True: indent continued function call parameters one indent level
+# False: align parameters under the open paren
+indent_func_call_param = false # false/true
+
+# Same as indent_func_call_param, but for function defs
+indent_func_def_param = false # false/true
+
+# Same as indent_func_call_param, but for function protos
+indent_func_proto_param = false # false/true
+
+# Same as indent_func_call_param, but for class declarations
+indent_func_class_param = false # false/true
+
+# Same as indent_func_call_param, but for class variable constructors
+indent_func_ctor_var_param = false # false/true
+
+# Same as indent_func_call_param, but for templates
+indent_template_param = false # false/true
+
+# Double the indent for indent_func_xxx_param options
+indent_func_param_double = false # false/true
+
+# Indentation column for standalone 'const' function decl/proto qualifier
+indent_func_const = 0 # number
+
+# Indentation column for standalone 'throw' function decl/proto qualifier
+indent_func_throw = 0 # number
+
+# The number of spaces to indent a continued '->' or '.'
+# Usually set to 0, 1, or indent_columns.
+indent_member = 0 # number
+
+# Spaces to indent single line ('//') comments on lines before code
+indent_sing_line_comments = 0 # number
+
+# If set, will indent trailing single line ('//') comments relative
+# to the code instead of trying to keep the same absolute column
+indent_relative_single_line_comments = false # false/true
+
+# Spaces to indent 'case' from 'switch'
+# Usually 0 or indent_columns.
+indent_switch_case = 4 # number
+
+# Spaces to shift the 'case' line, without affecting any other lines
+# Usually 0.
+indent_case_shift = 0 # number
+
+# Spaces to indent '{' from 'case'.
+# By default, the brace will appear under the 'c' in case.
+# Usually set to 0 or indent_columns.
+indent_case_brace = 0 # number
+
+# Whether to indent comments found in first column
+indent_col1_comment = false # false/true
+
+# How to indent goto labels
+# >0 : absolute column where 1 is the leftmost column
+# <=0 : subtract from brace indent
+indent_label = 1 # number
+
+# Same as indent_label, but for access specifiers that are followed by a colon
+indent_access_spec = 1 # number
+
+# Indent the code after an access specifier by one level.
+# If set, this option forces 'indent_access_spec=0'
+indent_access_spec_body = true # false/true
+
+# If an open paren is followed by a newline, indent the next line so that it lines up after the open paren (not recommended)
+indent_paren_nl = false # false/true
+
+# Controls the indent of a close paren after a newline.
+# 0: Indent to body level
+# 1: Align under the open paren
+# 2: Indent to the brace level
+indent_paren_close = 1 # number
+
+# Controls the indent of a comma when inside a paren.If TRUE, aligns under the open paren
+indent_comma_paren = false # false/true
+
+# Controls the indent of a BOOL operator when inside a paren.If TRUE, aligns under the open paren
+indent_bool_paren = false # false/true
+
+# If an open square is followed by a newline, indent the next line so that it lines up after the open square (not recommended)
+indent_square_nl = false # false/true
+
+# Don't change the relative indent of ESQL/C 'EXEC SQL' bodies
+indent_preserve_sql = false # false/true
+
+# Align continued statements at the '='. Default=True
+# If FALSE or the '=' is followed by a newline, the next line is indent one tab.
+indent_align_assign = true # false/true
+
+#
+# Spacing options
+#
+
+# Add or remove space around arithmetic operator '+', '-', '/', '*', etc
+sp_arith = force # ignore/add/remove/force
+
+# Add or remove space around assignment operator '=', '+=', etc
+sp_assign = force # ignore/add/remove/force
+
+# Add or remove space before assignment operator '=', '+=', etc. Overrides sp_assign.
+sp_before_assign = ignore # ignore/add/remove/force
+
+# Add or remove space after assignment operator '=', '+=', etc. Overrides sp_assign.
+sp_after_assign = ignore # ignore/add/remove/force
+
+# Add or remove space around assignment '=' in enum
+sp_enum_assign = force # ignore/add/remove/force
+
+# Add or remove space before assignment '=' in enum. Overrides sp_enum_assign.
+sp_enum_before_assign = ignore # ignore/add/remove/force
+
+# Add or remove space after assignment '=' in enum. Overrides sp_enum_assign.
+sp_enum_after_assign = ignore # ignore/add/remove/force
+
+# Add or remove space around preprocessor '##' concatenation operator
+sp_pp_concat = add # ignore/add/remove/force
+
+# Add or remove space after preprocessor '#' stringify operator
+sp_pp_stringify = add # ignore/add/remove/force
+
+# Add or remove space around boolean operators '&&' and '||'
+sp_bool = force # ignore/add/remove/force
+
+# Add or remove space around compare operator '<', '>', '==', etc
+sp_compare = force # ignore/add/remove/force
+
+# Add or remove space inside '(' and ')'
+sp_inside_paren = remove # ignore/add/remove/force
+
+# Add or remove space between nested parens
+sp_paren_paren = remove # ignore/add/remove/force
+
+# Whether to balance spaces inside nested parens
+sp_balance_nested_parens = false # false/true
+
+# Add or remove space between ')' and '{'
+sp_paren_brace = force # ignore/add/remove/force
+
+# Add or remove space before pointer star '*'
+sp_before_ptr_star = force # ignore/add/remove/force
+
+# Add or remove space before pointer star '*' that isn't followed by a variable name
+# If set to 'ignore', sp_before_ptr_star is used instead.
+sp_before_unnamed_ptr_star = ignore # ignore/add/remove/force
+
+# Add or remove space between pointer stars '*'
+sp_between_ptr_star = remove # ignore/add/remove/force
+
+# Add or remove space after pointer star '*', if followed by a word.
+sp_after_ptr_star = remove # ignore/add/remove/force
+
+# Add or remove space after a pointer star '*', if followed by a func proto/def.
+sp_after_ptr_star_func = remove # ignore/add/remove/force
+
+# Add or remove space before a pointer star '*', if followed by a func proto/def.
+sp_before_ptr_star_func = force # ignore/add/remove/force
+
+# Add or remove space before a reference sign '&'
+sp_before_byref = force # ignore/add/remove/force
+
+# Add or remove space before a reference sign '&' that isn't followed by a variable name
+# If set to 'ignore', sp_before_byref is used instead.
+sp_before_unnamed_byref = ignore # ignore/add/remove/force
+
+# Add or remove space after reference sign '&', if followed by a word.
+sp_after_byref = remove # ignore/add/remove/force
+
+# Add or remove space after a reference sign '&', if followed by a func proto/def.
+sp_after_byref_func = ignore # ignore/add/remove/force
+
+# Add or remove space before a reference sign '&', if followed by a func proto/def.
+sp_before_byref_func = ignore # ignore/add/remove/force
+
+# Add or remove space between type and word
+sp_after_type = remove # ignore/add/remove/force
+
+# Add or remove space in 'template <' vs 'template<'.
+# If set to ignore, sp_before_angle is used.
+sp_template_angle = ignore # ignore/add/remove/force
+
+# Add or remove space before '<>'
+sp_before_angle = remove # ignore/add/remove/force
+
+# Add or remove space inside '<' and '>'
+sp_inside_angle = remove # ignore/add/remove/force
+
+# Add or remove space after '<>'
+sp_after_angle = ignore # ignore/add/remove/force
+
+# Add or remove space between '<>' and '(' as found in 'new List<byte>();'
+sp_angle_paren = remove # ignore/add/remove/force
+
+# Add or remove space between '<>' and a word as in 'List<byte> m;'
+sp_angle_word = ignore # ignore/add/remove/force
+
+# Add or remove space before '(' of 'if', 'for', 'switch', and 'while'
+sp_before_sparen = remove # ignore/add/remove/force
+
+# Add or remove space inside if-condition '(' and ')'
+sp_inside_sparen = remove # ignore/add/remove/force
+
+# Add or remove space before if-condition ')'. Overrides sp_inside_sparen.
+sp_inside_sparen_close = remove # ignore/add/remove/force
+
+# Add or remove space after ')' of 'if', 'for', 'switch', and 'while'
+sp_after_sparen = remove # ignore/add/remove/force
+
+# Add or remove space between ')' and '{' of 'if', 'for', 'switch', and 'while'
+sp_sparen_brace = force # ignore/add/remove/force
+
+# Add or remove space between 'invariant' and '(' in the D language.
+sp_invariant_paren = ignore # ignore/add/remove/force
+
+# Add or remove space after the ')' in 'invariant (C) c' in the D language.
+sp_after_invariant_paren = ignore # ignore/add/remove/force
+
+# Add or remove space before empty statement ';' on 'if', 'for' and 'while'
+sp_special_semi = ignore # ignore/add/remove/force
+
+# Add or remove space before ';'
+sp_before_semi = remove # ignore/add/remove/force
+
+# Add or remove space before ';' in non-empty 'for' statements
+sp_before_semi_for = remove # ignore/add/remove/force
+
+# Add or remove space before a semicolon of an empty part of a for statment.
+sp_before_semi_for_empty = remove # ignore/add/remove/force
+
+# Add or remove space after the final semicolon of an empty part of a for statment: for ( ; ; <here> ).
+sp_after_semi_for_empty = remove # ignore/add/remove/force
+
+# Add or remove space before '[' (except '[]')
+sp_before_square = ignore # ignore/add/remove/force
+
+# Add or remove space before '[]'
+sp_before_squares = ignore # ignore/add/remove/force
+
+# Add or remove space inside '[' and ']'
+sp_inside_square = remove # ignore/add/remove/force
+
+# Add or remove space after ','
+sp_after_comma = force # ignore/add/remove/force
+
+# Add or remove space before ','
+sp_before_comma = remove # ignore/add/remove/force
+
+# Add or remove space after class ':'
+sp_after_class_colon = remove # ignore/add/remove/force
+
+# Add or remove space before class ':'
+sp_before_class_colon = remove # ignore/add/remove/force
+
+# Add or remove space before case ':'
+sp_before_case_colon = remove # ignore/add/remove/force
+
+# Add or remove space between 'operator' and operator sign
+sp_after_operator = ignore # ignore/add/remove/force
+
+# Add or remove space between the operator symbol and the open paren, as in 'operator ++('
+sp_after_operator_sym = ignore # ignore/add/remove/force
+
+# Add or remove space after C/D cast, ie 'cast(int)a' vs 'cast(int) a' or '(int)a' vs '(int) a'
+sp_after_cast = ignore # ignore/add/remove/force
+
+# Add or remove spaces inside cast parens
+sp_inside_paren_cast = remove # ignore/add/remove/force
+
+# Add or remove space between the type and open paren in a C++ cast, ie 'int(exp)' vs 'int (exp)'
+sp_cpp_cast_paren = remove # ignore/add/remove/force
+
+# Add or remove space between 'sizeof' and '('
+sp_sizeof_paren = remove # ignore/add/remove/force
+
+# Add or remove space after the tag keyword (Pawn)
+sp_after_tag = ignore # ignore/add/remove/force
+
+# Add or remove space inside enum '{' and '}'
+sp_inside_braces_enum = ignore # ignore/add/remove/force
+
+# Add or remove space inside struct/union '{' and '}'
+sp_inside_braces_struct = remove # ignore/add/remove/force
+
+# Add or remove space inside '{' and '}'
+sp_inside_braces = ignore # ignore/add/remove/force
+
+# Add or remove space inside '{}'
+sp_inside_braces_empty = ignore # ignore/add/remove/force
+
+# Add or remove space between return type and function name
+# A minimum of 1 is forced except for pointer return types.
+sp_type_func = ignore # ignore/add/remove/force
+
+# Add or remove space between function name and '(' on function declaration
+sp_func_proto_paren = remove # ignore/add/remove/force
+
+# Add or remove space between function name and '(' on function definition
+sp_func_def_paren = remove # ignore/add/remove/force
+
+# Add or remove space inside empty function '()'
+sp_inside_fparens = remove # ignore/add/remove/force
+
+# Add or remove space inside function '(' and ')'
+sp_inside_fparen = remove # ignore/add/remove/force
+
+# Add or remove space between ']' and '(' when part of a function call.
+sp_square_fparen = ignore # ignore/add/remove/force
+
+# Add or remove space between ')' and '{' of function
+sp_fparen_brace = force # ignore/add/remove/force
+
+# Add or remove space between function name and '(' on function calls
+sp_func_call_paren = remove # ignore/add/remove/force
+
+# Add or remove space between the user function name and '(' on function calls
+# You need to set a keyword to be a user function, like this: 'set func_call_user _' in the config file.
+sp_func_call_user_paren = ignore # ignore/add/remove/force
+
+# Add or remove space between a constructor/destructor and the open paren
+sp_func_class_paren = remove # ignore/add/remove/force
+
+# Add or remove space between 'return' and '('
+sp_return_paren = remove # ignore/add/remove/force
+
+# Add or remove space between '__attribute__' and '('
+sp_attribute_paren = ignore # ignore/add/remove/force
+
+# Add or remove space between 'defined' and '(' in '#if defined (FOO)'
+sp_defined_paren = ignore # ignore/add/remove/force
+
+# Add or remove space between 'throw' and '(' in 'throw (something)'
+sp_throw_paren = ignore # ignore/add/remove/force
+
+# Add or remove space between macro and value
+sp_macro = ignore # ignore/add/remove/force
+
+# Add or remove space between macro function ')' and value
+sp_macro_func = ignore # ignore/add/remove/force
+
+# Add or remove space between 'else' and '{' if on the same line
+sp_else_brace = force # ignore/add/remove/force
+
+# Add or remove space between '}' and 'else' if on the same line
+sp_brace_else = ignore # ignore/add/remove/force
+
+# Add or remove space between '}' and the name of a typedef on the same line
+sp_brace_typedef = ignore # ignore/add/remove/force
+
+# Add or remove space between 'catch' and '{' if on the same line
+sp_catch_brace = ignore # ignore/add/remove/force
+
+# Add or remove space between '}' and 'catch' if on the same line
+sp_brace_catch = ignore # ignore/add/remove/force
+
+# Add or remove space between 'finally' and '{' if on the same line
+sp_finally_brace = ignore # ignore/add/remove/force
+
+# Add or remove space between '}' and 'finally' if on the same line
+sp_brace_finally = ignore # ignore/add/remove/force
+
+# Add or remove space between 'try' and '{' if on the same line
+sp_try_brace = ignore # ignore/add/remove/force
+
+# Add or remove space between get/set and '{' if on the same line
+sp_getset_brace = ignore # ignore/add/remove/force
+
+# Add or remove space before the '::' operator
+sp_before_dc = remove # ignore/add/remove/force
+
+# Add or remove space after the '::' operator
+sp_after_dc = remove # ignore/add/remove/force
+
+# Add or remove around the D named array initializer ':' operator
+sp_d_array_colon = ignore # ignore/add/remove/force
+
+# Add or remove space after the '!' (not) operator.
+sp_not = remove # ignore/add/remove/force
+
+# Add or remove space after the '~' (invert) operator.
+sp_inv = remove # ignore/add/remove/force
+
+# Add or remove space after the '&' (address-of) operator.
+# This does not affect the spacing after a '&' that is part of a type.
+sp_addr = remove # ignore/add/remove/force
+
+# Add or remove space around the '.' or '->' operators
+sp_member = remove # ignore/add/remove/force
+
+# Add or remove space after the '*' (dereference) operator.
+# This does not affect the spacing after a '*' that is part of a type.
+sp_deref = remove # ignore/add/remove/force
+
+# Add or remove space after '+' or '-', as in 'x = -5' or 'y = +7'
+sp_sign = remove # ignore/add/remove/force
+
+# Add or remove space before or after '++' and '--', as in '(--x)' or 'y++;'
+sp_incdec = remove # ignore/add/remove/force
+
+# Add or remove space before a backslash-newline at the end of a line
+sp_before_nl_cont = force # ignore/add/remove/force
+
+# Add or remove space after the scope '+' or '-', as in '-(void) foo;' or '+(int) bar;'
+sp_after_oc_scope = ignore # ignore/add/remove/force
+
+# Add or remove space after the colon in message specs
+# '-(int) f:(int) x;' vs '-(int) f: (int) x;'
+sp_after_oc_colon = ignore # ignore/add/remove/force
+
+# Add or remove space before the colon in message specs
+# '-(int) f: (int) x;' vs '-(int) f : (int) x;'
+sp_before_oc_colon = ignore # ignore/add/remove/force
+
+# Add or remove space after the colon in message specs
+# '[object setValue:1];' vs '[object setValue: 1];'
+sp_after_send_oc_colon = ignore # ignore/add/remove/force
+
+# Add or remove space before the colon in message specs
+# '[object setValue:1];' vs '[object setValue :1];'
+sp_before_send_oc_colon = ignore # ignore/add/remove/force
+
+# Add or remove space after the (type) in message specs
+# '-(int) f: (int) x;' vs '-(int) f: (int)x;'
+sp_after_oc_type = ignore # ignore/add/remove/force
+
+# Add or remove space around the ':' in 'b ? t : f'
+sp_cond_colon = ignore # ignore/add/remove/force
+
+# Add or remove space around the '?' in 'b ? t : f'
+sp_cond_question = ignore # ignore/add/remove/force
+
+# Fix the spacing between 'case' and the label. Only 'ignore' and 'force' make sense here.
+sp_case_label = ignore # ignore/add/remove/force
+
+# Control the space around the D '..' operator.
+sp_range = ignore # ignore/add/remove/force
+
+# Control the space after the opening of a C++ comment '// A' vs '//A'
+sp_cmt_cpp_start = ignore # ignore/add/remove/force
+
+#
+# Code alignment (not left column spaces/tabs)
+#
+
+# Whether to keep non-indenting tabs
+align_keep_tabs = false # false/true
+
+# Whether to use tabs for alinging
+align_with_tabs = false # false/true
+
+# Whether to bump out to the next tab when aligning
+align_on_tabstop = false # false/true
+
+# Whether to left-align numbers
+align_number_left = false # false/true
+
+# Align variable definitions in prototypes and functions
+align_func_params = false # false/true
+
+# Align parameters in single-line functions that have the same name.
+# The function names must already be aligned with each other.
+align_same_func_call_params = false # false/true
+
+# The span for aligning variable definitions (0=don't align)
+align_var_def_span = 1 # number
+
+# How to align the star in variable definitions.
+# 0=Part of the type 'void * foo;'
+# 1=Part of the variable 'void *foo;'
+# 2=Dangling 'void *foo;'
+align_var_def_star_style = 2 # number
+
+# How to align the '&' in variable definitions.
+# 0=Part of the type
+# 1=Part of the variable
+# 2=Dangling
+align_var_def_amp_style = 2 # number
+
+# The threshold for aligning variable definitions (0=no limit)
+align_var_def_thresh = 5 # number
+
+# The gap for aligning variable definitions
+align_var_def_gap = 1 # number
+
+# Whether to align the colon in struct bit fields
+align_var_def_colon = false # false/true
+
+# Whether to align any attribute after the variable name
+align_var_def_attribute = false # false/true
+
+# Whether to align inline struct/enum/union variable definitions
+align_var_def_inline = false # false/true
+
+# The span for aligning on '=' in assignments (0=don't align)
+align_assign_span = 1 # number
+
+# The threshold for aligning on '=' in assignments (0=no limit)
+align_assign_thresh = 5 # number
+
+# The span for aligning on '=' in enums (0=don't align)
+align_enum_equ_span = 1 # number
+
+# The threshold for aligning on '=' in enums (0=no limit)
+align_enum_equ_thresh = 5 # number
+
+# The span for aligning struct/union (0=don't align)
+align_var_struct_span = 1 # number
+
+# The threshold for aligning struct/union member definitions (0=no limit)
+align_var_struct_thresh = 5 # number
+
+# The gap for aligning struct/union member definitions
+align_var_struct_gap = 0 # number
+
+# The span for aligning struct initializer values (0=don't align)
+align_struct_init_span = 1 # number
+
+# The minimum space between the type and the synonym of a typedef
+align_typedef_gap = 0 # number
+
+# The span for aligning single-line typedefs (0=don't align)
+align_typedef_span = 1 # number
+
+# How to align typedef'd functions with other typedefs
+# 0: Don't mix them at all
+# 1: align the open paren with the types
+# 2: align the function type name with the other type names
+align_typedef_func = 0 # number
+
+# Controls the positioning of the '*' in typedefs. Just try it.
+# 0: Align on typdef type, ignore '*'
+# 1: The '*' is part of type name: typedef int *pint;
+# 2: The '*' is part of the type, but dangling: typedef int *pint;
+align_typedef_star_style = 0 # number
+
+# Controls the positioning of the '&' in typedefs. Just try it.
+# 0: Align on typdef type, ignore '&'
+# 1: The '&' is part of type name: typedef int &pint;
+# 2: The '&' is part of the type, but dangling: typedef int &pint;
+align_typedef_amp_style = 0 # number
+
+# The span for aligning comments that end lines (0=don't align)
+align_right_cmt_span = 0 # number
+
+# If aligning comments, mix with comments after '}' and #endif with less than 3 spaces before the comment
+align_right_cmt_mix = false # false/true
+
+# If a trailing comment is more than this number of columns away from the text it follows,
+# it will qualify for being aligned.
+align_right_cmt_gap = 0 # number
+
+# Align trailing comment at or beyond column N; 'pulls in' comments as a bonus side effect (0=ignore)
+align_right_cmt_at_col = 0 # number
+
+# The span for aligning function prototypes (0=don't align)
+align_func_proto_span = 0 # number
+
+# Minimum gap between the return type and the function name.
+align_func_proto_gap = 0 # number
+
+# Align function protos on the 'operator' keyword instead of what follows
+align_on_operator = false # false/true
+
+# Whether to mix aligning prototype and variable declarations.
+# If true, align_var_def_XXX options are used instead of align_func_proto_XXX options.
+align_mix_var_proto = false # false/true
+
+# Align single-line functions with function prototypes, uses align_func_proto_span
+align_single_line_func = false # false/true
+
+# Aligning the open brace of single-line functions.
+# Requires align_single_line_func=true, uses align_func_proto_span
+align_single_line_brace = false # false/true
+
+# Gap for align_single_line_brace.
+align_single_line_brace_gap = 0 # number
+
+# The span for aligning ObjC msg spec (0=don't align)
+align_oc_msg_spec_span = 0 # number
+
+# Whether to align macros wrapped with a backslash and a newline.
+# This will not work right if the macro contains a multi-line comment.
+align_nl_cont = false # false/true
+
+# The minimum space between label and value of a preprocessor define
+align_pp_define_gap = 0 # number
+
+# The span for aligning on '#define' bodies (0=don't align)
+align_pp_define_span = 0 # number
+
+# Align lines that start with '<<' with previous '<<'. Default=true
+align_left_shift = true # false/true
+
+#
+# Newline adding and removing options
+#
+
+# Whether to collapse empty blocks between '{' and '}'
+nl_collapse_empty_body = true # false/true
+
+# Don't split one-line braced assignments - 'foo_t f = { 1, 2 };'
+nl_assign_leave_one_liners = true # false/true
+
+# Don't split one-line braced statements inside a class xx { } body
+nl_class_leave_one_liners = false # false/true
+
+# Don't split one-line enums: 'enum foo { BAR = 15 };'
+nl_enum_leave_one_liners = false # false/true
+
+# Don't split one-line get or set functions
+nl_getset_leave_one_liners = true # false/true
+
+# Don't split one-line function definitions - 'int foo() { return 0; }'
+nl_func_leave_one_liners = true # false/true
+
+# Don't split one-line if/else statements - 'if(a) b++;'
+nl_if_leave_one_liners = false # false/true
+
+# Add or remove newlines at the start of the file
+nl_start_of_file = ignore # ignore/add/remove/force
+
+# The number of newlines at the start of the file (only used if nl_start_of_file is 'add' or 'force'
+nl_start_of_file_min = 0 # number
+
+# Add or remove newline at the end of the file
+nl_end_of_file = force # ignore/add/remove/force
+
+# The number of newlines at the end of the file (only used if nl_end_of_file is 'add' or 'force')
+nl_end_of_file_min = 1 # number
+
+# Add or remove newline between '=' and '{'
+nl_assign_brace = ignore # ignore/add/remove/force
+
+# Add or remove newline between '=' and '[' (D only)
+nl_assign_square = ignore # ignore/add/remove/force
+
+# Add or remove newline after '= [' (D only). Will also affect the newline before the ']'
+nl_after_square_assign = ignore # ignore/add/remove/force
+
+# The number of newlines after a block of variable definitions
+nl_func_var_def_blk = 0 # number
+
+# Add or remove newline between a function call's ')' and '{', as in:
+# list_for_each(item, &list) { }
+nl_fcall_brace = force # ignore/add/remove/force
+
+# Add or remove newline between 'enum' and '{'
+nl_enum_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between 'struct and '{'
+nl_struct_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between 'union' and '{'
+nl_union_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between 'if' and '{'
+nl_if_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between '}' and 'else'
+nl_brace_else = force # ignore/add/remove/force
+
+# Add or remove newline between 'else if' and '{'
+# If set to ignore, nl_if_brace is used instead
+nl_elseif_brace = ignore # ignore/add/remove/force
+
+# Add or remove newline between 'else' and '{'
+nl_else_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between 'else' and 'if'
+nl_else_if = add # ignore/add/remove/force
+
+# Add or remove newline between '}' and 'finally'
+nl_brace_finally = force # ignore/add/remove/force
+
+# Add or remove newline between 'finally' and '{'
+nl_finally_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between 'try' and '{'
+nl_try_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between get/set and '{'
+nl_getset_brace = ignore # ignore/add/remove/force
+
+# Add or remove newline between 'for' and '{'
+nl_for_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between 'catch' and '{'
+nl_catch_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between '}' and 'catch'
+nl_brace_catch = force # ignore/add/remove/force
+
+# Add or remove newline between 'while' and '{'
+nl_while_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between 'do' and '{'
+nl_do_brace = remove # ignore/add/remove/force
+
+# Add or remove newline between '}' and 'while' of 'do' statement
+nl_brace_while = remove # ignore/add/remove/force
+
+# Add or remove newline between 'switch' and '{'
+nl_switch_brace = remove # ignore/add/remove/force
+
+# Add a newline between ')' and '{' if the ')' is on a different line than the if/for/etc.
+# Overrides nl_for_brace, nl_if_brace, nl_switch_brace, nl_while_switch, and nl_catch_brace.
+nl_multi_line_cond = false # false/true
+
+# Force a newline in a define after the macro name for multi-line defines.
+nl_multi_line_define = false # false/true
+
+# Whether to put a newline before 'case' statement
+nl_before_case = false # false/true
+
+# Add or remove newline between ')' and 'throw'
+nl_before_throw = ignore # ignore/add/remove/force
+
+# Whether to put a newline after 'case' statement
+nl_after_case = false # false/true
+
+# Newline between namespace and {
+nl_namespace_brace = ignore # ignore/add/remove/force
+
+# Add or remove newline between 'template<>' and whatever follows.
+nl_template_class = ignore # ignore/add/remove/force
+
+# Add or remove newline between 'class' and '{'
+nl_class_brace = force # ignore/add/remove/force
+
+# Add or remove newline after each ',' in the constructor member initialization
+nl_class_init_args = ignore # ignore/add/remove/force
+
+# Add or remove newline between return type and function name in definition
+nl_func_type_name = ignore # ignore/add/remove/force
+
+# Add or remove newline between function scope and name in a definition
+# Controls the newline after '::' in 'void A::f() { }'
+nl_func_scope_name = ignore # ignore/add/remove/force
+
+# Add or remove newline between return type and function name in a prototype
+nl_func_proto_type_name = ignore # ignore/add/remove/force
+
+# Add or remove newline between a function name and the opening '('
+nl_func_paren = remove # ignore/add/remove/force
+
+# Add or remove newline after '(' in a function declaration
+nl_func_decl_start = ignore # ignore/add/remove/force
+
+# Add or remove newline after each ',' in a function declaration
+nl_func_decl_args = ignore # ignore/add/remove/force
+
+# Add or remove newline before the ')' in a function declaration
+nl_func_decl_end = ignore # ignore/add/remove/force
+
+# Add or remove newline between function signature and '{'
+nl_fdef_brace = ignore # ignore/add/remove/force
+
+# Whether to put a newline after 'return' statement
+nl_after_return = false # false/true
+
+# Add or remove a newline between the return keyword and return expression.
+nl_return_expr = ignore # ignore/add/remove/force
+
+# Whether to put a newline after semicolons, except in 'for' statements
+nl_after_semicolon = false # false/true
+
+# Whether to put a newline after brace open.
+# This also adds a newline before the matching brace close.
+nl_after_brace_open = true # false/true
+
+# If nl_after_brace_open and nl_after_brace_open_cmt are true, a newline is
+# placed between the open brace and a trailing single-line comment.
+nl_after_brace_open_cmt = false # false/true
+
+# Whether to put a newline after a virtual brace open.
+# These occur in un-braced if/while/do/for statement bodies.
+nl_after_vbrace_open = true # false/true
+
+# Whether to put a newline after a brace close.
+# Does not apply if followed by a necessary ';'.
+nl_after_brace_close = false # false/true
+
+# Whether to alter newlines in '#define' macros
+nl_define_macro = false # false/true
+
+# Whether to not put blanks after '#ifxx', '#elxx', or before '#endif'
+nl_squeeze_ifdef = false # false/true
+
+# Add or remove newline before 'if'
+nl_before_if = ignore # ignore/add/remove/force
+
+# Add or remove newline after 'if'
+nl_after_if = ignore # ignore/add/remove/force
+
+# Add or remove newline before 'for'
+nl_before_for = ignore # ignore/add/remove/force
+
+# Add or remove newline after 'for'
+nl_after_for = ignore # ignore/add/remove/force
+
+# Add or remove newline before 'while'
+nl_before_while = ignore # ignore/add/remove/force
+
+# Add or remove newline after 'while'
+nl_after_while = ignore # ignore/add/remove/force
+
+# Add or remove newline before 'switch'
+nl_before_switch = ignore # ignore/add/remove/force
+
+# Add or remove newline after 'switch'
+nl_after_switch = ignore # ignore/add/remove/force
+
+# Add or remove newline before 'do'
+nl_before_do = ignore # ignore/add/remove/force
+
+# Add or remove newline after 'do'
+nl_after_do = ignore # ignore/add/remove/force
+
+# Whether to double-space commented-entries in struct/enum
+nl_ds_struct_enum_cmt = false # false/true
+
+# Whether to double-space before the close brace of a struct/union/enum
+nl_ds_struct_enum_close_brace = false # false/true
+
+# Add or remove a newline around a class colon.
+# Related to pos_class_colon, nl_class_init_args, and pos_comma.
+nl_class_colon = ignore # ignore/add/remove/force
+
+# Change simple unbraced if statements into a one-liner
+# 'if(b)\n i++;' => 'if(b) i++;'
+nl_create_if_one_liner = false # false/true
+
+# Change simple unbraced for statements into a one-liner
+# 'for (i=0;i<5;i++)\n foo(i);' => 'for (i=0;i<5;i++) foo(i);'
+nl_create_for_one_liner = false # false/true
+
+# Change simple unbraced while statements into a one-liner
+# 'while (i<5)\n foo(i++);' => 'while (i<5) foo(i++);'
+nl_create_while_one_liner = false # false/true
+
+#
+# Positioning options
+#
+
+# The position of arithmetic operators in wrapped expressions
+pos_arith = lead # ignore/lead/trail
+
+# The position of assignment in wrapped expressions
+pos_assign = trail # ignore/lead/trail
+
+# The position of boolean operators in wrapped expressions
+pos_bool = lead # ignore/lead/trail
+
+# The position of the comma in wrapped expressions
+pos_comma = trail # ignore/lead/trail
+
+# The position of the comma in the constructor initialization list
+pos_class_comma = trail # ignore/lead/trail
+
+# The position of colons between constructor and member initialization
+pos_class_colon = lead # ignore/lead/trail
+
+#
+# Line Splitting options
+#
+
+# Try to limit code width to N number of columns
+code_width = 80 # number
+
+# Whether to fully split long 'for' statements at semi-colons
+ls_for_split_full = true # false/true
+
+# Whether to fully split long function protos/calls at commas
+ls_func_split_full = true # false/true
+
+#
+# Blank line options
+#
+
+# The maximum consecutive newlines
+nl_max = 0 # number
+
+# The number of newlines after a function prototype, if followed by another function prototype
+nl_after_func_proto = 0 # number
+
+# The number of newlines after a function prototype, if not followed by another function prototype
+nl_after_func_proto_group = 0 # number
+
+# The number of newlines after '}' of a multi-line function body
+nl_after_func_body = 0 # number
+
+# The number of newlines after '}' of a single line function body
+nl_after_func_body_one_liner = 0 # number
+
+# The minimum number of newlines before a multi-line comment.
+# Doesn't apply if after a brace open or another multi-line comment.
+nl_before_block_comment = 0 # number
+
+# The minimum number of newlines before a single-line C comment.
+# Doesn't apply if after a brace open or other single-line C comments.
+nl_before_c_comment = 0 # number
+
+# The minimum number of newlines before a CPP comment.
+# Doesn't apply if after a brace open or other CPP comments.
+nl_before_cpp_comment = 0 # number
+
+# Whether to force a newline after a mulit-line comment.
+nl_after_multiline_comment = false # false/true
+
+# The number of newlines before a 'private:', 'public:', 'protected:', 'signals:', or 'slots:' label.
+# Will not change the newline count if after a brace open.
+# 0 = No change.
+nl_before_access_spec = 0 # number
+
+# The number of newlines after a 'private:', 'public:', 'protected:', 'signals:', or 'slots:' label.
+# 0 = No change.
+nl_after_access_spec = 0 # number
+
+# The number of newlines between a function def and the function comment.
+# 0 = No change.
+nl_comment_func_def = 0 # number
+
+# The number of newlines after a try-catch-finally block that isn't followed by a brace close.
+# 0 = No change.
+nl_after_try_catch_finally = 0 # number
+
+# The number of newlines before and after a property, indexer or event decl.
+# 0 = No change.
+nl_around_cs_property = 0 # number
+
+# The number of newlines between the get/set/add/remove handlers in C#.
+# 0 = No change.
+nl_between_get_set = 0 # number
+
+# Whether to remove blank lines after '{'
+eat_blanks_after_open_brace = true # false/true
+
+# Whether to remove blank lines before '}'
+eat_blanks_before_close_brace = true # false/true
+
+#
+# Code modifying options (non-whitespace)
+#
+
+# Add or remove braces on single-line 'do' statement
+mod_full_brace_do = ignore # ignore/add/remove/force
+
+# Add or remove braces on single-line 'for' statement
+mod_full_brace_for = remove # ignore/add/remove/force
+
+# Add or remove braces on single-line function defintions. (Pawn)
+mod_full_brace_function = ignore # ignore/add/remove/force
+
+# Add or remove braces on single-line 'if' statement. Will not remove the braces if they contain an 'else'.
+mod_full_brace_if = remove # ignore/add/remove/force
+
+# Don't remove braces around statements that span N newlines
+mod_full_brace_nl = 0 # number
+
+# Add or remove braces on single-line 'while' statement
+mod_full_brace_while = ignore # ignore/add/remove/force
+
+# Add or remove unnecessary paren on 'return' statement
+mod_paren_on_return = remove # ignore/add/remove/force
+
+# Whether to change optional semicolons to real semicolons
+mod_pawn_semicolon = false # false/true
+
+# Add parens on 'while' and 'if' statement around bools
+mod_full_paren_if_bool = true # false/true
+
+# Whether to remove superfluous semicolons
+mod_remove_extra_semicolon = true # false/true
+
+# If a function body exceeds the specified number of newlines and doesn't have a comment after
+# the close brace, a comment will be added.
+mod_add_long_function_closebrace_comment = 0 # number
+
+# If a switch body exceeds the specified number of newlines and doesn't have a comment after
+# the close brace, a comment will be added.
+mod_add_long_switch_closebrace_comment = 0 # number
+
+# If an #ifdef body exceeds the specified number of newlines and doesn't have a comment after
+# the #else, a comment will be added.
+mod_add_long_ifdef_endif_comment = 0 # number
+
+# If an #ifdef or #else body exceeds the specified number of newlines and doesn't have a comment after
+# the #endif, a comment will be added.
+mod_add_long_ifdef_else_comment = 0 # number
+
+# If TRUE, will sort consecutive single-line 'import' statements [Java, D]
+mod_sort_import = false # false/true
+
+# If TRUE, will sort consecutive single-line 'using' statements [C#]
+mod_sort_using = false # false/true
+
+# If TRUE, will sort consecutive single-line '#include' statements [C/C++] and '#import' statements [Obj-C]
+# This is generally a bad idea, as it may break your code.
+mod_sort_include = false # false/true
+
+# If TRUE, it will move a 'break' that appears after a fully braced 'case' before the close brace.
+mod_move_case_break = false # false/true
+
+# If TRUE, it will remove a void 'return;' that appears as the last statement in a function.
+mod_remove_empty_return = false # false/true
+
+#
+# Comment modifications
+#
+
+# Try to wrap comments at cmt_width columns
+cmt_width = 0 # number
+
+# If false, disable all multi-line comment changes, including cmt_width and leading chars.
+# Default is true.
+cmt_indent_multi = false # false/true
+
+# Whether to group c-comments that look like they are in a block
+cmt_c_group = false # false/true
+
+# Whether to put an empty '/*' on the first line of the combined c-comment
+cmt_c_nl_start = false # false/true
+
+# Whether to put a newline before the closing '*/' of the combined c-comment
+cmt_c_nl_end = false # false/true
+
+# Whether to group cpp-comments that look like they are in a block
+cmt_cpp_group = false # false/true
+
+# Whether to put an empty '/*' on the first line of the combined cpp-comment
+cmt_cpp_nl_start = false # false/true
+
+# Whether to put a newline before the closing '*/' of the combined cpp-comment
+cmt_cpp_nl_end = false # false/true
+
+# Whether to change cpp-comments into c-comments
+cmt_cpp_to_c = false # false/true
+
+# Whether to put a star on subsequent comment lines
+cmt_star_cont = false # false/true
+
+# The number of spaces to insert at the start of subsequent comment lines
+cmt_sp_before_star_cont = 0 # number
+
+# The number of spaces to insert after the star on subsequent comment lines
+cmt_sp_after_star_cont = 1 # number
+
+# For multi-line comments with a '*' lead, remove leading spaces if the first and last lines of
+# the comment are the same length. Default=True
+cmt_multi_check_last = true # false/true
+
+# The filename that contains text to insert at the head of a file if the file doesn't start with a C/C++ comment.
+# Will substitue $(filename) with the current file's name.
+cmt_insert_file_header = "" # string
+
+# The filename that contains text to insert at the end of a file if the file doesn't end with a C/C++ comment.
+# Will substitue $(filename) with the current file's name.
+cmt_insert_file_footer = "" # string
+
+# The filename that contains text to insert before a function implementation if the function isn't preceeded with a C/C++ comment.
+# Will substitue $(function) with the function name and $(javaparam) with the javadoc @param and @return stuff.
+# Will also substitute $(fclass) with the class name: void CFoo::Bar() { ... }
+cmt_insert_func_header = "" # string
+
+# The filename that contains text to insert before a class if the class isn't preceeded with a C/C++ comment.
+# Will substitue $(class) with the class name.
+cmt_insert_class_header = "" # string
+
+# If a preprocessor is encountered when stepping backwards from a function name, then
+# this option decides whether the comment should be inserted.
+# Affects cmt_insert_func_header and cmt_insert_class_header.
+cmt_insert_before_preproc = false # false/true
+
+#
+# Preprocessor options
+#
+
+# Control indent of preprocessors inside #if blocks at brace level 0
+pp_indent = ignore # ignore/add/remove/force
+
+# Whether to indent #if/#else/#endif at the brace level (true) or from column 1 (false)
+pp_indent_at_level = false # false/true
+
+# If pp_indent_at_level=false, specifies the number of columns to indent per level. Default=1.
+pp_indent_count = 1 # number
+
+# Add or remove space after # based on pp_level of #if blocks
+pp_space = ignore # ignore/add/remove/force
+
+# Sets the number of spaces added with pp_space
+pp_space_count = 0 # number
+
+# The indent for #region and #endregion in C# and '#pragma region' in C/C++
+pp_indent_region = 0 # number
+
+# Whether to indent the code between #region and #endregion
+pp_region_indent_code = false # false/true
+
+# If pp_indent_at_level=true, sets the indent for #if, #else, and #endif when not at file-level
+pp_indent_if = 0 # number
+
+# Control whether to indent the code between #if, #else and #endif when not at file-level
+pp_if_indent_code = false # false/true
+
+# Whether to indent '#define' at the brace level (true) or from column 1 (false)
+pp_define_at_level = false # false/true
+
+# You can force a token to be a type with the 'type' option.
+# Example:
+# type myfoo1 myfoo2
+#
+# You can create custom macro-based indentation using macro-open,
+# macro-else and macro-close.
+# Example:
+# macro-open BEGIN_TEMPLATE_MESSAGE_MAP
+# macro-open BEGIN_MESSAGE_MAP
+# macro-close END_MESSAGE_MAP
+#
+# You can assign any keyword to any type with the set option.
+# set func_call_user _ N_
+#
+# The full syntax description of all custom definition config entries
+# is shown below:
+#
+# define custom tokens as:
+# - embed whitespace in token using '' escape character, or
+# put token in quotes
+# - these: ' " and ` are recognized as quote delimiters
+#
+# type token1 token2 token3 ...
+# ^ optionally specify multiple tokens on a single line
+# define def_token output_token
+# ^ output_token is optional, then NULL is assumed
+# macro-open token
+# macro-close token
+# macro-else token
+# set id token1 token2 ...
+# ^ optionally specify multiple tokens on a single line
+# ^ id is one of the names in token_enum.h sans the CT_ prefix,
+# e.g. PP_PRAGMA
+#
+# all tokens are separated by any mix of ',' commas, '=' equal signs
+# and whitespace (space, tab)
+#
diff --git a/style.sh b/style.sh
@@ -0,0 +1,3 @@
+#!/bin/sh
+uncrustify -c style.cfg --no-backup -l CPP `find . | grep -e "\.h$"`
+uncrustify -c style.cfg --no-backup -l CPP `find . | grep -e "\.cpp$"`
