commit 6c127c360a86633fdc5d8b4ea340bb1157112cb0
parent 7f1a38a479c98ad795d1fe489db0a756ec72cdef
Author: Harald Hvaal <[email protected]>
Date: Tue, 23 Jun 2009 22:25:12 +0900
Ran astyle on the whole codebase, with the following command:
find -name \*.C -or -name \*.cpp -or -name \*.h | xargs astyle --style=kr -l
Diffstat:
131 files changed, 12910 insertions(+), 11566 deletions(-)
diff --git a/src/Controls/Control.C b/src/Controls/Control.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Control.C - Control base class
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -22,7 +22,7 @@
#include "Control.h"
Control::Control(char ndefaultval)
- :defaultval(ndefaultval),lockqueue(-1),locked(false)
+ :defaultval(ndefaultval),lockqueue(-1),locked(false)
{}
void Control::lock()
@@ -33,7 +33,7 @@ void Control::lock()
void Control::ulock()
{
- if(locked&&lockqueue>=0)
+ if (locked&&lockqueue>=0)
setmVal(lockqueue);
locked=false;
}
diff --git a/src/Controls/Control.h b/src/Controls/Control.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Control.h - Control base class
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,33 +25,33 @@
/**A control for a parameter within the program*/
class Control
{
- public:
- Control(char ndefaultval);/**\todo create proper initialization list*/
- ~Control(){};
- /**Return the string, which represents the internal value
- * @return a string representation of the current value*/
- virtual std::string getString()const=0;
- /**Set the Control to the given value
- * @param nval A number 0-127*/
- virtual void setmVal(char nval)=0;
- /**Return the midi value (aka the char)
- * @return the current value*/
- virtual char getmVal()const=0;
- /** Will lock the Control until it is ulocked
- *
- * Locking a Control will Stop it from having
- * its internal data altered*/
- void lock();
- /** Will unlock the Control
- *
- * This will also update the Control
- * if something attempted to update it while it was locked*/
- void ulock();
- private:
- char defaultval;/**<Default value of the Control*/
- char lockqueue; /**<The value is that is stored, while the Control is locked
+public:
+ Control(char ndefaultval);/**\todo create proper initialization list*/
+ ~Control() {};
+ /**Return the string, which represents the internal value
+ * @return a string representation of the current value*/
+ virtual std::string getString()const=0;
+ /**Set the Control to the given value
+ * @param nval A number 0-127*/
+ virtual void setmVal(char nval)=0;
+ /**Return the midi value (aka the char)
+ * @return the current value*/
+ virtual char getmVal()const=0;
+ /** Will lock the Control until it is ulocked
+ *
+ * Locking a Control will Stop it from having
+ * its internal data altered*/
+ void lock();
+ /** Will unlock the Control
+ *
+ * This will also update the Control
+ * if something attempted to update it while it was locked*/
+ void ulock();
+private:
+ char defaultval;/**<Default value of the Control*/
+ char lockqueue; /**<The value is that is stored, while the Control is locked
* and something attempts to update it*/
- bool locked;//upgrade this to a integer lock level if problems occur
+ bool locked;//upgrade this to a integer lock level if problems occur
};
#endif
diff --git a/src/Controls/DelayCtl.C b/src/Controls/DelayCtl.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
DelayCtl.C - Control For Delays
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -23,7 +23,7 @@
#include <sstream>
DelayCtl::DelayCtl()
- :Control(64),value(64/127.0*1.5){} /**\todo finishme*/
+ :Control(64),value(64/127.0*1.5) {} /**\todo finishme*/
std::string DelayCtl::getString() const
{
diff --git a/src/Controls/DelayCtl.h b/src/Controls/DelayCtl.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
DelayCtl.h - Control For Delays
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,19 +24,19 @@
#ifndef DELAYCTL_H
#define DELAYCTL_H
/**A Control for Delays
- *
+ *
* Will vary from 0 seconds to 1.5 seconds*/
class DelayCtl:public Control
{
- public:
- DelayCtl();
- ~DelayCtl(){};
- std::string getString() const;
- void setmVal(char nval);
- char getmVal() const;
- float getiVal() const;
- private:
- float value;
+public:
+ DelayCtl();
+ ~DelayCtl() {};
+ std::string getString() const;
+ void setmVal(char nval);
+ char getmVal() const;
+ float getiVal() const;
+private:
+ float value;
};
#endif
diff --git a/src/DSP/AnalogFilter.C b/src/DSP/AnalogFilter.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
AnalogFilter.C - Several analog filters (lowpass, highpass...)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,11 +24,14 @@
#include <stdio.h>
#include "AnalogFilter.h"
-AnalogFilter::AnalogFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages){
+AnalogFilter::AnalogFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages)
+{
stages=Fstages;
- for (int i=0;i<3;i++){
- oldc[i]=0.0;oldd[i]=0.0;
- c[i]=0.0;d[i]=0.0;
+ for (int i=0;i<3;i++) {
+ oldc[i]=0.0;
+ oldd[i]=0.0;
+ c[i]=0.0;
+ d[i]=0.0;
};
type=Ftype;
freq=Ffreq;
@@ -37,27 +40,33 @@ AnalogFilter::AnalogFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsig
if (stages>=MAX_FILTER_STAGES) stages=MAX_FILTER_STAGES;
cleanup();
firsttime=0;
- abovenq=0;oldabovenq=0;
+ abovenq=0;
+ oldabovenq=0;
setfreq_and_q(Ffreq,Fq);
firsttime=1;
d[0]=0;//this is not used
outgain=1.0;
};
-AnalogFilter::~AnalogFilter(){
+AnalogFilter::~AnalogFilter()
+{
};
-void AnalogFilter::cleanup(){
- for (int i=0;i<MAX_FILTER_STAGES+1;i++){
- x[i].c1=0.0;x[i].c2=0.0;
- y[i].c1=0.0;y[i].c2=0.0;
- oldx[i]=x[i];
- oldy[i]=y[i];
+void AnalogFilter::cleanup()
+{
+ for (int i=0;i<MAX_FILTER_STAGES+1;i++) {
+ x[i].c1=0.0;
+ x[i].c2=0.0;
+ y[i].c1=0.0;
+ y[i].c2=0.0;
+ oldx[i]=x[i];
+ oldy[i]=y[i];
};
needsinterpolation=0;
};
-void AnalogFilter::computefiltercoefs(){
+void AnalogFilter::computefiltercoefs()
+{
REALTYPE tmp;
REALTYPE omega,sn,cs,alpha,beta;
int zerocoefs=0;//this is used if the freq is too high
@@ -65,199 +74,230 @@ void AnalogFilter::computefiltercoefs(){
//do not allow frequencies bigger than samplerate/2
REALTYPE freq=this->freq;
if (freq>(SAMPLE_RATE/2-500.0)) {
- freq=SAMPLE_RATE/2-500.0;
- zerocoefs=1;
+ freq=SAMPLE_RATE/2-500.0;
+ zerocoefs=1;
};
if (freq<0.1) freq=0.1;
//do not allow bogus Q
if (q<0.0) q=0.0;
REALTYPE tmpq,tmpgain;
if (stages==0) {
- tmpq=q;
- tmpgain=gain;
+ tmpq=q;
+ tmpgain=gain;
} else {
tmpq=(q>1.0 ? pow(q,1.0/(stages+1)) : q);
- tmpgain=pow(gain,1.0/(stages+1));
+ tmpgain=pow(gain,1.0/(stages+1));
};
-
- //most of theese are implementations of
+
+ //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==0) tmp=exp(-2.0*PI*freq/SAMPLE_RATE);
- else tmp=0.0;
- c[0]=1.0-tmp;c[1]=0.0;c[2]=0.0;
- d[1]=tmp;d[2]=0.0;
- order=1;
- break;
- case 1://HPF 1 pole
- if (zerocoefs==0) tmp=exp(-2.0*PI*freq/SAMPLE_RATE);
- else tmp=0.0;
- c[0]=(1.0+tmp)/2.0;c[1]=-(1.0+tmp)/2.0;c[2]=0.0;
- d[1]=tmp;d[2]=0.0;
- order=1;
- break;
- case 2://LPF 2 poles
- if (zerocoefs==0){
- omega=2*PI*freq/SAMPLE_RATE;
- sn=sin(omega);
- cs=cos(omega);
- alpha=sn/(2*tmpq);
- tmp=1+alpha;
- c[0]=(1.0-cs)/2.0/tmp;
- c[1]=(1.0-cs)/tmp;
- c[2]=(1.0-cs)/2.0/tmp;
- d[1]=-2*cs/tmp*(-1);
- d[2]=(1-alpha)/tmp*(-1);
- } else {
- c[0]=1.0;c[1]=0.0;c[2]=0.0;
- d[1]=0.0;d[2]=0.0;
- };
- order=2;
- break;
- case 3://HPF 2 poles
- if (zerocoefs==0){
- omega=2*PI*freq/SAMPLE_RATE;
- sn=sin(omega);
- cs=cos(omega);
- alpha=sn/(2*tmpq);
- tmp=1+alpha;
- c[0]=(1.0+cs)/2.0/tmp;
- c[1]=-(1.0+cs)/tmp;
- c[2]=(1.0+cs)/2.0/tmp;
- d[1]=-2*cs/tmp*(-1);
- d[2]=(1-alpha)/tmp*(-1);
- } else {
- c[0]=0.0;c[1]=0.0;c[2]=0.0;
- d[1]=0.0;d[2]=0.0;
- };
- order=2;
- break;
- case 4://BPF 2 poles
- if (zerocoefs==0){
- omega=2*PI*freq/SAMPLE_RATE;
- sn=sin(omega);
- cs=cos(omega);
- 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]=0.0;c[1]=0.0;c[2]=0.0;
- d[1]=0.0;d[2]=0.0;
- };
- order=2;
- break;
- case 5://NOTCH 2 poles
- if (zerocoefs==0){
- omega=2*PI*freq/SAMPLE_RATE;
- sn=sin(omega);
- cs=cos(omega);
- 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]=1.0;c[1]=0.0;c[2]=0.0;
- d[1]=0.0;d[2]=0.0;
- };
- order=2;
- break;
- case 6://PEAK (2 poles)
- if (zerocoefs==0){
- omega=2*PI*freq/SAMPLE_RATE;
- sn=sin(omega);
- cs=cos(omega);
- tmpq*=3.0;
- alpha=sn/(2*tmpq);
- tmp=1+alpha/tmpgain;
- c[0]=(1.0+alpha*tmpgain)/tmp;
- c[1]=(-2.0*cs)/tmp;
- c[2]=(1.0-alpha*tmpgain)/tmp;
- d[1]=-2*cs/tmp*(-1);
- d[2]=(1-alpha/tmpgain)/tmp*(-1);
- } else {
- c[0]=1.0;c[1]=0.0;c[2]=0.0;
- d[1]=0.0;d[2]=0.0;
- };
- order=2;
- break;
- case 7://Low Shelf - 2 poles
- if (zerocoefs==0){
- omega=2*PI*freq/SAMPLE_RATE;
- sn=sin(omega);
- cs=cos(omega);
- tmpq=sqrt(tmpq);
- alpha=sn/(2*tmpq);
- beta=sqrt(tmpgain)/tmpq;
- tmp=(tmpgain+1.0)+(tmpgain-1.0)*cs+beta*sn;
-
- c[0]=tmpgain*((tmpgain+1.0)-(tmpgain-1.0)*cs+beta*sn)/tmp;
- c[1]=2.0*tmpgain*((tmpgain-1.0)-(tmpgain+1.0)*cs)/tmp;
- c[2]=tmpgain*((tmpgain+1.0)-(tmpgain-1.0)*cs-beta*sn)/tmp;
- d[1]=-2.0*((tmpgain-1.0)+(tmpgain+1.0)*cs)/tmp*(-1);
- d[2]=((tmpgain+1.0)+(tmpgain-1.0)*cs-beta*sn)/tmp*(-1);
- } else {
- c[0]=tmpgain;c[1]=0.0;c[2]=0.0;
- d[1]=0.0;d[2]=0.0;
- };
- order=2;
- break;
- case 8://High Shelf - 2 poles
- if (zerocoefs==0){
- omega=2*PI*freq/SAMPLE_RATE;
- sn=sin(omega);
- cs=cos(omega);
- tmpq=sqrt(tmpq);
- alpha=sn/(2*tmpq);
- beta=sqrt(tmpgain)/tmpq;
- tmp=(tmpgain+1.0)-(tmpgain-1.0)*cs+beta*sn;
-
- c[0]=tmpgain*((tmpgain+1.0)+(tmpgain-1.0)*cs+beta*sn)/tmp;
- c[1]=-2.0*tmpgain*((tmpgain-1.0)+(tmpgain+1.0)*cs)/tmp;
- c[2]=tmpgain*((tmpgain+1.0)+(tmpgain-1.0)*cs-beta*sn)/tmp;
- d[1]=2.0*((tmpgain-1.0)-(tmpgain+1.0)*cs)/tmp*(-1);
- d[2]=((tmpgain+1.0)-(tmpgain-1.0)*cs-beta*sn)/tmp*(-1);
- } else {
- c[0]=1.0;c[1]=0.0;c[2]=0.0;
- d[1]=0.0;d[2]=0.0;
- };
- order=2;
- break;
- default://wrong type
- type=0;
- computefiltercoefs();
- break;
+ switch (type) {
+ case 0://LPF 1 pole
+ if (zerocoefs==0) tmp=exp(-2.0*PI*freq/SAMPLE_RATE);
+ else tmp=0.0;
+ c[0]=1.0-tmp;
+ c[1]=0.0;
+ c[2]=0.0;
+ d[1]=tmp;
+ d[2]=0.0;
+ order=1;
+ break;
+ case 1://HPF 1 pole
+ if (zerocoefs==0) tmp=exp(-2.0*PI*freq/SAMPLE_RATE);
+ else tmp=0.0;
+ c[0]=(1.0+tmp)/2.0;
+ c[1]=-(1.0+tmp)/2.0;
+ c[2]=0.0;
+ d[1]=tmp;
+ d[2]=0.0;
+ order=1;
+ break;
+ case 2://LPF 2 poles
+ if (zerocoefs==0) {
+ omega=2*PI*freq/SAMPLE_RATE;
+ sn=sin(omega);
+ cs=cos(omega);
+ alpha=sn/(2*tmpq);
+ tmp=1+alpha;
+ c[0]=(1.0-cs)/2.0/tmp;
+ c[1]=(1.0-cs)/tmp;
+ c[2]=(1.0-cs)/2.0/tmp;
+ d[1]=-2*cs/tmp*(-1);
+ d[2]=(1-alpha)/tmp*(-1);
+ } else {
+ c[0]=1.0;
+ c[1]=0.0;
+ c[2]=0.0;
+ d[1]=0.0;
+ d[2]=0.0;
+ };
+ order=2;
+ break;
+ case 3://HPF 2 poles
+ if (zerocoefs==0) {
+ omega=2*PI*freq/SAMPLE_RATE;
+ sn=sin(omega);
+ cs=cos(omega);
+ alpha=sn/(2*tmpq);
+ tmp=1+alpha;
+ c[0]=(1.0+cs)/2.0/tmp;
+ c[1]=-(1.0+cs)/tmp;
+ c[2]=(1.0+cs)/2.0/tmp;
+ d[1]=-2*cs/tmp*(-1);
+ d[2]=(1-alpha)/tmp*(-1);
+ } else {
+ c[0]=0.0;
+ c[1]=0.0;
+ c[2]=0.0;
+ d[1]=0.0;
+ d[2]=0.0;
+ };
+ order=2;
+ break;
+ case 4://BPF 2 poles
+ if (zerocoefs==0) {
+ omega=2*PI*freq/SAMPLE_RATE;
+ sn=sin(omega);
+ cs=cos(omega);
+ 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]=0.0;
+ c[1]=0.0;
+ c[2]=0.0;
+ d[1]=0.0;
+ d[2]=0.0;
+ };
+ order=2;
+ break;
+ case 5://NOTCH 2 poles
+ if (zerocoefs==0) {
+ omega=2*PI*freq/SAMPLE_RATE;
+ sn=sin(omega);
+ cs=cos(omega);
+ 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]=1.0;
+ c[1]=0.0;
+ c[2]=0.0;
+ d[1]=0.0;
+ d[2]=0.0;
+ };
+ order=2;
+ break;
+ case 6://PEAK (2 poles)
+ if (zerocoefs==0) {
+ omega=2*PI*freq/SAMPLE_RATE;
+ sn=sin(omega);
+ cs=cos(omega);
+ tmpq*=3.0;
+ alpha=sn/(2*tmpq);
+ tmp=1+alpha/tmpgain;
+ c[0]=(1.0+alpha*tmpgain)/tmp;
+ c[1]=(-2.0*cs)/tmp;
+ c[2]=(1.0-alpha*tmpgain)/tmp;
+ d[1]=-2*cs/tmp*(-1);
+ d[2]=(1-alpha/tmpgain)/tmp*(-1);
+ } else {
+ c[0]=1.0;
+ c[1]=0.0;
+ c[2]=0.0;
+ d[1]=0.0;
+ d[2]=0.0;
+ };
+ order=2;
+ break;
+ case 7://Low Shelf - 2 poles
+ if (zerocoefs==0) {
+ omega=2*PI*freq/SAMPLE_RATE;
+ sn=sin(omega);
+ cs=cos(omega);
+ tmpq=sqrt(tmpq);
+ alpha=sn/(2*tmpq);
+ beta=sqrt(tmpgain)/tmpq;
+ tmp=(tmpgain+1.0)+(tmpgain-1.0)*cs+beta*sn;
+
+ c[0]=tmpgain*((tmpgain+1.0)-(tmpgain-1.0)*cs+beta*sn)/tmp;
+ c[1]=2.0*tmpgain*((tmpgain-1.0)-(tmpgain+1.0)*cs)/tmp;
+ c[2]=tmpgain*((tmpgain+1.0)-(tmpgain-1.0)*cs-beta*sn)/tmp;
+ d[1]=-2.0*((tmpgain-1.0)+(tmpgain+1.0)*cs)/tmp*(-1);
+ d[2]=((tmpgain+1.0)+(tmpgain-1.0)*cs-beta*sn)/tmp*(-1);
+ } else {
+ c[0]=tmpgain;
+ c[1]=0.0;
+ c[2]=0.0;
+ d[1]=0.0;
+ d[2]=0.0;
+ };
+ order=2;
+ break;
+ case 8://High Shelf - 2 poles
+ if (zerocoefs==0) {
+ omega=2*PI*freq/SAMPLE_RATE;
+ sn=sin(omega);
+ cs=cos(omega);
+ tmpq=sqrt(tmpq);
+ alpha=sn/(2*tmpq);
+ beta=sqrt(tmpgain)/tmpq;
+ tmp=(tmpgain+1.0)-(tmpgain-1.0)*cs+beta*sn;
+
+ c[0]=tmpgain*((tmpgain+1.0)+(tmpgain-1.0)*cs+beta*sn)/tmp;
+ c[1]=-2.0*tmpgain*((tmpgain-1.0)+(tmpgain+1.0)*cs)/tmp;
+ c[2]=tmpgain*((tmpgain+1.0)+(tmpgain-1.0)*cs-beta*sn)/tmp;
+ d[1]=2.0*((tmpgain-1.0)-(tmpgain+1.0)*cs)/tmp*(-1);
+ d[2]=((tmpgain+1.0)-(tmpgain-1.0)*cs-beta*sn)/tmp*(-1);
+ } else {
+ c[0]=1.0;
+ c[1]=0.0;
+ c[2]=0.0;
+ d[1]=0.0;
+ d[2]=0.0;
+ };
+ order=2;
+ break;
+ default://wrong type
+ type=0;
+ computefiltercoefs();
+ break;
};
};
-void AnalogFilter::setfreq(REALTYPE frequency){
+void AnalogFilter::setfreq(REALTYPE frequency)
+{
if (frequency<0.1) frequency=0.1;
- REALTYPE rap=freq/frequency;if (rap<1.0) rap=1.0/rap;
-
- oldabovenq=abovenq;abovenq=frequency>(SAMPLE_RATE/2-500.0);
-
+ REALTYPE rap=freq/frequency;
+ if (rap<1.0) rap=1.0/rap;
+
+ oldabovenq=abovenq;
+ abovenq=frequency>(SAMPLE_RATE/2-500.0);
+
int nyquistthresh=(abovenq^oldabovenq);
- if ((rap>3.0)||(nyquistthresh!=0)){//if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
- for (int i=0;i<3;i++){
- oldc[i]=c[i];oldd[i]=d[i];
- };
- for (int i=0;i<MAX_FILTER_STAGES+1;i++){
- oldx[i]=x[i];
- oldy[i]=y[i];
- };
- if (firsttime==0) needsinterpolation=1;
+ if ((rap>3.0)||(nyquistthresh!=0)) {//if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
+ for (int i=0;i<3;i++) {
+ oldc[i]=c[i];
+ oldd[i]=d[i];
+ };
+ for (int i=0;i<MAX_FILTER_STAGES+1;i++) {
+ oldx[i]=x[i];
+ oldy[i]=y[i];
+ };
+ if (firsttime==0) needsinterpolation=1;
};
freq=frequency;
computefiltercoefs();
@@ -265,92 +305,101 @@ void AnalogFilter::setfreq(REALTYPE frequency){
};
-void AnalogFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
+void AnalogFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_)
+{
q=q_;
setfreq(frequency);
};
-void AnalogFilter::setq(REALTYPE q_){
+void AnalogFilter::setq(REALTYPE q_)
+{
q=q_;
computefiltercoefs();
};
-void AnalogFilter::settype(int type_){
+void AnalogFilter::settype(int type_)
+{
type=type_;
computefiltercoefs();
};
-void AnalogFilter::setgain(REALTYPE dBgain){
+void AnalogFilter::setgain(REALTYPE dBgain)
+{
gain=dB2rap(dBgain);
computefiltercoefs();
};
-void AnalogFilter::setstages(int stages_){
+void AnalogFilter::setstages(int stages_)
+{
if (stages_>=MAX_FILTER_STAGES) stages_=MAX_FILTER_STAGES-1;
stages=stages_;
cleanup();
computefiltercoefs();
};
-void AnalogFilter::singlefilterout(REALTYPE *smp,fstage &x,fstage &y,REALTYPE *c,REALTYPE *d){
+void AnalogFilter::singlefilterout(REALTYPE *smp,fstage &x,fstage &y,REALTYPE *c,REALTYPE *d)
+{
int i;
REALTYPE y0;
if (order==1) {//First order filter
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- y0=smp[i]*c[0]+x.c1*c[1]+y.c1*d[1];
- y.c1=y0;
- x.c1=smp[i];
- //output
- smp[i]=y0;
- };
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ y0=smp[i]*c[0]+x.c1*c[1]+y.c1*d[1];
+ y.c1=y0;
+ x.c1=smp[i];
+ //output
+ smp[i]=y0;
+ };
};
if (order==2) {//Second order filter
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- y0=smp[i]*c[0]+x.c1*c[1]+x.c2*c[2]+y.c1*d[1]+y.c2*d[2];
- y.c2=y.c1;
- y.c1=y0;
- x.c2=x.c1;
- x.c1=smp[i];
- //output
- smp[i]=y0;
- };
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ y0=smp[i]*c[0]+x.c1*c[1]+x.c2*c[2]+y.c1*d[1]+y.c2*d[2];
+ y.c2=y.c1;
+ y.c1=y0;
+ x.c2=x.c1;
+ x.c1=smp[i];
+ //output
+ smp[i]=y0;
+ };
};
};
-void AnalogFilter::filterout(REALTYPE *smp){
+void AnalogFilter::filterout(REALTYPE *smp)
+{
REALTYPE *ismp=NULL;//used if it needs interpolation
int i;
- if (needsinterpolation!=0){
- ismp=new REALTYPE[SOUND_BUFFER_SIZE];
- for (i=0;i<SOUND_BUFFER_SIZE;i++) ismp[i]=smp[i];
- for (i=0;i<stages+1;i++) singlefilterout(ismp,oldx[i],oldy[i],oldc,oldd);
+ if (needsinterpolation!=0) {
+ ismp=new REALTYPE[SOUND_BUFFER_SIZE];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) ismp[i]=smp[i];
+ for (i=0;i<stages+1;i++) singlefilterout(ismp,oldx[i],oldy[i],oldc,oldd);
};
-
+
for (i=0;i<stages+1;i++) singlefilterout(smp,x[i],y[i],c,d);
- if (needsinterpolation!=0){
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- REALTYPE x=i/(REALTYPE) SOUND_BUFFER_SIZE;
- smp[i]=ismp[i]*(1.0-x)+smp[i]*x;
- };
- delete [] ismp;
- needsinterpolation=0;
+ if (needsinterpolation!=0) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE x=i/(REALTYPE) SOUND_BUFFER_SIZE;
+ smp[i]=ismp[i]*(1.0-x)+smp[i]*x;
+ };
+ delete [] ismp;
+ needsinterpolation=0;
};
for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]*=outgain;
};
-REALTYPE AnalogFilter::H(REALTYPE freq){
+REALTYPE AnalogFilter::H(REALTYPE freq)
+{
REALTYPE fr=freq/SAMPLE_RATE*PI*2.0;
REALTYPE x=c[0],y=0.0;
- for (int n=1;n<3;n++){
- x+=cos(n*fr)*c[n];
- y-=sin(n*fr)*c[n];
+ for (int n=1;n<3;n++) {
+ x+=cos(n*fr)*c[n];
+ y-=sin(n*fr)*c[n];
};
REALTYPE h=x*x+y*y;
- x=1.0;y=0.0;
- for (int n=1;n<3;n++){
- x-=cos(n*fr)*d[n];
- y+=sin(n*fr)*d[n];
+ x=1.0;
+ y=0.0;
+ for (int n=1;n<3;n++) {
+ x-=cos(n*fr)*d[n];
+ y+=sin(n*fr)*d[n];
};
h=h/(x*x+y*y);
return(pow(h,(stages+1.0)/2.0));
diff --git a/src/DSP/AnalogFilter.h b/src/DSP/AnalogFilter.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Analog Filter.h - Several analog filters (lowpass, highpass...)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,10 +27,11 @@
#include "Filter_.h"
/**Implementation of Several analog filters (lowpass, highpass...)*/
-class AnalogFilter:public Filter_{
- public:
+class AnalogFilter:public Filter_
+{
+public:
AnalogFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages);
- ~AnalogFilter();
+ ~AnalogFilter();
void filterout(REALTYPE *smp);
void setfreq(REALTYPE frequency);
void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
@@ -40,14 +41,14 @@ class AnalogFilter:public Filter_{
void setgain(REALTYPE dBgain);
void setstages(int stages_);
void cleanup();
-
+
REALTYPE H(REALTYPE freq);//Obtains the response for a given frequency
-
- private:
- struct fstage{
- REALTYPE c1,c2;
+
+private:
+ struct fstage {
+ REALTYPE c1,c2;
} x[MAX_FILTER_STAGES+1],y[MAX_FILTER_STAGES+1],
- oldx[MAX_FILTER_STAGES+1],oldy[MAX_FILTER_STAGES+1];
+ oldx[MAX_FILTER_STAGES+1],oldy[MAX_FILTER_STAGES+1];
void singlefilterout(REALTYPE *smp,fstage &x,fstage &y,REALTYPE *c,REALTYPE *d);
void computefiltercoefs();
@@ -56,7 +57,7 @@ class AnalogFilter:public Filter_{
REALTYPE freq;//Frequency given in Hz
REALTYPE q; //Q factor (resonance or Q factor)
REALTYPE gain;//the gain of the filter (if are shelf/peak) filters
-
+
int order;//the order of the filter (number of poles)
REALTYPE c[3],d[3];//coefficients
diff --git a/src/DSP/FFTwrapper.C b/src/DSP/FFTwrapper.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
FFTwrapper.c - A wrapper for Fast Fourier Transforms
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -23,7 +23,8 @@
#include <math.h>
#include "FFTwrapper.h"
-FFTwrapper::FFTwrapper(int fftsize_){
+FFTwrapper::FFTwrapper(int fftsize_)
+{
fftsize=fftsize_;
tmpfftdata1=new fftw_real[fftsize];
tmpfftdata2=new fftw_real[fftsize];
@@ -36,11 +37,12 @@ FFTwrapper::FFTwrapper(int fftsize_){
#endif
};
-FFTwrapper::~FFTwrapper(){
+FFTwrapper::~FFTwrapper()
+{
#ifdef FFTW_VERSION_2
rfftw_destroy_plan(planfftw);
rfftw_destroy_plan(planfftw_inv);
-#else
+#else
fftw_destroy_plan(planfftw);
fftw_destroy_plan(planfftw_inv);
#endif
@@ -52,20 +54,21 @@ FFTwrapper::~FFTwrapper(){
/*
* do the Fast Fourier Transform
*/
-void FFTwrapper::smps2freqs(REALTYPE *smps,FFTFREQS freqs){
+void FFTwrapper::smps2freqs(REALTYPE *smps,FFTFREQS freqs)
+{
#ifdef FFTW_VERSION_2
for (int i=0;i<fftsize;i++) tmpfftdata1[i]=smps[i];
rfftw_one(planfftw,tmpfftdata1,tmpfftdata2);
for (int i=0;i<fftsize/2;i++) {
- freqs.c[i]=tmpfftdata2[i];
- if (i!=0) freqs.s[i]=tmpfftdata2[fftsize-i];
+ freqs.c[i]=tmpfftdata2[i];
+ if (i!=0) freqs.s[i]=tmpfftdata2[fftsize-i];
};
#else
for (int i=0;i<fftsize;i++) tmpfftdata1[i]=smps[i];
fftw_execute(planfftw);
for (int i=0;i<fftsize/2;i++) {
- freqs.c[i]=tmpfftdata1[i];
- if (i!=0) freqs.s[i]=tmpfftdata1[fftsize-i];
+ freqs.c[i]=tmpfftdata1[i];
+ if (i!=0) freqs.s[i]=tmpfftdata1[fftsize-i];
};
#endif
tmpfftdata2[fftsize/2]=0.0;
@@ -74,19 +77,20 @@ void FFTwrapper::smps2freqs(REALTYPE *smps,FFTFREQS freqs){
/*
* do the Inverse Fast Fourier Transform
*/
-void FFTwrapper::freqs2smps(FFTFREQS freqs,REALTYPE *smps){
+void FFTwrapper::freqs2smps(FFTFREQS freqs,REALTYPE *smps)
+{
tmpfftdata2[fftsize/2]=0.0;
#ifdef FFTW_VERSION_2
for (int i=0;i<fftsize/2;i++) {
- tmpfftdata1[i]=freqs.c[i];
- if (i!=0) tmpfftdata1[fftsize-i]=freqs.s[i];
+ tmpfftdata1[i]=freqs.c[i];
+ if (i!=0) tmpfftdata1[fftsize-i]=freqs.s[i];
};
rfftw_one(planfftw_inv,tmpfftdata1,tmpfftdata2);
for (int i=0;i<fftsize;i++) smps[i]=tmpfftdata2[i];
#else
for (int i=0;i<fftsize/2;i++) {
- tmpfftdata2[i]=freqs.c[i];
- if (i!=0) tmpfftdata2[fftsize-i]=freqs.s[i];
+ tmpfftdata2[i]=freqs.c[i];
+ if (i!=0) tmpfftdata2[fftsize-i]=freqs.s[i];
};
fftw_execute(planfftw_inv);
for (int i=0;i<fftsize;i++) smps[i]=tmpfftdata2[i];
diff --git a/src/DSP/FFTwrapper.h b/src/DSP/FFTwrapper.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
FFTwrapper.h - A wrapper for Fast Fourier Transforms
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -31,7 +31,7 @@
/* If you got error messages about rfftw.h, replace the next include line with "#include <srfftw.h>"
or with "#include <drfftw.h> (if one doesn't work try the other). It may be necessary to replace
-the <fftw.h> with <dfftw.h> or <sfftw.h>. If the neither one doesn't work,
+the <fftw.h> with <dfftw.h> or <sfftw.h>. If the neither one doesn't work,
please install latest version of fftw(recomanded from the sources) from www.fftw.org.
If you'll install fftw3 you need to change the Makefile.inc
Hope all goes right." */
@@ -45,22 +45,23 @@ Hope all goes right." */
#endif
/**A wrapper for the FFTW library (Fast Fourier Transforms)*/
-class FFTwrapper{
- public:
- /**Constructor
- * @param fftsize The size of samples to be fed to fftw*/
- FFTwrapper(int fftsize_);
- /**Destructor*/
- ~FFTwrapper();
- /**Convert Samples to Frequencies using Fourier Transform
- * @param smps Pointer to Samples to be converted; has length fftsize_
- * @param freqs Structure FFTFREQS which stores the frequencies*/
- void smps2freqs(REALTYPE *smps,FFTFREQS freqs);
- void freqs2smps(FFTFREQS freqs,REALTYPE *smps);
- private:
- int fftsize;
- fftw_real *tmpfftdata1,*tmpfftdata2;
- rfftw_plan planfftw,planfftw_inv;
+class FFTwrapper
+{
+public:
+ /**Constructor
+ * @param fftsize The size of samples to be fed to fftw*/
+ FFTwrapper(int fftsize_);
+ /**Destructor*/
+ ~FFTwrapper();
+ /**Convert Samples to Frequencies using Fourier Transform
+ * @param smps Pointer to Samples to be converted; has length fftsize_
+ * @param freqs Structure FFTFREQS which stores the frequencies*/
+ void smps2freqs(REALTYPE *smps,FFTFREQS freqs);
+ void freqs2smps(FFTFREQS freqs,REALTYPE *smps);
+private:
+ int fftsize;
+ fftw_real *tmpfftdata1,*tmpfftdata2;
+ rfftw_plan planfftw,planfftw_inv;
};
#endif
diff --git a/src/DSP/Filter.C b/src/DSP/Filter.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Filter.C - Filters, uses analog,formant,etc. filters
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,48 +25,58 @@
#include "Filter.h"
-Filter::Filter(FilterParams *pars){
+Filter::Filter(FilterParams *pars)
+{
unsigned char Ftype=pars->Ptype;
unsigned char Fstages=pars->Pstages;
-
+
category=pars->Pcategory;
-
+
switch (category) {
- case 1:filter=new FormantFilter(pars);
- break;
- case 2:filter=new SVFilter(Ftype,1000.0,pars->getq(),Fstages);
- filter->outgain=dB2rap(pars->getgain());
- if (filter->outgain>1.0) filter->outgain=sqrt(filter->outgain);
- break;
- default:filter=new AnalogFilter(Ftype,1000.0,pars->getq(),Fstages);
- if ((Ftype>=6)&&(Ftype<=8)) filter->setgain(pars->getgain());
- else filter->outgain=dB2rap(pars->getgain());
- break;
+ case 1:
+ filter=new FormantFilter(pars);
+ break;
+ case 2:
+ filter=new SVFilter(Ftype,1000.0,pars->getq(),Fstages);
+ filter->outgain=dB2rap(pars->getgain());
+ if (filter->outgain>1.0) filter->outgain=sqrt(filter->outgain);
+ break;
+ default:
+ filter=new AnalogFilter(Ftype,1000.0,pars->getq(),Fstages);
+ if ((Ftype>=6)&&(Ftype<=8)) filter->setgain(pars->getgain());
+ else filter->outgain=dB2rap(pars->getgain());
+ break;
};
};
-Filter::~Filter(){
+Filter::~Filter()
+{
delete (filter);
};
-void Filter::filterout(REALTYPE *smp){
+void Filter::filterout(REALTYPE *smp)
+{
filter->filterout(smp);
};
-void Filter::setfreq(REALTYPE frequency){
+void Filter::setfreq(REALTYPE frequency)
+{
filter->setfreq(frequency);
};
-void Filter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
+void Filter::setfreq_and_q(REALTYPE frequency,REALTYPE q_)
+{
filter->setfreq_and_q(frequency,q_);
};
-void Filter::setq(REALTYPE q_){
+void Filter::setq(REALTYPE q_)
+{
filter->setq(q_);
};
-REALTYPE Filter::getrealfreq(REALTYPE freqpitch){
+REALTYPE Filter::getrealfreq(REALTYPE freqpitch)
+{
if ((category==0)||(category==2)) return(pow(2.0,freqpitch+9.96578428));//log2(1000)=9.95748
- else return(freqpitch);
+ else return(freqpitch);
};
diff --git a/src/DSP/Filter.h b/src/DSP/Filter.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Filter.h - Filters, uses analog,formant,etc. filters
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -31,17 +31,18 @@
#include "SVFilter.h"
#include "../Params/FilterParams.h"
-class Filter{
- public:
+class Filter
+{
+public:
Filter(FilterParams *pars);
- ~Filter();
+ ~Filter();
void filterout(REALTYPE *smp);
void setfreq(REALTYPE frequency);
void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
void setq(REALTYPE q_);
REALTYPE getrealfreq(REALTYPE freqpitch);
- private:
+private:
Filter_ *filter;
unsigned char category;
};
diff --git a/src/DSP/Filter_.h b/src/DSP/Filter_.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Filter_.h - This class is inherited by filter classes
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,16 +25,17 @@
#include "../globals.h"
-class Filter_{
- public:
- virtual ~Filter_(){};
+class Filter_
+{
+public:
+ virtual ~Filter_() {};
virtual void filterout(REALTYPE *smp)=0;
virtual void setfreq(REALTYPE frequency)=0;
virtual void setfreq_and_q(REALTYPE frequency,REALTYPE q_)=0;
virtual void setq(REALTYPE q_)=0;
- virtual void setgain(REALTYPE dBgain){};
+ virtual void setgain(REALTYPE dBgain) {};
REALTYPE outgain;
- private:
+private:
};
diff --git a/src/DSP/FormantFilter.C b/src/DSP/FormantFilter.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
FormantFilter.C - formant filters
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,31 +24,33 @@
#include <stdio.h>
#include "FormantFilter.h"
-FormantFilter::FormantFilter(FilterParams *pars){
+FormantFilter::FormantFilter(FilterParams *pars)
+{
numformants=pars->Pnumformants;
for (int i=0;i<numformants;i++) formant[i]=new AnalogFilter(4/*BPF*/,1000.0,10.0,pars->Pstages);
cleanup();
inbuffer=new REALTYPE [SOUND_BUFFER_SIZE];
tmpbuf=new REALTYPE [SOUND_BUFFER_SIZE];
- for (int j=0;j<FF_MAX_VOWELS;j++)
- 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(pars->Pvowels[j].formants[i].amp);
- formantpar[j][i].q=pars->getformantq(pars->Pvowels[j].formants[i].q);
- };
+ for (int j=0;j<FF_MAX_VOWELS;j++)
+ 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(pars->Pvowels[j].formants[i].amp);
+ formantpar[j][i].q=pars->getformantq(pars->Pvowels[j].formants[i].q);
+ };
for (int i=0;i<FF_MAX_FORMANTS;i++) oldformantamp[i]=1.0;
- for (int i=0;i<numformants;i++){
- currentformants[i].freq=1000.0;
- currentformants[i].amp=1.0;
- currentformants[i].q=2.0;
+ for (int i=0;i<numformants;i++) {
+ currentformants[i].freq=1000.0;
+ currentformants[i].amp=1.0;
+ currentformants[i].q=2.0;
};
formantslowness=pow(1.0-(pars->Pformantslowness/128.0),3.0);
- sequencesize=pars->Psequencesize;if (sequencesize==0) sequencesize=1;
+ sequencesize=pars->Psequencesize;
+ if (sequencesize==0) sequencesize=1;
for (int k=0;k<sequencesize;k++) sequence[k].nvowel=pars->Psequence[k].nvowel;
-
+
vowelclearness=pow(10.0,(pars->Pvowelclearness-32.0)/48.0);
sequencestretch=pow(0.1,(pars->Psequencestretch-32.0)/48.0);
@@ -57,11 +59,13 @@ FormantFilter::FormantFilter(FilterParams *pars){
outgain=dB2rap(pars->getgain());
oldinput=-1.0;
- Qfactor=1.0;oldQfactor=Qfactor;
+ Qfactor=1.0;
+ oldQfactor=Qfactor;
firsttime=1;
};
-FormantFilter::~FormantFilter(){
+FormantFilter::~FormantFilter()
+{
for (int i=0;i<numformants;i++) delete(formant[i]);
delete[] inbuffer;
delete[] tmpbuf;
@@ -70,94 +74,103 @@ FormantFilter::~FormantFilter(){
-void FormantFilter::cleanup(){
+void FormantFilter::cleanup()
+{
for (int i=0;i<numformants;i++) formant[i]->cleanup();
};
-void FormantFilter::setpos(REALTYPE input){
+void FormantFilter::setpos(REALTYPE input)
+{
int p1,p2;
-
+
if (firsttime!=0) slowinput=input;
- else slowinput=slowinput*(1.0-formantslowness)+input*formantslowness;
+ else slowinput=slowinput*(1.0-formantslowness)+input*formantslowness;
if ((fabs(oldinput-input)<0.001)&&(fabs(slowinput-input)<0.001)&&
- (fabs(Qfactor-oldQfactor)<0.001)) {
+ (fabs(Qfactor-oldQfactor)<0.001)) {
// oldinput=input; daca setez asta, o sa faca probleme la schimbari foarte lente
- firsttime=0;
- return;
+ firsttime=0;
+ return;
} else oldinput=input;
- REALTYPE pos=fmod(input*sequencestretch,1.0);if (pos<0.0) pos+=1.0;
-
+ REALTYPE pos=fmod(input*sequencestretch,1.0);
+ if (pos<0.0) pos+=1.0;
+
F2I(pos*sequencesize,p2);
- p1=p2-1;if (p1<0) p1+=sequencesize;
+ p1=p2-1;
+ if (p1<0) p1+=sequencesize;
- pos=fmod(pos*sequencesize,1.0);
- if (pos<0.0) pos=0.0; else if (pos>1.0) pos=1.0;
+ pos=fmod(pos*sequencesize,1.0);
+ if (pos<0.0) pos=0.0;
+ else if (pos>1.0) pos=1.0;
pos=(atan((pos*2.0-1.0)*vowelclearness)/atan(vowelclearness)+1.0)*0.5;
p1=sequence[p1].nvowel;
p2=sequence[p2].nvowel;
-
+
if (firsttime!=0) {
- for (int i=0;i<numformants;i++){
- currentformants[i].freq=formantpar[p1][i].freq*(1.0-pos)+formantpar[p2][i].freq*pos;
- currentformants[i].amp=formantpar[p1][i].amp*(1.0-pos)+formantpar[p2][i].amp*pos;
- currentformants[i].q=formantpar[p1][i].q*(1.0-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;
+ for (int i=0;i<numformants;i++) {
+ currentformants[i].freq=formantpar[p1][i].freq*(1.0-pos)+formantpar[p2][i].freq*pos;
+ currentformants[i].amp=formantpar[p1][i].amp*(1.0-pos)+formantpar[p2][i].amp*pos;
+ currentformants[i].q=formantpar[p1][i].q*(1.0-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 {
- for (int i=0;i<numformants;i++){
- currentformants[i].freq=currentformants[i].freq*(1.0-formantslowness)
- +(formantpar[p1][i].freq*(1.0-pos)+formantpar[p2][i].freq*pos)*formantslowness;
+ for (int i=0;i<numformants;i++) {
+ currentformants[i].freq=currentformants[i].freq*(1.0-formantslowness)
+ +(formantpar[p1][i].freq*(1.0-pos)+formantpar[p2][i].freq*pos)*formantslowness;
- currentformants[i].amp=currentformants[i].amp*(1.0-formantslowness)
- +(formantpar[p1][i].amp*(1.0-pos)+formantpar[p2][i].amp*pos)*formantslowness;
+ currentformants[i].amp=currentformants[i].amp*(1.0-formantslowness)
+ +(formantpar[p1][i].amp*(1.0-pos)+formantpar[p2][i].amp*pos)*formantslowness;
- currentformants[i].q=currentformants[i].q*(1.0-formantslowness)
- +(formantpar[p1][i].q*(1.0-pos)+formantpar[p2][i].q*pos)*formantslowness;
+ currentformants[i].q=currentformants[i].q*(1.0-formantslowness)
+ +(formantpar[p1][i].q*(1.0-pos)+formantpar[p2][i].q*pos)*formantslowness;
- formant[i]->setfreq_and_q(currentformants[i].freq,currentformants[i].q*Qfactor);
- };
+ formant[i]->setfreq_and_q(currentformants[i].freq,currentformants[i].q*Qfactor);
+ };
};
-
+
oldQfactor=Qfactor;
};
-void FormantFilter::setfreq(REALTYPE frequency){
+void FormantFilter::setfreq(REALTYPE frequency)
+{
setpos(frequency);
};
-void FormantFilter::setq(REALTYPE q_){
+void FormantFilter::setq(REALTYPE q_)
+{
Qfactor=q_;
for (int i=0;i<numformants;i++) formant[i]->setq(Qfactor*currentformants[i].q);
};
-void FormantFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
- Qfactor=q_;
+void FormantFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_)
+{
+ Qfactor=q_;
setpos(frequency);
};
-void FormantFilter::filterout(REALTYPE *smp){
+void FormantFilter::filterout(REALTYPE *smp)
+{
int i,j;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- inbuffer[i]=smp[i];
- smp[i]=0.0;
+ inbuffer[i]=smp[i];
+ smp[i]=0.0;
};
-
+
for (j=0;j<numformants;j++) {
for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpbuf[i]=inbuffer[i]*outgain;
- formant[j]->filterout(tmpbuf);
+ formant[j]->filterout(tmpbuf);
- if (ABOVE_AMPLITUDE_THRESHOLD(oldformantamp[j],currentformants[j].amp))
- for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]+=tmpbuf[i]*
- INTERPOLATE_AMPLITUDE(oldformantamp[j],currentformants[j].amp,i,SOUND_BUFFER_SIZE);
- else for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]+=tmpbuf[i]*currentformants[j].amp;
- oldformantamp[j]=currentformants[j].amp;
+ if (ABOVE_AMPLITUDE_THRESHOLD(oldformantamp[j],currentformants[j].amp))
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]+=tmpbuf[i]*
+ INTERPOLATE_AMPLITUDE(oldformantamp[j],currentformants[j].amp,i,SOUND_BUFFER_SIZE);
+ else for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]+=tmpbuf[i]*currentformants[j].amp;
+ oldformantamp[j]=currentformants[j].amp;
};
};
diff --git a/src/DSP/FormantFilter.h b/src/DSP/FormantFilter.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
FormantFilter.h - formant filter
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,37 +29,38 @@
#include "../Params/FilterParams.h"
-class FormantFilter:public Filter_{
- public:
+class FormantFilter:public Filter_
+{
+public:
FormantFilter(FilterParams *pars);
- ~FormantFilter();
+ ~FormantFilter();
void filterout(REALTYPE *smp);
void setfreq(REALTYPE frequency);
void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
void setq(REALTYPE q_);
void cleanup();
- private:
+private:
AnalogFilter *formant[FF_MAX_FORMANTS];
REALTYPE *inbuffer,*tmpbuf;
struct {
- REALTYPE freq,amp,q;//frequency,amplitude,Q
+ REALTYPE freq,amp,q;//frequency,amplitude,Q
} formantpar[FF_MAX_VOWELS][FF_MAX_FORMANTS],currentformants[FF_MAX_FORMANTS];
struct {
- unsigned char nvowel;
+ unsigned char nvowel;
} sequence [FF_MAX_SEQUENCE];
-
+
REALTYPE oldformantamp[FF_MAX_FORMANTS];
-
+
int sequencesize,numformants,firsttime;
REALTYPE oldinput,slowinput;
REALTYPE Qfactor,formantslowness,oldQfactor;
REALTYPE vowelclearness,sequencestretch;
-
+
void setpos(REALTYPE input);
-
+
};
diff --git a/src/DSP/SVFilter.C b/src/DSP/SVFilter.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
SVFilter.C - Several state-variable filters
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,7 +24,8 @@
#include <stdio.h>
#include "SVFilter.h"
-SVFilter::SVFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages){
+SVFilter::SVFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages)
+{
stages=Fstages;
type=Ftype;
freq=Ffreq;
@@ -38,19 +39,24 @@ SVFilter::SVFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char
setfreq_and_q(Ffreq,Fq);
};
-SVFilter::~SVFilter(){
+SVFilter::~SVFilter()
+{
};
-void SVFilter::cleanup(){
- for (int i=0;i<MAX_FILTER_STAGES+1;i++){
- st[i].low=0.0;st[i].high=0.0;
- st[i].band=0.0;st[i].notch=0.0;
+void SVFilter::cleanup()
+{
+ for (int i=0;i<MAX_FILTER_STAGES+1;i++) {
+ st[i].low=0.0;
+ st[i].high=0.0;
+ st[i].band=0.0;
+ st[i].notch=0.0;
};
oldabovenq=0;
abovenq=0;
};
-void SVFilter::computefiltercoefs(){
+void SVFilter::computefiltercoefs()
+{
par.f=freq / SAMPLE_RATE*4.0;
if (par.f>0.99999) par.f=0.99999;
par.q=1.0-atan(sqrt(q))*2.0/PI;
@@ -59,18 +65,21 @@ void SVFilter::computefiltercoefs(){
};
-void SVFilter::setfreq(REALTYPE frequency){
+void SVFilter::setfreq(REALTYPE frequency)
+{
if (frequency<0.1) frequency=0.1;
- REALTYPE rap=freq/frequency;if (rap<1.0) rap=1.0/rap;
-
- oldabovenq=abovenq;abovenq=frequency>(SAMPLE_RATE/2-500.0);
-
+ REALTYPE rap=freq/frequency;
+ if (rap<1.0) rap=1.0/rap;
+
+ oldabovenq=abovenq;
+ abovenq=frequency>(SAMPLE_RATE/2-500.0);
+
int nyquistthresh=(abovenq^oldabovenq);
- if ((rap>3.0)||(nyquistthresh!=0)){//if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
- if (firsttime==0) needsinterpolation=1;
- ipar=par;
+ if ((rap>3.0)||(nyquistthresh!=0)) {//if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
+ if (firsttime==0) needsinterpolation=1;
+ ipar=par;
};
freq=frequency;
computefiltercoefs();
@@ -78,72 +87,87 @@ void SVFilter::setfreq(REALTYPE frequency){
};
-void SVFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_){
+void SVFilter::setfreq_and_q(REALTYPE frequency,REALTYPE q_)
+{
q=q_;
setfreq(frequency);
};
-void SVFilter::setq(REALTYPE q_){
+void SVFilter::setq(REALTYPE q_)
+{
q=q_;
computefiltercoefs();
};
-void SVFilter::settype(int type_){
+void SVFilter::settype(int type_)
+{
type=type_;
computefiltercoefs();
};
-void SVFilter::setgain(REALTYPE dBgain){
+void SVFilter::setgain(REALTYPE dBgain)
+{
gain=dB2rap(dBgain);
computefiltercoefs();
};
-void SVFilter::setstages(int stages_){
+void SVFilter::setstages(int stages_)
+{
if (stages_>=MAX_FILTER_STAGES) stages_=MAX_FILTER_STAGES-1;
stages=stages_;
cleanup();
computefiltercoefs();
};
-void SVFilter::singlefilterout(REALTYPE *smp,fstage &x,parameters &par){
+void SVFilter::singlefilterout(REALTYPE *smp,fstage &x,parameters &par)
+{
int i;
REALTYPE *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;
+ 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;
};
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- x.low = x.low + par.f * x.band;
- x.high = par.q_sqrt * smp[i] - x.low - par.q*x.band;
- x.band = par.f * x.high + x.band;
- x.notch = x.high + x.low;
-
- smp[i]= *out;
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ x.low = x.low + par.f * x.band;
+ x.high = par.q_sqrt * smp[i] - x.low - par.q*x.band;
+ x.band = par.f * x.high + x.band;
+ x.notch = x.high + x.low;
+
+ smp[i]= *out;
};
};
-void SVFilter::filterout(REALTYPE *smp){
+void SVFilter::filterout(REALTYPE *smp)
+{
int i;
REALTYPE *ismp=NULL;
-
- if (needsinterpolation!=0){
- ismp=new REALTYPE[SOUND_BUFFER_SIZE];
- for (i=0;i<SOUND_BUFFER_SIZE;i++) ismp[i]=smp[i];
- for (i=0;i<stages+1;i++) singlefilterout(ismp,st[i],ipar);
+
+ if (needsinterpolation!=0) {
+ ismp=new REALTYPE[SOUND_BUFFER_SIZE];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) ismp[i]=smp[i];
+ for (i=0;i<stages+1;i++) singlefilterout(ismp,st[i],ipar);
};
for (i=0;i<stages+1;i++) singlefilterout(smp,st[i],par);
- if (needsinterpolation!=0){
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- REALTYPE x=i/(REALTYPE) SOUND_BUFFER_SIZE;
- smp[i]=ismp[i]*(1.0-x)+smp[i]*x;
- };
- delete [] ismp;
- needsinterpolation=0;
+ if (needsinterpolation!=0) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE x=i/(REALTYPE) SOUND_BUFFER_SIZE;
+ smp[i]=ismp[i]*(1.0-x)+smp[i]*x;
+ };
+ delete [] ismp;
+ needsinterpolation=0;
};
for (i=0;i<SOUND_BUFFER_SIZE;i++) smp[i]*=outgain;
diff --git a/src/DSP/SVFilter.h b/src/DSP/SVFilter.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
SV Filter.h - Several state-variable filters
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,10 +25,11 @@
#include "../globals.h"
#include "Filter_.h"
-class SVFilter:public Filter_{
- public:
+class SVFilter:public Filter_
+{
+public:
SVFilter(unsigned char Ftype,REALTYPE Ffreq, REALTYPE Fq,unsigned char Fstages);
- ~SVFilter();
+ ~SVFilter();
void filterout(REALTYPE *smp);
void setfreq(REALTYPE frequency);
void setfreq_and_q(REALTYPE frequency,REALTYPE q_);
@@ -39,13 +40,13 @@ class SVFilter:public Filter_{
void setstages(int stages_);
void cleanup();
- private:
- struct fstage{
- REALTYPE low,high,band,notch;
+private:
+ struct fstage {
+ REALTYPE low,high,band,notch;
} st[MAX_FILTER_STAGES+1];
- struct parameters{
- REALTYPE f,q,q_sqrt;
+ struct parameters {
+ REALTYPE f,q,q_sqrt;
}par,ipar;
@@ -56,7 +57,7 @@ class SVFilter:public Filter_{
REALTYPE freq;//Frequency given in Hz
REALTYPE q; //Q factor (resonance or Q factor)
REALTYPE gain;//the gain of the filter (if are shelf/peak) filters
-
+
int abovenq;//this is 1 if the frequency is above the nyquist
int oldabovenq;
int needsinterpolation,firsttime;
diff --git a/src/Effects/Alienwah.C b/src/Effects/Alienwah.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Alienwah.C - "AlienWah" effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,7 +24,7 @@
#include "Alienwah.h"
Alienwah::Alienwah(const int &insertion_,REALTYPE *const efxoutl_,REALTYPE *const efxoutr_)
- :Effect(insertion_,efxoutl_,efxoutr_,NULL,0),oldl(NULL),oldr(NULL)
+ :Effect(insertion_,efxoutl_,efxoutr_,NULL,0),oldl(NULL),oldr(NULL)
{
setpreset(Ppreset);
cleanup();
@@ -32,7 +32,8 @@ Alienwah::Alienwah(const int &insertion_,REALTYPE *const efxoutl_,REALTYPE *cons
oldclfor=complex<REALTYPE>(fb,0.0);
};
-Alienwah::~Alienwah(){
+Alienwah::~Alienwah()
+{
if (oldl!=NULL) delete [] oldl;
if (oldr!=NULL) delete [] oldr ;
};
@@ -41,7 +42,8 @@ Alienwah::~Alienwah(){
/*
* Apply the effect
*/
-void Alienwah::out(REALTYPE *smpsl,REALTYPE *smpsr){
+void Alienwah::out(REALTYPE *smpsl,REALTYPE *smpsr)
+{
REALTYPE lfol,lfor; //Left/Right LFOs
complex<REALTYPE> clfol,clfor,out,tmp;
/**\todo Rework, as optimization can be used when the new complex type is
@@ -49,36 +51,37 @@ void Alienwah::out(REALTYPE *smpsl,REALTYPE *smpsr){
* Before all calculations needed to be done with individual REALTYPE,
* but now they can be done together*/
lfo.effectlfoout(&lfol,&lfor);
- lfol*=depth*PI*2.0;lfor*=depth*PI*2.0;
+ lfol*=depth*PI*2.0;
+ lfor*=depth*PI*2.0;
clfol=complex<REALTYPE>(cos(lfol+phase)*fb,sin(lfol+phase)*fb); //rework
clfor=complex<REALTYPE>(cos(lfor+phase)*fb,sin(lfor+phase)*fb); //rework
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- REALTYPE x=((REALTYPE) i)/SOUND_BUFFER_SIZE;
- REALTYPE x1=1.0-x;
- //left
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE x=((REALTYPE) i)/SOUND_BUFFER_SIZE;
+ REALTYPE x1=1.0-x;
+ //left
tmp=clfol*x+oldclfol*x1;
out=tmp*oldl[oldk];
out.real()+=(1-fabs(fb))*smpsr[i]*(1.0-panning);
-
+
oldl[oldk]=out;
- REALTYPE l=out.real()*10.0*(fb+0.1);
-
- //right
+ REALTYPE l=out.real()*10.0*(fb+0.1);
+
+ //right
tmp=clfor*x+oldclfor*x1;
-
+
out=tmp*oldr[oldk];
out.real()+=(1-fabs(fb))*smpsr[i]*(1.0-panning);
-
+
oldr[oldk]=out;
- REALTYPE r=out.real()*10.0*(fb+0.1);
+ REALTYPE r=out.real()*10.0*(fb+0.1);
- if (++oldk>=Pdelay) oldk=0;
- //LRcross
- efxoutl[i]=l*(1.0-lrcross)+r*lrcross;
- efxoutr[i]=r*(1.0-lrcross)+l*lrcross;
+ if (++oldk>=Pdelay) oldk=0;
+ //LRcross
+ efxoutl[i]=l*(1.0-lrcross)+r*lrcross;
+ efxoutr[i]=r*(1.0-lrcross)+l*lrcross;
};
oldclfol=clfol;
@@ -89,10 +92,11 @@ void Alienwah::out(REALTYPE *smpsl,REALTYPE *smpsr){
/*
* Cleanup the effect
*/
-void Alienwah::cleanup(){
+void Alienwah::cleanup()
+{
for (int i=0;i<Pdelay;i++) {
- oldl[i]=complex<REALTYPE>(0.0,0.0);
- oldr[i]=complex<REALTYPE>(0.0,0.0);
+ oldl[i]=complex<REALTYPE>(0.0,0.0);
+ oldr[i]=complex<REALTYPE>(0.0,0.0);
};
oldk=0;
};
@@ -102,12 +106,14 @@ void Alienwah::cleanup(){
* Parameter control
*/
-void Alienwah::setdepth(const unsigned char &Pdepth){
+void Alienwah::setdepth(const unsigned char &Pdepth)
+{
this->Pdepth=Pdepth;
depth=(Pdepth/127.0);
};
-void Alienwah::setfb(const unsigned char &Pfb){
+void Alienwah::setfb(const unsigned char &Pfb)
+{
this->Pfb=Pfb;
fb=fabs((Pfb-64.0)/64.1);
fb=sqrt(fb);
@@ -115,29 +121,34 @@ void Alienwah::setfb(const unsigned char &Pfb){
if (Pfb<64) fb=-fb;
};
-void Alienwah::setvolume(const unsigned char &Pvolume){
+void Alienwah::setvolume(const unsigned char &Pvolume)
+{
this->Pvolume=Pvolume;
outvolume=Pvolume/127.0;
if (insertion==0) volume=1.0;
- else volume=outvolume;
+ else volume=outvolume;
};
-void Alienwah::setpanning(const unsigned char &Ppanning){
+void Alienwah::setpanning(const unsigned char &Ppanning)
+{
this->Ppanning=Ppanning;
panning=Ppanning/127.0;
};
-void Alienwah::setlrcross(const unsigned char &Plrcross){
+void Alienwah::setlrcross(const unsigned char &Plrcross)
+{
this->Plrcross=Plrcross;
lrcross=Plrcross/127.0;
};
-void Alienwah::setphase(const unsigned char &Pphase){
+void Alienwah::setphase(const unsigned char &Pphase)
+{
this->Pphase=Pphase;
phase=(Pphase-64.0)/64.0*PI;
};
-void Alienwah::setdelay(const unsigned char &Pdelay){
+void Alienwah::setdelay(const unsigned char &Pdelay)
+{
if (oldl!=NULL) delete [] oldl;
if (oldr!=NULL) delete [] oldr;
if (Pdelay>=MAX_ALIENWAH_DELAY) this->Pdelay=MAX_ALIENWAH_DELAY;
@@ -147,19 +158,21 @@ void Alienwah::setdelay(const unsigned char &Pdelay){
cleanup();
};
-void Alienwah::setpreset(unsigned char npreset){
+void Alienwah::setpreset(unsigned char npreset)
+{
const int PRESET_SIZE=11;
const int NUM_PRESETS=4;
unsigned char presets[NUM_PRESETS][PRESET_SIZE]={
- //AlienWah1
- {127,64,70,0,0,62,60,105,25,0,64},
- //AlienWah2
- {127,64,73,106,0,101,60,105,17,0,64},
- //AlienWah3
- {127,64,63,0,1,100,112,105,31,0,42},
- //AlienWah4
- {93,64,25,0,1,66,101,11,47,0,86}};
-
+ //AlienWah1
+ {127,64,70,0,0,62,60,105,25,0,64},
+ //AlienWah2
+ {127,64,73,106,0,101,60,105,17,0,64},
+ //AlienWah3
+ {127,64,63,0,1,100,112,105,31,0,42},
+ //AlienWah4
+ {93,64,25,0,1,66,101,11,47,0,86}
+ };
+
if (npreset>=NUM_PRESETS) npreset=NUM_PRESETS-1;
for (int n=0;n<PRESET_SIZE;n++) changepar(n,presets[npreset][n]);
if (insertion==0) changepar(0,presets[npreset][0]/2);//lower the volume if this is system effect
@@ -167,63 +180,88 @@ void Alienwah::setpreset(unsigned char npreset){
};
-void Alienwah::changepar(const int &npar,const unsigned char &value){
- switch(npar){
- case 0: setvolume(value);
- break;
- case 1: setpanning(value);
- break;
- case 2: lfo.Pfreq=value;
- lfo.updateparams();
- break;
- case 3: lfo.Prandomness=value;
- lfo.updateparams();
- break;
- case 4: lfo.PLFOtype=value;
- lfo.updateparams();
- break;
- case 5: lfo.Pstereo=value;
- lfo.updateparams();
- break;
- case 6: setdepth(value);
- break;
- case 7: setfb(value);
- break;
- case 8: setdelay(value);
- break;
- case 9: setlrcross(value);
- break;
- case 10:setphase(value);
- break;
+void Alienwah::changepar(const int &npar,const unsigned char &value)
+{
+ switch (npar) {
+ case 0:
+ setvolume(value);
+ break;
+ case 1:
+ setpanning(value);
+ break;
+ case 2:
+ lfo.Pfreq=value;
+ lfo.updateparams();
+ break;
+ case 3:
+ lfo.Prandomness=value;
+ lfo.updateparams();
+ break;
+ case 4:
+ lfo.PLFOtype=value;
+ lfo.updateparams();
+ break;
+ case 5:
+ lfo.Pstereo=value;
+ lfo.updateparams();
+ break;
+ case 6:
+ setdepth(value);
+ break;
+ case 7:
+ setfb(value);
+ break;
+ case 8:
+ setdelay(value);
+ break;
+ case 9:
+ setlrcross(value);
+ break;
+ case 10:
+ setphase(value);
+ break;
};
};
-unsigned char Alienwah::getpar(const int &npar)const{
- switch (npar){
- case 0: return(Pvolume);
- break;
- case 1: return(Ppanning);
- break;
- case 2: return(lfo.Pfreq);
- break;
- case 3: return(lfo.Prandomness);
- break;
- case 4: return(lfo.PLFOtype);
- break;
- case 5: return(lfo.Pstereo);
- break;
- case 6: return(Pdepth);
- break;
- case 7: return(Pfb);
- break;
- case 8: return(Pdelay);
- break;
- case 9: return(Plrcross);
- break;
- case 10:return(Pphase);
- break;
- default:return (0);
+unsigned char Alienwah::getpar(const int &npar)const
+{
+ switch (npar) {
+ case 0:
+ return(Pvolume);
+ break;
+ case 1:
+ return(Ppanning);
+ break;
+ case 2:
+ return(lfo.Pfreq);
+ break;
+ case 3:
+ return(lfo.Prandomness);
+ break;
+ case 4:
+ return(lfo.PLFOtype);
+ break;
+ case 5:
+ return(lfo.Pstereo);
+ break;
+ case 6:
+ return(Pdepth);
+ break;
+ case 7:
+ return(Pfb);
+ break;
+ case 8:
+ return(Pdelay);
+ break;
+ case 9:
+ return(Plrcross);
+ break;
+ case 10:
+ return(Pphase);
+ break;
+ default:
+ return (0);
};
-
+
};
diff --git a/src/Effects/Alienwah.h b/src/Effects/Alienwah.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Alienwah.h - "AlienWah" effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -32,50 +32,51 @@ using namespace std;
#define MAX_ALIENWAH_DELAY 100
/**"AlienWah" Effect*/
-class Alienwah:public Effect {
- public:
- /**
- * Constructor
- * @param insetion_ 1 for insertion Effect, 0 for others
- * @param efxoutl_ Pointer to Alienwah's left channel output buffer
- * @param efxoutr_ Pointer to Alienwah's left channel output buffer
- * @return Initialized Alienwah
- */
- Alienwah(const int &insetion_,REALTYPE *const efxoutl_,REALTYPE *const efxoutr_);
- ~Alienwah();
- void out(REALTYPE *const smpsl,REALTYPE *const smpsr);
-
- void setpreset(unsigned char npreset);
- void changepar(const int &npar,const unsigned char &value);
- unsigned char getpar(const int &npar)const;
- void cleanup();
-
- private:
- //Alienwah Parameters
- EffectLFO lfo;//lfo-ul Alienwah
- unsigned char Pvolume;
- unsigned char Ppanning;
- unsigned char Pdepth;//the depth of the Alienwah
- unsigned char Pfb;//feedback
- unsigned char Plrcross;//feedback
- unsigned char Pdelay;
- unsigned char Pphase;
-
-
- //Control Parameters
- void setvolume(const unsigned char &Pvolume);
- void setpanning(const unsigned char &Ppanning);
- void setdepth(const unsigned char &Pdepth);
- void setfb(const unsigned char &Pfb);
- void setlrcross(const unsigned char &Plrcross);
- void setdelay(const unsigned char &Pdelay);
- void setphase(const unsigned char &Pphase);
-
- //Internal Values
- REALTYPE panning,fb,depth,lrcross,phase;
- complex<REALTYPE> *oldl,*oldr;
- complex<REALTYPE> oldclfol,oldclfor;
- int oldk;
+class Alienwah:public Effect
+{
+public:
+ /**
+ * Constructor
+ * @param insetion_ 1 for insertion Effect, 0 for others
+ * @param efxoutl_ Pointer to Alienwah's left channel output buffer
+ * @param efxoutr_ Pointer to Alienwah's left channel output buffer
+ * @return Initialized Alienwah
+ */
+ Alienwah(const int &insetion_,REALTYPE *const efxoutl_,REALTYPE *const efxoutr_);
+ ~Alienwah();
+ void out(REALTYPE *const smpsl,REALTYPE *const smpsr);
+
+ void setpreset(unsigned char npreset);
+ void changepar(const int &npar,const unsigned char &value);
+ unsigned char getpar(const int &npar)const;
+ void cleanup();
+
+private:
+ //Alienwah Parameters
+ EffectLFO lfo;//lfo-ul Alienwah
+ unsigned char Pvolume;
+ unsigned char Ppanning;
+ unsigned char Pdepth;//the depth of the Alienwah
+ unsigned char Pfb;//feedback
+ unsigned char Plrcross;//feedback
+ unsigned char Pdelay;
+ unsigned char Pphase;
+
+
+ //Control Parameters
+ void setvolume(const unsigned char &Pvolume);
+ void setpanning(const unsigned char &Ppanning);
+ void setdepth(const unsigned char &Pdepth);
+ void setfb(const unsigned char &Pfb);
+ void setlrcross(const unsigned char &Plrcross);
+ void setdelay(const unsigned char &Pdelay);
+ void setphase(const unsigned char &Pphase);
+
+ //Internal Values
+ REALTYPE panning,fb,depth,lrcross,phase;
+ complex<REALTYPE> *oldl,*oldr;
+ complex<REALTYPE> oldclfol,oldclfor;
+ int oldk;
};
#endif
diff --git a/src/Effects/Chorus.C b/src/Effects/Chorus.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Chorus.C - Chorus and Flange effects
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,11 +27,12 @@
using namespace std;
Chorus::Chorus(const int &insertion_,REALTYPE *const efxoutl_,REALTYPE *const efxoutr_)
- :Effect(insertion_,efxoutl_,efxoutr_,NULL,0),
- maxdelay((int)(MAX_CHORUS_DELAY/1000.0*SAMPLE_RATE)),
- delaySample(maxdelay)
+ :Effect(insertion_,efxoutl_,efxoutr_,NULL,0),
+ maxdelay((int)(MAX_CHORUS_DELAY/1000.0*SAMPLE_RATE)),
+ delaySample(maxdelay)
{
- dlk=0;drk=0;
+ dlk=0;
+ drk=0;
//maxdelay=(int)(MAX_CHORUS_DELAY/1000.0*SAMPLE_RATE);
//delayl=new REALTYPE[maxdelay];
//delayr=new REALTYPE[maxdelay];
@@ -44,22 +45,23 @@ Chorus::Chorus(const int &insertion_,REALTYPE *const efxoutl_,REALTYPE *const ef
cleanup();
};
-Chorus::~Chorus(){};
+Chorus::~Chorus() {};
/*
* get the delay value in samples; xlfo is the current lfo value
*/
-REALTYPE Chorus::getdelay(REALTYPE xlfo){
+REALTYPE Chorus::getdelay(REALTYPE xlfo)
+{
REALTYPE result;
- if (Pflangemode==0){
- result=(delay+xlfo*depth)*SAMPLE_RATE;
+ if (Pflangemode==0) {
+ result=(delay+xlfo*depth)*SAMPLE_RATE;
} else result=0;
-
+
//check if it is too big delay(caused bu errornous setdelay() and setdepth()
/**\todo fix setdelay() and setdepth(), so this error cannot occur*/
if ((result+0.5)>=maxdelay) {
- cerr << "WARNING: Chorus.C::getdelay(..) too big delay (see setdelay and setdepth funcs.)\n";
- result=maxdelay-1.0;
+ cerr << "WARNING: Chorus.C::getdelay(..) too big delay (see setdelay and setdepth funcs.)\n";
+ result=maxdelay-1.0;
};
return(result);
};
@@ -67,140 +69,151 @@ REALTYPE Chorus::getdelay(REALTYPE xlfo){
/*
* Apply the effect
*/
-void Chorus::out(REALTYPE *smpsl,REALTYPE *smpsr){
- const Stereo<AuSample> input(AuSample(smpsl,SOUND_BUFFER_SIZE),AuSample(smpsr,SOUND_BUFFER_SIZE));
+void Chorus::out(REALTYPE *smpsl,REALTYPE *smpsr)
+{
+ const Stereo<AuSample> input(AuSample(smpsl,SOUND_BUFFER_SIZE),AuSample(smpsr,SOUND_BUFFER_SIZE));
const REALTYPE one=1.0;
- dl1=dl2;dr1=dr2;
+ dl1=dl2;
+ dr1=dr2;
lfo.effectlfoout(&lfol,&lfor);
-
+
dl2=getdelay(lfol);
dr2=getdelay(lfor);
- for (int i=0;i<input.l().size();i++){
- REALTYPE inl=input.l()[i];
- REALTYPE inr=input.r()[i];
- //LRcross
- Stereo<REALTYPE> tmpc(inl,inr);
- //REALTYPE r=inr;
- inl=tmpc.l()*(1.0-lrcross)+tmpc.r()*lrcross;
- inr=tmpc.r()*(1.0-lrcross)+tmpc.l()*lrcross;
-
- //Left channel
-
- //compute the delay in samples using linear interpolation between the lfo delays
- mdel=(dl1*(SOUND_BUFFER_SIZE-i)+dl2*i)/SOUND_BUFFER_SIZE;
- if (++dlk>=maxdelay) dlk=0;
- REALTYPE tmp=dlk-mdel+maxdelay*2.0;//where should I get the sample from
-
- F2I(tmp,dlhi);
- dlhi%=maxdelay;
-
- dlhi2=(dlhi-1+maxdelay)%maxdelay;
- dllo=1.0-fmod(tmp,one);
+ for (int i=0;i<input.l().size();i++) {
+ REALTYPE inl=input.l()[i];
+ REALTYPE inr=input.r()[i];
+ //LRcross
+ Stereo<REALTYPE> tmpc(inl,inr);
+ //REALTYPE r=inr;
+ inl=tmpc.l()*(1.0-lrcross)+tmpc.r()*lrcross;
+ inr=tmpc.r()*(1.0-lrcross)+tmpc.l()*lrcross;
+
+ //Left channel
+
+ //compute the delay in samples using linear interpolation between the lfo delays
+ mdel=(dl1*(SOUND_BUFFER_SIZE-i)+dl2*i)/SOUND_BUFFER_SIZE;
+ if (++dlk>=maxdelay) dlk=0;
+ REALTYPE tmp=dlk-mdel+maxdelay*2.0;//where should I get the sample from
+
+ F2I(tmp,dlhi);
+ dlhi%=maxdelay;
+
+ dlhi2=(dlhi-1+maxdelay)%maxdelay;
+ dllo=1.0-fmod(tmp,one);
efxoutl[i]=delaySample.l()[dlhi2]*dllo+delaySample.l()[dlhi]*(1.0-dllo);
- delaySample.l()[dlk]=inl+efxoutl[i]*fb;
+ delaySample.l()[dlk]=inl+efxoutl[i]*fb;
- //Right channel
+ //Right channel
- //compute the delay in samples using linear interpolation between the lfo delays
- mdel=(dr1*(SOUND_BUFFER_SIZE-i)+dr2*i)/SOUND_BUFFER_SIZE;
- if (++drk>=maxdelay) drk=0;
- tmp=drk*1.0-mdel+maxdelay*2.0;//where should I get the sample from
+ //compute the delay in samples using linear interpolation between the lfo delays
+ mdel=(dr1*(SOUND_BUFFER_SIZE-i)+dr2*i)/SOUND_BUFFER_SIZE;
+ if (++drk>=maxdelay) drk=0;
+ tmp=drk*1.0-mdel+maxdelay*2.0;//where should I get the sample from
- F2I(tmp,dlhi);
- dlhi%=maxdelay;
-
- dlhi2=(dlhi-1+maxdelay)%maxdelay;
- dllo=1.0-fmod(tmp,one);
+ F2I(tmp,dlhi);
+ dlhi%=maxdelay;
+
+ dlhi2=(dlhi-1+maxdelay)%maxdelay;
+ dllo=1.0-fmod(tmp,one);
efxoutr[i]=delaySample.r()[dlhi2]*dllo+delaySample.r()[dlhi]*(1.0-dllo);
- delaySample.r()[dlk]=inr+efxoutr[i]*fb;
+ delaySample.r()[dlk]=inr+efxoutr[i]*fb;
};
if (Poutsub!=0)
- for (int i=0;i<input.l().size();i++){
- efxoutl[i] *= -1.0;
+ for (int i=0;i<input.l().size();i++) {
+ efxoutl[i] *= -1.0;
efxoutr[i] *= -1.0;
- };
+ };
- for (int i=0;i<input.l().size();i++){
- efxoutl[i]*=panning;
- efxoutr[i]*=(1.0-panning);
+ for (int i=0;i<input.l().size();i++) {
+ efxoutl[i]*=panning;
+ efxoutr[i]*=(1.0-panning);
};
};
/*
* Cleanup the effect
*/
-void Chorus::cleanup(){
+void Chorus::cleanup()
+{
delaySample.l().clear();
delaySample.r().clear();
//for (int i=0;i<maxdelay;i++){
// delayl[i]=0.0;
// delayr[i]=0.0;
//};
-
+
};
/*
* Parameter control
*/
-void Chorus::setdepth(const unsigned char &Pdepth){
+void Chorus::setdepth(const unsigned char &Pdepth)
+{
this->Pdepth=Pdepth;
depth=(pow(8.0,(Pdepth/127.0)*2.0)-1.0)/1000.0;//seconds
};
-void Chorus::setdelay(const unsigned char &Pdelay){
+void Chorus::setdelay(const unsigned char &Pdelay)
+{
this->Pdelay=Pdelay;
delay=(pow(10.0,(Pdelay/127.0)*2.0)-1.0)/1000.0;//seconds
};
-void Chorus::setfb(const unsigned char &Pfb){
+void Chorus::setfb(const unsigned char &Pfb)
+{
this->Pfb=Pfb;
fb=(Pfb-64.0)/64.1;
};
-void Chorus::setvolume(const unsigned char &Pvolume){
+void Chorus::setvolume(const unsigned char &Pvolume)
+{
this->Pvolume=Pvolume;
outvolume=Pvolume/127.0;
if (insertion==0) volume=1.0;
- else volume=outvolume;
+ else volume=outvolume;
};
-void Chorus::setpanning(const unsigned char &Ppanning){
+void Chorus::setpanning(const unsigned char &Ppanning)
+{
this->Ppanning=Ppanning;
panning=Ppanning/127.0;
};
-void Chorus::setlrcross(const unsigned char &Plrcross){
+void Chorus::setlrcross(const unsigned char &Plrcross)
+{
this->Plrcross=Plrcross;
lrcross=Plrcross/127.0;
};
-void Chorus::setpreset(unsigned char npreset){
+void Chorus::setpreset(unsigned char npreset)
+{
const int PRESET_SIZE=12;
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},
- //Chorus2
- {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},
- //Celeste1
- {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},
- //Flange1
- {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},
- //Flange3
- {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},
- //Flange5
- {64,64,55,105,0,24,39,19,17,0,0,1}};
+ //Chorus1
+ {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},
+ //Chorus3
+ {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},
+ //Celeste2
+ {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},
+ //Flange2
+ {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},
+ //Flange4
+ {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}
+ };
if (npreset>=NUM_PRESETS) npreset=NUM_PRESETS-1;
for (int n=0;n<PRESET_SIZE;n++) changepar(n,presets[npreset][n]);
@@ -208,69 +221,96 @@ void Chorus::setpreset(unsigned char npreset){
};
-void Chorus::changepar(const int &npar,const unsigned char &value){
- switch(npar){
- case 0: setvolume(value);
- break;
- case 1: setpanning(value);
- break;
- case 2: lfo.Pfreq=value;
- lfo.updateparams();
- break;
- case 3: lfo.Prandomness=value;
- lfo.updateparams();
- break;
- case 4: lfo.PLFOtype=value;
- lfo.updateparams();
- break;
- case 5: lfo.Pstereo=value;
- lfo.updateparams();
- break;
- case 6: setdepth(value);
- break;
- case 7: setdelay(value);
- break;
- case 8: setfb(value);
- break;
- case 9: setlrcross(value);
- break;
- case 10:if (value>1) Pflangemode=1;
- else Pflangemode=value;
- break;
- case 11:if (value>1) Poutsub=1;
- else Poutsub=value;
- break;
+void Chorus::changepar(const int &npar,const unsigned char &value)
+{
+ switch (npar) {
+ case 0:
+ setvolume(value);
+ break;
+ case 1:
+ setpanning(value);
+ break;
+ case 2:
+ lfo.Pfreq=value;
+ lfo.updateparams();
+ break;
+ case 3:
+ lfo.Prandomness=value;
+ lfo.updateparams();
+ break;
+ case 4:
+ lfo.PLFOtype=value;
+ lfo.updateparams();
+ break;
+ case 5:
+ lfo.Pstereo=value;
+ lfo.updateparams();
+ break;
+ case 6:
+ setdepth(value);
+ break;
+ case 7:
+ setdelay(value);
+ break;
+ case 8:
+ setfb(value);
+ break;
+ case 9:
+ setlrcross(value);
+ break;
+ case 10:
+ if (value>1) Pflangemode=1;
+ else Pflangemode=value;
+ break;
+ case 11:
+ if (value>1) Poutsub=1;
+ else Poutsub=value;
+ break;
};
};
-unsigned char Chorus::getpar(const int &npar)const{
- switch (npar){
- case 0: return(Pvolume);
- break;
- case 1: return(Ppanning);
- break;
- case 2: return(lfo.Pfreq);
- break;
- case 3: return(lfo.Prandomness);
- break;
- case 4: return(lfo.PLFOtype);
- break;
- case 5: return(lfo.Pstereo);
- break;
- case 6: return(Pdepth);
- break;
- case 7: return(Pdelay);
- break;
- case 8: return(Pfb);
- break;
- case 9: return(Plrcross);
- break;
- case 10:return(Pflangemode);
- break;
- case 11:return(Poutsub);
- break;
- default:return (0);
+unsigned char Chorus::getpar(const int &npar)const
+{
+ switch (npar) {
+ case 0:
+ return(Pvolume);
+ break;
+ case 1:
+ return(Ppanning);
+ break;
+ case 2:
+ return(lfo.Pfreq);
+ break;
+ case 3:
+ return(lfo.Prandomness);
+ break;
+ case 4:
+ return(lfo.PLFOtype);
+ break;
+ case 5:
+ return(lfo.Pstereo);
+ break;
+ case 6:
+ return(Pdepth);
+ break;
+ case 7:
+ return(Pdelay);
+ break;
+ case 8:
+ return(Pfb);
+ break;
+ case 9:
+ return(Plrcross);
+ break;
+ case 10:
+ return(Pflangemode);
+ break;
+ case 11:
+ return(Poutsub);
+ break;
+ default:
+ return (0);
};
-
+
};
diff --git a/src/Effects/Chorus.h b/src/Effects/Chorus.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Chorus.h - Chorus and Flange effects
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -31,83 +31,84 @@
#define MAX_CHORUS_DELAY 250.0 //ms
/**Chorus and Flange effects*/
-class Chorus:public Effect {
- public:
- Chorus(const int &insetion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
- /**Destructor*/
- ~Chorus();
- void out(REALTYPE *smpsl,REALTYPE *smpsr);
- void setpreset(unsigned char npreset);
- /**
- * Sets the value of the chosen variable
- *
- * The possible parameters are:
- * -# Volume
- * -# Panning
- * -# LFO Frequency
- * -# LFO Randomness
- * -# LFO Type
- * -# LFO stereo
- * -# Depth
- * -# Delay
- * -# Feedback
- * -# Flange Mode
- * -# Subtractive
- * @param npar number of chosen parameter
- * @param value the new value
- */
- void changepar(const int &npar,const unsigned char &value);
- /**
- * Gets the value of the chosen variable
- *
- * The possible parameters are:
- * -# Volume
- * -# Panning
- * -# LFO Frequency
- * -# LFO Randomness
- * -# LFO Type
- * -# LFO stereo
- * -# Depth
- * -# Delay
- * -# Feedback
- * -# Flange Mode
- * -# Subtractive
- * @param npar number of chosen parameter
- * @return the value of the parameter
- */
- unsigned char getpar(const int &npar)const;
- void cleanup();
-
- private:
- //Chorus Parameters
- EffectLFO lfo;//lfo-ul chorus
- unsigned char Pvolume;
- unsigned char Ppanning;
- unsigned char Pdepth;//the depth of the Chorus(ms)
- unsigned char Pdelay;//the delay (ms)
- unsigned char Pfb;//feedback
- unsigned char Plrcross;//feedback
- unsigned char Pflangemode;//how the LFO is scaled, to result chorus or flange
- unsigned char Poutsub;//if I wish to substract the output instead of the adding it
+class Chorus:public Effect
+{
+public:
+ Chorus(const int &insetion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
+ /**Destructor*/
+ ~Chorus();
+ void out(REALTYPE *smpsl,REALTYPE *smpsr);
+ void setpreset(unsigned char npreset);
+ /**
+ * Sets the value of the chosen variable
+ *
+ * The possible parameters are:
+ * -# Volume
+ * -# Panning
+ * -# LFO Frequency
+ * -# LFO Randomness
+ * -# LFO Type
+ * -# LFO stereo
+ * -# Depth
+ * -# Delay
+ * -# Feedback
+ * -# Flange Mode
+ * -# Subtractive
+ * @param npar number of chosen parameter
+ * @param value the new value
+ */
+ void changepar(const int &npar,const unsigned char &value);
+ /**
+ * Gets the value of the chosen variable
+ *
+ * The possible parameters are:
+ * -# Volume
+ * -# Panning
+ * -# LFO Frequency
+ * -# LFO Randomness
+ * -# LFO Type
+ * -# LFO stereo
+ * -# Depth
+ * -# Delay
+ * -# Feedback
+ * -# Flange Mode
+ * -# Subtractive
+ * @param npar number of chosen parameter
+ * @return the value of the parameter
+ */
+ unsigned char getpar(const int &npar)const;
+ void cleanup();
+
+private:
+ //Chorus Parameters
+ EffectLFO lfo;//lfo-ul chorus
+ unsigned char Pvolume;
+ unsigned char Ppanning;
+ unsigned char Pdepth;//the depth of the Chorus(ms)
+ unsigned char Pdelay;//the delay (ms)
+ unsigned char Pfb;//feedback
+ unsigned char Plrcross;//feedback
+ unsigned char Pflangemode;//how the LFO is scaled, to result chorus or flange
+ unsigned char Poutsub;//if I wish to substract the output instead of the adding it
+
-
- //Parameter Controls
- void setvolume(const unsigned char &Pvolume);
- void setpanning(const unsigned char &Ppanning);
- void setdepth(const unsigned char &Pdepth);
- void setdelay(const unsigned char &Pdelay);
- void setfb(const unsigned char &Pfb);
- void setlrcross(const unsigned char &Plrcross);
+ //Parameter Controls
+ void setvolume(const unsigned char &Pvolume);
+ void setpanning(const unsigned char &Ppanning);
+ void setdepth(const unsigned char &Pdepth);
+ void setdelay(const unsigned char &Pdelay);
+ void setfb(const unsigned char &Pfb);
+ void setlrcross(const unsigned char &Plrcross);
- //Internal Values
- REALTYPE depth,delay,fb,lrcross,panning;
- REALTYPE dl1,dl2,dr1,dr2,lfol,lfor;
- int maxdelay;
- Stereo<AuSample> delaySample;
- //REALTYPE *delayl,*delayr;
- int dlk,drk,dlhi,dlhi2;
- REALTYPE getdelay(REALTYPE xlfo);
- REALTYPE dllo,mdel;
+ //Internal Values
+ REALTYPE depth,delay,fb,lrcross,panning;
+ REALTYPE dl1,dl2,dr1,dr2,lfol,lfor;
+ int maxdelay;
+ Stereo<AuSample> delaySample;
+ //REALTYPE *delayl,*delayr;
+ int dlk,drk,dlhi,dlhi2;
+ REALTYPE getdelay(REALTYPE xlfo);
+ REALTYPE dllo,mdel;
};
#endif
diff --git a/src/Effects/Distorsion.C b/src/Effects/Distorsion.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Distorsion.C - Distorsion effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,132 +28,147 @@
* Waveshape (this is called by OscilGen::waveshape and Distorsion::process)
*/
-void waveshapesmps(int n,REALTYPE *smps,unsigned char type,unsigned char drive){
+void waveshapesmps(int n,REALTYPE *smps,unsigned char type,unsigned char drive)
+{
int i;
REALTYPE ws=drive/127.0;
REALTYPE tmpv;
- switch(type){
- case 1: ws=pow(10,ws*ws*3.0)-1.0+0.001;//Arctangent
- for (i=0;i<n;i++)
- smps[i]=atan(smps[i]*ws)/atan(ws);
- break;
- case 2: ws=ws*ws*32.0+0.0001;//Asymmetric
- if (ws<1.0) tmpv=sin(ws)+0.1;
- else tmpv=1.1;
- for (i=0;i<n;i++) {
- smps[i]=sin(smps[i]*(0.1+ws-ws*smps[i]))/tmpv;
- };
- break;
- case 3: ws=ws*ws*ws*20.0+0.0001;//Pow
- for (i=0;i<n;i++) {
- smps[i]*=ws;
- if (fabs(smps[i])<1.0) {
- smps[i]=(smps[i]-pow(smps[i],3.0))*3.0;
- if (ws<1.0) smps[i]/=ws;
- } else smps[i]=0.0;
- };
- break;
- case 4: ws=ws*ws*ws*32.0+0.0001;//Sine
- if (ws<1.57) tmpv=sin(ws);
- else tmpv=1.0;
- for (i=0;i<n;i++) smps[i]=sin(smps[i]*ws)/tmpv;
- break;
- case 5: ws=ws*ws+0.000001;//Quantisize
- for (i=0;i<n;i++)
- smps[i]=floor(smps[i]/ws+0.5)*ws;
- break;
- case 6: ws=ws*ws*ws*32+0.0001;//Zigzag
- if (ws<1.0) tmpv=sin(ws);
- else tmpv=1.0;
- for (i=0;i<n;i++)
- smps[i]=asin(sin(smps[i]*ws))/tmpv;
- break;
- case 7: ws=pow(2.0,-ws*ws*8.0); //Limiter
- for (i=0;i<n;i++) {
- REALTYPE tmp=smps[i];
- if (fabs(tmp)>ws) {
- if (tmp>=0.0) smps[i]=1.0;
- else smps[i]=-1.0;
- } else smps[i]/=ws;
- };
- break;
- case 8: ws=pow(2.0,-ws*ws*8.0); //Upper Limiter
- for (i=0;i<n;i++) {
- REALTYPE tmp=smps[i];
- if (tmp>ws) smps[i]=ws;
- smps[i]*=2.0;
- };
- break;
- case 9: ws=pow(2.0,-ws*ws*8.0); //Lower Limiter
- for (i=0;i<n;i++) {
- REALTYPE tmp=smps[i];
- if (tmp<-ws) smps[i]=-ws;
- smps[i]*=2.0;
- };
- break;
- case 10:ws=(pow(2.0,ws*6.0)-1.0)/pow(2.0,6.0); //Inverse Limiter
- for (i=0;i<n;i++) {
- REALTYPE tmp=smps[i];
- if (fabs(tmp)>ws) {
- if (tmp>=0.0) smps[i]=tmp-ws;
- else smps[i]=tmp+ws;
- } else smps[i]=0;
- };
- break;
- case 11:ws=pow(5,ws*ws*1.0)-1.0;//Clip
- for (i=0;i<n;i++)
- smps[i]=smps[i]*(ws+0.5)*0.9999-floor(0.5+smps[i]*(ws+0.5)*0.9999);
- break;
- case 12:ws=ws*ws*ws*30+0.001;//Asym2
- if (ws<0.3) tmpv=ws;
- else tmpv=1.0;
- for (i=0;i<n;i++) {
- REALTYPE tmp=smps[i]*ws;
- if ((tmp>-2.0) && (tmp<1.0)) smps[i]=tmp*(1.0-tmp)*(tmp+2.0)/tmpv;
- else smps[i]=0.0;
- };
- break;
- case 13:ws=ws*ws*ws*32.0+0.0001;//Pow2
- if (ws<1.0) tmpv=ws*(1+ws)/2.0;
- else tmpv=1.0;
- for (i=0;i<n;i++) {
- REALTYPE tmp=smps[i]*ws;
- if ((tmp>-1.0)&&(tmp<1.618034)) smps[i]=tmp*(1.0-tmp)/tmpv;
- else if (tmp>0.0) smps[i]=-1.0;
- else smps[i]=-2.0;
- };
- break;
- case 14:ws=pow(ws,5.0)*80.0+0.0001;//sigmoid
- if (ws>10.0) tmpv=0.5;
- else tmpv=0.5-1.0/(exp(ws)+1.0);
- for (i=0;i<n;i++) {
- REALTYPE tmp=smps[i]*ws;
- if (tmp<-10.0) tmp=-10.0;
- else if (tmp>10.0) tmp=10.0;
- tmp=0.5-1.0/(exp(tmp)+1.0);
- smps[i]=tmp/tmpv;
- };
- break;
- /**\todo update to Distorsion::changepar (Ptype max) if there is added more waveshapings functions*/
+ switch (type) {
+ case 1:
+ ws=pow(10,ws*ws*3.0)-1.0+0.001;//Arctangent
+ for (i=0;i<n;i++)
+ smps[i]=atan(smps[i]*ws)/atan(ws);
+ break;
+ case 2:
+ ws=ws*ws*32.0+0.0001;//Asymmetric
+ if (ws<1.0) tmpv=sin(ws)+0.1;
+ else tmpv=1.1;
+ for (i=0;i<n;i++) {
+ smps[i]=sin(smps[i]*(0.1+ws-ws*smps[i]))/tmpv;
+ };
+ break;
+ case 3:
+ ws=ws*ws*ws*20.0+0.0001;//Pow
+ for (i=0;i<n;i++) {
+ smps[i]*=ws;
+ if (fabs(smps[i])<1.0) {
+ smps[i]=(smps[i]-pow(smps[i],3.0))*3.0;
+ if (ws<1.0) smps[i]/=ws;
+ } else smps[i]=0.0;
+ };
+ break;
+ case 4:
+ ws=ws*ws*ws*32.0+0.0001;//Sine
+ if (ws<1.57) tmpv=sin(ws);
+ else tmpv=1.0;
+ for (i=0;i<n;i++) smps[i]=sin(smps[i]*ws)/tmpv;
+ break;
+ case 5:
+ ws=ws*ws+0.000001;//Quantisize
+ for (i=0;i<n;i++)
+ smps[i]=floor(smps[i]/ws+0.5)*ws;
+ break;
+ case 6:
+ ws=ws*ws*ws*32+0.0001;//Zigzag
+ if (ws<1.0) tmpv=sin(ws);
+ else tmpv=1.0;
+ for (i=0;i<n;i++)
+ smps[i]=asin(sin(smps[i]*ws))/tmpv;
+ break;
+ case 7:
+ ws=pow(2.0,-ws*ws*8.0); //Limiter
+ for (i=0;i<n;i++) {
+ REALTYPE tmp=smps[i];
+ if (fabs(tmp)>ws) {
+ if (tmp>=0.0) smps[i]=1.0;
+ else smps[i]=-1.0;
+ } else smps[i]/=ws;
+ };
+ break;
+ case 8:
+ ws=pow(2.0,-ws*ws*8.0); //Upper Limiter
+ for (i=0;i<n;i++) {
+ REALTYPE tmp=smps[i];
+ if (tmp>ws) smps[i]=ws;
+ smps[i]*=2.0;
+ };
+ break;
+ case 9:
+ ws=pow(2.0,-ws*ws*8.0); //Lower Limiter
+ for (i=0;i<n;i++) {
+ REALTYPE tmp=smps[i];
+ if (tmp<-ws) smps[i]=-ws;
+ smps[i]*=2.0;
+ };
+ break;
+ case 10:
+ ws=(pow(2.0,ws*6.0)-1.0)/pow(2.0,6.0); //Inverse Limiter
+ for (i=0;i<n;i++) {
+ REALTYPE tmp=smps[i];
+ if (fabs(tmp)>ws) {
+ if (tmp>=0.0) smps[i]=tmp-ws;
+ else smps[i]=tmp+ws;
+ } else smps[i]=0;
+ };
+ break;
+ case 11:
+ ws=pow(5,ws*ws*1.0)-1.0;//Clip
+ for (i=0;i<n;i++)
+ smps[i]=smps[i]*(ws+0.5)*0.9999-floor(0.5+smps[i]*(ws+0.5)*0.9999);
+ break;
+ case 12:
+ ws=ws*ws*ws*30+0.001;//Asym2
+ if (ws<0.3) tmpv=ws;
+ else tmpv=1.0;
+ for (i=0;i<n;i++) {
+ REALTYPE tmp=smps[i]*ws;
+ if ((tmp>-2.0) && (tmp<1.0)) smps[i]=tmp*(1.0-tmp)*(tmp+2.0)/tmpv;
+ else smps[i]=0.0;
+ };
+ break;
+ case 13:
+ ws=ws*ws*ws*32.0+0.0001;//Pow2
+ if (ws<1.0) tmpv=ws*(1+ws)/2.0;
+ else tmpv=1.0;
+ for (i=0;i<n;i++) {
+ REALTYPE tmp=smps[i]*ws;
+ if ((tmp>-1.0)&&(tmp<1.618034)) smps[i]=tmp*(1.0-tmp)/tmpv;
+ else if (tmp>0.0) smps[i]=-1.0;
+ else smps[i]=-2.0;
+ };
+ break;
+ case 14:
+ ws=pow(ws,5.0)*80.0+0.0001;//sigmoid
+ if (ws>10.0) tmpv=0.5;
+ else tmpv=0.5-1.0/(exp(ws)+1.0);
+ for (i=0;i<n;i++) {
+ REALTYPE tmp=smps[i]*ws;
+ if (tmp<-10.0) tmp=-10.0;
+ else if (tmp>10.0) tmp=10.0;
+ tmp=0.5-1.0/(exp(tmp)+1.0);
+ smps[i]=tmp/tmpv;
+ };
+ break;
+ /**\todo update to Distorsion::changepar (Ptype max) if there is added more waveshapings functions*/
};
};
Distorsion::Distorsion(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_)
- :Effect(insertion_,efxoutl_,efxoutr_,NULL,0)
+ :Effect(insertion_,efxoutl_,efxoutr_,NULL,0)
{
lpfl=new AnalogFilter(2,22000,1,0);
lpfr=new AnalogFilter(2,22000,1,0);
hpfl=new AnalogFilter(3,20,1,0);
hpfr=new AnalogFilter(3,20,1,0);
-
+
//default values
Pvolume=50;
- Plrcross=40;
+ Plrcross=40;
Pdrive=90;
Plevel=64;
Ptype=0;
@@ -163,22 +178,24 @@ Distorsion::Distorsion(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxout
Pstereo=0;
Pprefiltering=0;
- setpreset(Ppreset);
+ setpreset(Ppreset);
cleanup();
};
-Distorsion::~Distorsion(){
+Distorsion::~Distorsion()
+{
delete lpfl;
delete lpfr;
delete hpfl;
delete hpfr;
-
+
};
/*
* Cleanup the effect
*/
-void Distorsion::cleanup(){
+void Distorsion::cleanup()
+{
lpfl->cleanup();
hpfl->cleanup();
lpfr->cleanup();
@@ -190,11 +207,12 @@ void Distorsion::cleanup(){
* Apply the filters
*/
-void Distorsion::applyfilters(REALTYPE *efxoutl,REALTYPE *efxoutr){
+void Distorsion::applyfilters(REALTYPE *efxoutl,REALTYPE *efxoutr)
+{
lpfl->filterout(efxoutl);
hpfl->filterout(efxoutl);
- if (Pstereo!=0){//stereo
- lpfr->filterout(efxoutr);
+ if (Pstereo!=0) {//stereo
+ lpfr->filterout(efxoutr);
hpfr->filterout(efxoutr);
};
@@ -204,24 +222,25 @@ void Distorsion::applyfilters(REALTYPE *efxoutl,REALTYPE *efxoutr){
/*
* Effect output
*/
-void Distorsion::out(REALTYPE *smpsl,REALTYPE *smpsr){
+void Distorsion::out(REALTYPE *smpsl,REALTYPE *smpsr)
+{
int i;
REALTYPE l,r,lout,rout;
REALTYPE inputvol=pow(5.0,(Pdrive-32.0)/127.0);
if (Pnegate!=0) inputvol*=-1.0;
-
- if (Pstereo!=0){//Stereo
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]=smpsl[i]*inputvol*panning;
- efxoutr[i]=smpsr[i]*inputvol*(1.0-panning);
- };
+
+ if (Pstereo!=0) {//Stereo
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]=smpsl[i]*inputvol*panning;
+ efxoutr[i]=smpsr[i]*inputvol*(1.0-panning);
+ };
} else {
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]=( smpsl[i]*panning + smpsr[i]*(1.0-panning) ) * inputvol;
- };
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]=( smpsl[i]*panning + smpsr[i]*(1.0-panning) ) * inputvol;
+ };
};
-
+
if (Pprefiltering!=0) applyfilters(efxoutl,efxoutr);
//no optimised, yet (no look table)
@@ -229,60 +248,66 @@ void Distorsion::out(REALTYPE *smpsl,REALTYPE *smpsr){
if (Pstereo!=0) waveshapesmps(SOUND_BUFFER_SIZE,efxoutr,Ptype+1,Pdrive);
if (Pprefiltering==0) applyfilters(efxoutl,efxoutr);
-
+
if (Pstereo==0) for (i=0;i<SOUND_BUFFER_SIZE;i++) efxoutr[i]=efxoutl[i];
-
+
REALTYPE level=dB2rap(60.0*Plevel/127.0-40.0);
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- lout=efxoutl[i];
- rout=efxoutr[i];
- l=lout*(1.0-lrcross)+rout*lrcross;
- r=rout*(1.0-lrcross)+lout*lrcross;
- lout=l;rout=r;
-
- efxoutl[i]=lout*2.0*level;
- efxoutr[i]=rout*2.0*level;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ lout=efxoutl[i];
+ rout=efxoutr[i];
+ l=lout*(1.0-lrcross)+rout*lrcross;
+ r=rout*(1.0-lrcross)+lout*lrcross;
+ lout=l;
+ rout=r;
+
+ efxoutl[i]=lout*2.0*level;
+ efxoutr[i]=rout*2.0*level;
};
-
+
};
/*
* Parameter control
*/
-void Distorsion::setvolume(const unsigned char &Pvolume){
+void Distorsion::setvolume(const unsigned char &Pvolume)
+{
this->Pvolume=Pvolume;
if (insertion==0) {
- outvolume=pow(0.01,(1.0-Pvolume/127.0))*4.0;
- volume=1.0;
+ outvolume=pow(0.01,(1.0-Pvolume/127.0))*4.0;
+ volume=1.0;
} else {
- volume=outvolume=Pvolume/127.0;
+ volume=outvolume=Pvolume/127.0;
};
if (Pvolume==0) cleanup();
};
-void Distorsion::setpanning(const unsigned char &Ppanning){
+void Distorsion::setpanning(const unsigned char &Ppanning)
+{
this->Ppanning=Ppanning;
panning=(Ppanning+0.5)/127.0;
};
-void Distorsion::setlrcross(const unsigned char &Plrcross){
+void Distorsion::setlrcross(const unsigned char &Plrcross)
+{
this->Plrcross=Plrcross;
lrcross=Plrcross/127.0*1.0;
};
-void Distorsion::setlpf(const unsigned char &Plpf){
+void Distorsion::setlpf(const unsigned char &Plpf)
+{
this->Plpf=Plpf;
REALTYPE fr=exp(pow(Plpf/127.0,0.5)*log(25000.0))+40;
lpfl->setfreq(fr);
lpfr->setfreq(fr);
};
-void Distorsion::sethpf(const unsigned char &Phpf){
+void Distorsion::sethpf(const unsigned char &Phpf)
+{
this->Phpf=Phpf;
REALTYPE fr=exp(pow(Phpf/127.0,0.5)*log(25000.0))+20.0;
hpfl->setfreq(fr);
@@ -290,22 +315,24 @@ void Distorsion::sethpf(const unsigned char &Phpf){
};
-void Distorsion::setpreset(unsigned char npreset){
+void Distorsion::setpreset(unsigned char npreset)
+{
const int PRESET_SIZE=11;
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},
- //Overdrive 2
- {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},
- //A. Exciter 2
- {64,64,35,85,62,1,0,127,118,1,0},
- //Guitar Amp
- {127,64,35,63,75,2,0,55,0,0,0},
- //Quantisize
- {127,64,35,88,75,4,0,127,0,1,0}};
+ //Overdrive 1
+ {127,64,35,56,70,0,0,96,0,0,0},
+ //Overdrive 2
+ {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},
+ //A. Exciter 2
+ {64,64,35,85,62,1,0,127,118,1,0},
+ //Guitar Amp
+ {127,64,35,63,75,2,0,55,0,0,0},
+ //Quantisize
+ {127,64,35,88,75,4,0,127,0,1,0}
+ };
if (npreset>=NUM_PRESETS) npreset=NUM_PRESETS-1;
@@ -316,60 +343,84 @@ void Distorsion::setpreset(unsigned char npreset){
};
-void Distorsion::changepar(const int &npar,const unsigned char &value){
- switch (npar){
- case 0: setvolume(value);
- break;
- case 1: setpanning(value);
- break;
- case 2: setlrcross(value);
- break;
- case 3: Pdrive=value;
- break;
- case 4: Plevel=value;
- break;
- case 5: if (value>13) Ptype=13;//this must be increased if more distorsion types are added
- else Ptype=value;
- break;
- case 6: if (value>1) Pnegate=1;
- else Pnegate=value;
- break;
- case 7: setlpf(value);
- break;
- case 8: sethpf(value);
- break;
- case 9: if (value>1) Pstereo=1;
- else Pstereo=value;
- break;
- case 10:Pprefiltering=value;
- break;
+void Distorsion::changepar(const int &npar,const unsigned char &value)
+{
+ switch (npar) {
+ case 0:
+ setvolume(value);
+ break;
+ case 1:
+ setpanning(value);
+ break;
+ case 2:
+ setlrcross(value);
+ break;
+ case 3:
+ Pdrive=value;
+ break;
+ case 4:
+ Plevel=value;
+ break;
+ case 5:
+ if (value>13) Ptype=13;//this must be increased if more distorsion types are added
+ else Ptype=value;
+ break;
+ case 6:
+ if (value>1) Pnegate=1;
+ else Pnegate=value;
+ break;
+ case 7:
+ setlpf(value);
+ break;
+ case 8:
+ sethpf(value);
+ break;
+ case 9:
+ if (value>1) Pstereo=1;
+ else Pstereo=value;
+ break;
+ case 10:
+ Pprefiltering=value;
+ break;
};
};
-unsigned char Distorsion::getpar(const int &npar)const{
- switch (npar){
- case 0: return(Pvolume);
- break;
- case 1: return(Ppanning);
- break;
- case 2: return(Plrcross);
- break;
- case 3: return(Pdrive);
- break;
- case 4: return(Plevel);
- break;
- case 5: return(Ptype);
- break;
- case 6: return(Pnegate);
- break;
- case 7: return(Plpf);
- break;
- case 8: return(Phpf);
- break;
- case 9: return(Pstereo);
- break;
- case 10:return(Pprefiltering);
- break;
+unsigned char Distorsion::getpar(const int &npar)const
+{
+ switch (npar) {
+ case 0:
+ return(Pvolume);
+ break;
+ case 1:
+ return(Ppanning);
+ break;
+ case 2:
+ return(Plrcross);
+ break;
+ case 3:
+ return(Pdrive);
+ break;
+ case 4:
+ return(Plevel);
+ break;
+ case 5:
+ return(Ptype);
+ break;
+ case 6:
+ return(Pnegate);
+ break;
+ case 7:
+ return(Plpf);
+ break;
+ case 8:
+ return(Phpf);
+ break;
+ case 9:
+ return(Pstereo);
+ break;
+ case 10:
+ return(Pprefiltering);
+ break;
};
return(0);//in case of bogus parameter number
};
diff --git a/src/Effects/Distorsion.h b/src/Effects/Distorsion.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Distorsion.h - Distorsion Effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -30,40 +30,41 @@
//Waveshaping(called by Distorsion effect and waveshape from OscilGen)
void waveshapesmps(int n,REALTYPE *smps,unsigned char type,unsigned char drive);
/**Distortion Effect*/
-class Distorsion:public Effect{
- public:
- Distorsion(const int &insertion,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
- ~Distorsion();
- void out(REALTYPE *smpsl,REALTYPE *smpr);
- void setpreset(unsigned char npreset);
- void changepar(const int &npar,const unsigned char &value);
- unsigned char getpar(const int &npar)const;
- void cleanup();
- void applyfilters(REALTYPE *efxoutl,REALTYPE *efxoutr);
+class Distorsion:public Effect
+{
+public:
+ Distorsion(const int &insertion,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
+ ~Distorsion();
+ void out(REALTYPE *smpsl,REALTYPE *smpr);
+ void setpreset(unsigned char npreset);
+ void changepar(const int &npar,const unsigned char &value);
+ unsigned char getpar(const int &npar)const;
+ void cleanup();
+ void applyfilters(REALTYPE *efxoutl,REALTYPE *efxoutr);
+
+private:
+ //Parametrii
+ unsigned char Pvolume; //Volumul or E/R
+ unsigned char Ppanning;//Panning
+ unsigned char Plrcross;// L/R Mixing
+ unsigned char Pdrive; //the input amplification
+ unsigned char Plevel; //the output amplification
+ unsigned char Ptype; //Distorsion type
+ unsigned char Pnegate; //if the input is negated
+ unsigned char Plpf; //lowpass filter
+ unsigned char Phpf; //highpass filter
+ unsigned char Pstereo; //0=mono,1=stereo
+ unsigned char Pprefiltering;//if you want to do the filtering before the distorsion
+
+ void setvolume(const unsigned char &Pvolume);
+ void setpanning(const unsigned char &Ppanning);
+ void setlrcross(const unsigned char &Plrcross);
+ void setlpf(const unsigned char &Plpf);
+ void sethpf(const unsigned char &Phpf);
- private:
- //Parametrii
- unsigned char Pvolume; //Volumul or E/R
- unsigned char Ppanning;//Panning
- unsigned char Plrcross;// L/R Mixing
- unsigned char Pdrive; //the input amplification
- unsigned char Plevel; //the output amplification
- unsigned char Ptype; //Distorsion type
- unsigned char Pnegate; //if the input is negated
- unsigned char Plpf; //lowpass filter
- unsigned char Phpf; //highpass filter
- unsigned char Pstereo; //0=mono,1=stereo
- unsigned char Pprefiltering;//if you want to do the filtering before the distorsion
-
- void setvolume(const unsigned char &Pvolume);
- void setpanning(const unsigned char &Ppanning);
- void setlrcross(const unsigned char &Plrcross);
- void setlpf(const unsigned char &Plpf);
- void sethpf(const unsigned char &Phpf);
-
- //Real Parameters
- REALTYPE panning,lrcross;
- AnalogFilter *lpfl,*lpfr,*hpfl,*hpfr;
+ //Real Parameters
+ REALTYPE panning,lrcross;
+ AnalogFilter *lpfl,*lpfr,*hpfl,*hpfr;
};
diff --git a/src/Effects/DynamicFilter.C b/src/Effects/DynamicFilter.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
DynamicFilter.C - "WahWah" effect and others
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,16 +24,17 @@
#include "DynamicFilter.h"
DynamicFilter::DynamicFilter(int insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_)
- :Effect(insertion_,efxoutl_,efxoutr_,new FilterParams(0,64,64),0),
- Pvolume(110),Ppanning(64),Pdepth(0),Pampsns(90),
- Pampsnsinv(0),Pampsmooth(60),
- filterl(NULL),filterr(NULL)
+ :Effect(insertion_,efxoutl_,efxoutr_,new FilterParams(0,64,64),0),
+ Pvolume(110),Ppanning(64),Pdepth(0),Pampsns(90),
+ Pampsnsinv(0),Pampsmooth(60),
+ filterl(NULL),filterr(NULL)
{
setpreset(Ppreset);
cleanup();
};
-DynamicFilter::~DynamicFilter(){
+DynamicFilter::~DynamicFilter()
+{
delete filterpars;
delete filterl;
delete filterr;
@@ -43,27 +44,29 @@ DynamicFilter::~DynamicFilter(){
/*
* Apply the effect
*/
-void DynamicFilter::out(REALTYPE *smpsl,REALTYPE *smpsr){
+void DynamicFilter::out(REALTYPE *smpsl,REALTYPE *smpsr)
+{
int i;
- if (filterpars->changed){
- filterpars->changed=false;
- cleanup();
+ if (filterpars->changed) {
+ filterpars->changed=false;
+ cleanup();
};
-
+
REALTYPE lfol,lfor;
lfo.effectlfoout(&lfol,&lfor);
- lfol*=depth*5.0;lfor*=depth*5.0;
+ lfol*=depth*5.0;
+ lfor*=depth*5.0;
REALTYPE freq=filterpars->getfreq();
REALTYPE q=filterpars->getq();
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]=smpsl[i];
- efxoutr[i]=smpsr[i];
-
- REALTYPE x=(fabs(smpsl[i])+fabs(smpsr[i]))*0.5;
- ms1=ms1*(1.0-ampsmooth)+x*ampsmooth+1e-10;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]=smpsl[i];
+ efxoutr[i]=smpsr[i];
+
+ REALTYPE x=(fabs(smpsl[i])+fabs(smpsr[i]))*0.5;
+ ms1=ms1*(1.0-ampsmooth)+x*ampsmooth+1e-10;
};
-
+
REALTYPE ampsmooth2=pow(ampsmooth,0.2)*0.3;
ms2=ms2*(1.0-ampsmooth2)+ms1*ampsmooth2;
@@ -73,18 +76,18 @@ void DynamicFilter::out(REALTYPE *smpsl,REALTYPE *smpsr){
REALTYPE frl=filterl->getrealfreq(freq+lfol+rms);
REALTYPE frr=filterr->getrealfreq(freq+lfor+rms);
-
+
filterl->setfreq_and_q(frl,q);
filterr->setfreq_and_q(frr,q);
-
-
+
+
filterl->filterout(efxoutl);
filterr->filterout(efxoutr);
- //panning
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]*=panning;
- efxoutr[i]*=(1.0-panning);
+ //panning
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]*=panning;
+ efxoutr[i]*=(1.0-panning);
};
};
@@ -92,7 +95,8 @@ void DynamicFilter::out(REALTYPE *smpsl,REALTYPE *smpsr){
/*
* Cleanup the effect
*/
-void DynamicFilter::cleanup(){
+void DynamicFilter::cleanup()
+{
reinitfilter();
ms1=0.0;
ms2=0.0;
@@ -105,206 +109,236 @@ void DynamicFilter::cleanup(){
* Parameter control
*/
-void DynamicFilter::setdepth(const unsigned char &Pdepth){
+void DynamicFilter::setdepth(const unsigned char &Pdepth)
+{
this->Pdepth=Pdepth;
depth=pow((Pdepth/127.0),2.0);
};
-void DynamicFilter::setvolume(const unsigned char &Pvolume){
+void DynamicFilter::setvolume(const unsigned char &Pvolume)
+{
this->Pvolume=Pvolume;
outvolume=Pvolume/127.0;
if (insertion==0) volume=1.0;
- else volume=outvolume;
+ else volume=outvolume;
};
-void DynamicFilter::setpanning(const unsigned char &Ppanning){
+void DynamicFilter::setpanning(const unsigned char &Ppanning)
+{
this->Ppanning=Ppanning;
panning=Ppanning/127.0;
};
-void DynamicFilter::setampsns(const unsigned char &Pampsns){
+void DynamicFilter::setampsns(const unsigned char &Pampsns)
+{
ampsns=pow(Pampsns/127.0,2.5)*10.0;
- if (Pampsnsinv!=0) ampsns=-ampsns;
+ if (Pampsnsinv!=0) ampsns=-ampsns;
ampsmooth=exp(-Pampsmooth/127.0*10.0)*0.99;
this->Pampsns=Pampsns;
};
-void DynamicFilter::reinitfilter(){
+void DynamicFilter::reinitfilter()
+{
if (filterl!=NULL) delete(filterl);
if (filterr!=NULL) delete(filterr);
filterl=new Filter(filterpars);
filterr=new Filter(filterpars);
};
-void DynamicFilter::setpreset(unsigned char npreset){
+void DynamicFilter::setpreset(unsigned char npreset)
+{
const int PRESET_SIZE=10;
const int NUM_PRESETS=5;
unsigned char presets[NUM_PRESETS][PRESET_SIZE]={
- //WahWah
- {110,64,80,0,0,64,0,90,0,60},
- //AutoWah
- {110,64,70,0,0,80,70,0,0,60},
- //Sweep
- {100,64,30,0,0,50,80,0,0,60},
- //VocalMorph1
- {110,64,80,0,0,64,0,64,0,60},
- //VocalMorph1
- {127,64,50,0,0,96,64,0,0,60}};
-
+ //WahWah
+ {110,64,80,0,0,64,0,90,0,60},
+ //AutoWah
+ {110,64,70,0,0,80,70,0,0,60},
+ //Sweep
+ {100,64,30,0,0,50,80,0,0,60},
+ //VocalMorph1
+ {110,64,80,0,0,64,0,64,0,60},
+ //VocalMorph1
+ {127,64,50,0,0,96,64,0,0,60}
+ };
+
if (npreset>=NUM_PRESETS) npreset=NUM_PRESETS-1;
for (int n=0;n<PRESET_SIZE;n++) changepar(n,presets[npreset][n]);
-
+
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;
+ 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;
};
-
+
// for (int i=0;i<5;i++){
// printf("freq=%d amp=%d q=%d\n",filterpars->Pvowels[0].formants[i].freq,filterpars->Pvowels[0].formants[i].amp,filterpars->Pvowels[0].formants[i].q);
// };
if (insertion==0) changepar(0,presets[npreset][0]/2);//lower the volume if this is system effect
Ppreset=npreset;
-
+
reinitfilter();
};
-void DynamicFilter::changepar(const int &npar,const unsigned char &value){
- switch(npar){
- case 0: setvolume(value);
- break;
- case 1: setpanning(value);
- break;
- case 2: lfo.Pfreq=value;
- lfo.updateparams();
- break;
- case 3: lfo.Prandomness=value;
- lfo.updateparams();
- break;
- case 4: lfo.PLFOtype=value;
- lfo.updateparams();
- break;
- case 5: lfo.Pstereo=value;
- lfo.updateparams();
- break;
- case 6: setdepth(value);
- break;
- case 7: setampsns(value);
- break;
- case 8: Pampsnsinv=value;
- setampsns(Pampsns);
- break;
- case 9: Pampsmooth=value;
- setampsns(Pampsns);
- break;
+void DynamicFilter::changepar(const int &npar,const unsigned char &value)
+{
+ switch (npar) {
+ case 0:
+ setvolume(value);
+ break;
+ case 1:
+ setpanning(value);
+ break;
+ case 2:
+ lfo.Pfreq=value;
+ lfo.updateparams();
+ break;
+ case 3:
+ lfo.Prandomness=value;
+ lfo.updateparams();
+ break;
+ case 4:
+ lfo.PLFOtype=value;
+ lfo.updateparams();
+ break;
+ case 5:
+ lfo.Pstereo=value;
+ lfo.updateparams();
+ break;
+ case 6:
+ setdepth(value);
+ break;
+ case 7:
+ setampsns(value);
+ break;
+ case 8:
+ Pampsnsinv=value;
+ setampsns(Pampsns);
+ break;
+ case 9:
+ Pampsmooth=value;
+ setampsns(Pampsns);
+ break;
};
};
-unsigned char DynamicFilter::getpar(const int &npar)const{
- switch (npar){
- case 0: return(Pvolume);
- break;
- case 1: return(Ppanning);
- break;
- case 2: return(lfo.Pfreq);
- break;
- case 3: return(lfo.Prandomness);
- break;
- case 4: return(lfo.PLFOtype);
- break;
- case 5: return(lfo.Pstereo);
- break;
- case 6: return(Pdepth);
- break;
- case 7: return(Pampsns);
- break;
- case 8: return(Pampsnsinv);
- break;
- case 9: return(Pampsmooth);
- break;
- default:return (0);
+unsigned char DynamicFilter::getpar(const int &npar)const
+{
+ switch (npar) {
+ case 0:
+ return(Pvolume);
+ break;
+ case 1:
+ return(Ppanning);
+ break;
+ case 2:
+ return(lfo.Pfreq);
+ break;
+ case 3:
+ return(lfo.Prandomness);
+ break;
+ case 4:
+ return(lfo.PLFOtype);
+ break;
+ case 5:
+ return(lfo.Pstereo);
+ break;
+ case 6:
+ return(Pdepth);
+ break;
+ case 7:
+ return(Pampsns);
+ break;
+ case 8:
+ return(Pampsnsinv);
+ break;
+ case 9:
+ return(Pampsmooth);
+ break;
+ default:
+ return (0);
};
-
+
};
diff --git a/src/Effects/DynamicFilter.h b/src/Effects/DynamicFilter.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
DynamicFilter.h - "WahWah" effect and others
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,43 +28,44 @@
#include "../DSP/Filter.h"
/**DynamicFilter Effect*/
-class DynamicFilter:public Effect {
- public:
- DynamicFilter(int insetion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
- ~DynamicFilter();
- void out(REALTYPE *smpsl,REALTYPE *smpsr);
+class DynamicFilter:public Effect
+{
+public:
+ DynamicFilter(int insetion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
+ ~DynamicFilter();
+ void out(REALTYPE *smpsl,REALTYPE *smpsr);
- void setpreset(unsigned char npreset);
- void changepar(const int &npar,const unsigned char &value);
- unsigned char getpar(const int &npar)const;
- void cleanup();
+ void setpreset(unsigned char npreset);
+ void changepar(const int &npar,const unsigned char &value);
+ unsigned char getpar(const int &npar)const;
+ void cleanup();
// void setdryonly();
-
- private:
- //Parametrii DynamicFilter
- EffectLFO lfo;//lfo-ul DynamicFilter
- unsigned char Pvolume;
- unsigned char Ppanning;
- unsigned char Pdepth;/**<the depth of the lfo of the DynamicFilter*/
- unsigned char Pampsns;/**<how the filter varies according to the input amplitude*/
- unsigned char Pampsnsinv;//if the filter freq is lowered if the input amplitude rises
- unsigned char Pampsmooth;//how smooth the input amplitude changes the filter
-
- //Parameter Control
- void setvolume(const unsigned char &Pvolume);
- void setpanning(const unsigned char &Ppanning);
- void setdepth(const unsigned char &Pdepth);
- void setampsns(const unsigned char &Pampsns);
-
- void reinitfilter();
-
- //Internal Values
- REALTYPE panning,depth,ampsns,ampsmooth;
-
- Filter *filterl,*filterr;
-
- REALTYPE ms1,ms2,ms3,ms4;//mean squares
+
+private:
+ //Parametrii DynamicFilter
+ EffectLFO lfo;//lfo-ul DynamicFilter
+ unsigned char Pvolume;
+ unsigned char Ppanning;
+ unsigned char Pdepth;/**<the depth of the lfo of the DynamicFilter*/
+ unsigned char Pampsns;/**<how the filter varies according to the input amplitude*/
+ unsigned char Pampsnsinv;//if the filter freq is lowered if the input amplitude rises
+ unsigned char Pampsmooth;//how smooth the input amplitude changes the filter
+
+ //Parameter Control
+ void setvolume(const unsigned char &Pvolume);
+ void setpanning(const unsigned char &Ppanning);
+ void setdepth(const unsigned char &Pdepth);
+ void setampsns(const unsigned char &Pampsns);
+
+ void reinitfilter();
+
+ //Internal Values
+ REALTYPE panning,depth,ampsns,ampsmooth;
+
+ Filter *filterl,*filterr;
+
+ REALTYPE ms1,ms2,ms3,ms4;//mean squares
};
#endif
diff --git a/src/Effects/EQ.C b/src/Effects/EQ.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EQ.C - EQ effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,35 +24,37 @@
#include "EQ.h"
EQ::EQ(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_)
- :Effect(insertion_,efxoutl_,efxoutr_,NULL,0)
+ :Effect(insertion_,efxoutl_,efxoutr_,NULL,0)
{
-
- for (int i=0;i<MAX_EQ_BANDS;i++){
- filter[i].Ptype=0;
- filter[i].Pfreq=64;
- filter[i].Pgain=64;
- filter[i].Pq=64;
- filter[i].Pstages=0;
- filter[i].l=new AnalogFilter(6,1000.0,1.0,0);
- filter[i].r=new AnalogFilter(6,1000.0,1.0,0);
+
+ for (int i=0;i<MAX_EQ_BANDS;i++) {
+ filter[i].Ptype=0;
+ filter[i].Pfreq=64;
+ filter[i].Pgain=64;
+ filter[i].Pq=64;
+ filter[i].Pstages=0;
+ filter[i].l=new AnalogFilter(6,1000.0,1.0,0);
+ filter[i].r=new AnalogFilter(6,1000.0,1.0,0);
};
//default values
Pvolume=50;
-
- setpreset(Ppreset);
+
+ setpreset(Ppreset);
cleanup();
};
-EQ::~EQ(){
+EQ::~EQ()
+{
};
/*
* Cleanup the effect
*/
-void EQ::cleanup(){
- for (int i=0;i<MAX_EQ_BANDS;i++){
- filter[i].l->cleanup();
- filter[i].r->cleanup();
+void EQ::cleanup()
+{
+ for (int i=0;i<MAX_EQ_BANDS;i++) {
+ filter[i].l->cleanup();
+ filter[i].r->cleanup();
};
};
@@ -61,17 +63,18 @@ void EQ::cleanup(){
/*
* Effect output
*/
-void EQ::out(REALTYPE *smpsl,REALTYPE *smpsr){
+void EQ::out(REALTYPE *smpsl,REALTYPE *smpsr)
+{
int i;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]=smpsl[i]*volume;
- efxoutr[i]=smpsr[i]*volume;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]=smpsl[i]*volume;
+ efxoutr[i]=smpsr[i]*volume;
};
-
- for (i=0;i<MAX_EQ_BANDS;i++){
- if (filter[i].Ptype==0) continue;
- filter[i].l->filterout(efxoutl);
- filter[i].r->filterout(efxoutr);
+
+ for (i=0;i<MAX_EQ_BANDS;i++) {
+ if (filter[i].Ptype==0) continue;
+ filter[i].l->filterout(efxoutl);
+ filter[i].r->filterout(efxoutr);
};
};
@@ -79,27 +82,30 @@ void EQ::out(REALTYPE *smpsl,REALTYPE *smpsr){
/*
* Parameter control
*/
-void EQ::setvolume(const unsigned char &Pvolume){
+void EQ::setvolume(const unsigned char &Pvolume)
+{
this->Pvolume=Pvolume;
outvolume=pow(0.005,(1.0-Pvolume/127.0))*10.0;
if (insertion==0) {
- volume=1.0;
+ volume=1.0;
} else {
- volume=outvolume;
+ volume=outvolume;
};
};
-void EQ::setpreset(unsigned char npreset){
+void EQ::setpreset(unsigned char npreset)
+{
const int PRESET_SIZE=1;
const int NUM_PRESETS=2;
unsigned char presets[NUM_PRESETS][PRESET_SIZE]={
- //EQ 1
- {67},
- //EQ 2
- {67}};
+ //EQ 1
+ {67},
+ //EQ 2
+ {67}
+ };
if (npreset>=NUM_PRESETS) npreset=NUM_PRESETS-1;
for (int n=0;n<PRESET_SIZE;n++) changepar(n,presets[npreset][n]);
@@ -107,71 +113,85 @@ void EQ::setpreset(unsigned char npreset){
};
-void EQ::changepar(const int &npar,const unsigned char &value){
- switch (npar){
- case 0: setvolume(value);
- break;
+void EQ::changepar(const int &npar,const unsigned char &value)
+{
+ switch (npar) {
+ case 0:
+ setvolume(value);
+ break;
};
if (npar<10) return;
-
+
int nb=(npar-10)/5;//number of the band (filter)
if (nb>=MAX_EQ_BANDS) return;
int bp=npar%5;//band paramenter
-
+
REALTYPE 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=600.0*pow(30.0,(value-64.0)/64.0);
- filter[nb].l->setfreq(tmp);
- filter[nb].r->setfreq(tmp);
- break;
- case 2: filter[nb].Pgain=value;
- tmp=30.0*(value-64.0)/64.0;
- filter[nb].l->setgain(tmp);
- filter[nb].r->setgain(tmp);
- break;
- case 3: filter[nb].Pq=value;
- tmp=pow(30.0,(value-64.0)/64.0);
- 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;
+ 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=600.0*pow(30.0,(value-64.0)/64.0);
+ filter[nb].l->setfreq(tmp);
+ filter[nb].r->setfreq(tmp);
+ break;
+ case 2:
+ filter[nb].Pgain=value;
+ tmp=30.0*(value-64.0)/64.0;
+ filter[nb].l->setgain(tmp);
+ filter[nb].r->setgain(tmp);
+ break;
+ case 3:
+ filter[nb].Pq=value;
+ tmp=pow(30.0,(value-64.0)/64.0);
+ 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(const int &npar)const{
- switch (npar){
- case 0: return(Pvolume);
- break;
+unsigned char EQ::getpar(const int &npar)const
+{
+ switch (npar) {
+ case 0:
+ return(Pvolume);
+ break;
};
if (npar<10) return(0);
-
+
int nb=(npar-10)/5;//number of the band (filter)
if (nb>=MAX_EQ_BANDS) 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;
+ 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;
};
return(0);//in case of bogus parameter number
@@ -180,12 +200,13 @@ unsigned char EQ::getpar(const int &npar)const{
-REALTYPE EQ::getfreqresponse(REALTYPE freq){
+REALTYPE EQ::getfreqresponse(REALTYPE freq)
+{
REALTYPE resp=1.0;
- for (int i=0;i<MAX_EQ_BANDS;i++){
- if (filter[i].Ptype==0) continue;
- resp*=filter[i].l->H(freq);
+ for (int i=0;i<MAX_EQ_BANDS;i++) {
+ if (filter[i].Ptype==0) continue;
+ resp*=filter[i].l->H(freq);
};
return(rap2dB(resp*outvolume));
};
diff --git a/src/Effects/EQ.h b/src/Effects/EQ.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EQ.h - EQ Effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,29 +28,30 @@
#include "Effect.h"
/**EQ Effect*/
-class EQ:public Effect{
- public:
- EQ(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
- ~EQ();
- void out(REALTYPE *smpsl,REALTYPE *smpr);
- void setpreset(unsigned char npreset);
- void changepar(const int &npar,const unsigned char &value);
- unsigned char getpar(const int &npar)const;
- void cleanup();
- REALTYPE getfreqresponse(REALTYPE freq);
- private:
- //Parameters
- unsigned char Pvolume;/**<Volume*/
-
- void setvolume(const unsigned char &Pvolume);
-
- struct {
- //parameters
- unsigned char Ptype,Pfreq,Pgain,Pq,Pstages;
- //internal values
- AnalogFilter *l,*r;
- }filter[MAX_EQ_BANDS];
-
+class EQ:public Effect
+{
+public:
+ EQ(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
+ ~EQ();
+ void out(REALTYPE *smpsl,REALTYPE *smpr);
+ void setpreset(unsigned char npreset);
+ void changepar(const int &npar,const unsigned char &value);
+ unsigned char getpar(const int &npar)const;
+ void cleanup();
+ REALTYPE getfreqresponse(REALTYPE freq);
+private:
+ //Parameters
+ unsigned char Pvolume;/**<Volume*/
+
+ void setvolume(const unsigned char &Pvolume);
+
+ struct {
+ //parameters
+ unsigned char Ptype,Pfreq,Pgain,Pq,Pstages;
+ //internal values
+ AnalogFilter *l,*r;
+ }filter[MAX_EQ_BANDS];
+
};
diff --git a/src/Effects/Echo.C b/src/Effects/Echo.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Echo.C - Echo effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,21 +25,22 @@
#include "Echo.h"
Echo::Echo(const int & insertion_,REALTYPE *const efxoutl_,REALTYPE *const efxoutr_)
- : Effect(insertion_,efxoutl_,efxoutr_,NULL,0),
- Pvolume(50),Ppanning(64),//Pdelay(60),
- Plrdelay(100),Plrcross(100),Pfb(40),Phidamp(60),
- lrdelay(0),delaySample(1),old(0.0)
+ : Effect(insertion_,efxoutl_,efxoutr_,NULL,0),
+ Pvolume(50),Ppanning(64),//Pdelay(60),
+ Plrdelay(100),Plrcross(100),Pfb(40),Phidamp(60),
+ lrdelay(0),delaySample(1),old(0.0)
{
- setpreset(Ppreset);
+ setpreset(Ppreset);
cleanup();
}
-Echo::~Echo(){}
+Echo::~Echo() {}
/*
* Cleanup the effect
*/
-void Echo::cleanup(){
+void Echo::cleanup()
+{
delaySample.left().clear();
delaySample.right().clear();
old=Stereo<REALTYPE>(0.0);
@@ -49,12 +50,16 @@ void Echo::cleanup(){
/*
* Initialize the delays
*/
-void Echo::initdelays(){
- /**\todo make this adjust insted of destroy old delays*/
- kl=0;kr=0;
- dl=(int)(1+delay.getiVal()*SAMPLE_RATE-lrdelay);if (dl<1) dl=1;
- dr=(int)(1+delay.getiVal()*SAMPLE_RATE+lrdelay);if (dr<1) dr=1;
-
+void Echo::initdelays()
+{
+ /**\todo make this adjust insted of destroy old delays*/
+ kl=0;
+ kr=0;
+ dl=(int)(1+delay.getiVal()*SAMPLE_RATE-lrdelay);
+ if (dl<1) dl=1;
+ dr=(int)(1+delay.getiVal()*SAMPLE_RATE+lrdelay);
+ if (dr<1) dr=1;
+
delaySample.left()=AuSample(dl);
delaySample.right()=AuSample(dr);
@@ -64,18 +69,20 @@ void Echo::initdelays(){
/*
* Effect output
*/
-void Echo::out(REALTYPE *const smpsl,REALTYPE *const smpsr){
+void Echo::out(REALTYPE *const smpsl,REALTYPE *const smpsr)
+{
//void Echo::out(const Stereo<AuSample> & input){ //ideal
REALTYPE l,r,ldl,rdl;/**\todo move l+r->? ldl+rdl->?*/
Stereo<AuSample> input(AuSample(smpsl,SOUND_BUFFER_SIZE),AuSample(smpsr,SOUND_BUFFER_SIZE));
- for (int i=0;i<input.left().size();i++){
+ for (int i=0;i<input.left().size();i++) {
ldl=delaySample.left()[kl];
rdl=delaySample.right()[kr];
l=ldl*(1.0-lrcross)+rdl*lrcross;
r=rdl*(1.0-lrcross)+ldl*lrcross;
- ldl=l;rdl=r;
-
+ ldl=l;
+ rdl=r;
+
efxoutl[i]=ldl*2.0;
efxoutr[i]=rdl*2.0;
@@ -92,14 +99,15 @@ void Echo::out(REALTYPE *const smpsl,REALTYPE *const smpsr){
if (++kl>=dl) kl=0;
if (++kr>=dr) kr=0;
};
-
+
}
/*
* Parameter control
*/
-void Echo::setvolume(const unsigned char & Pvolume){
+void Echo::setvolume(const unsigned char & Pvolume)
+{
this->Pvolume=Pvolume;
if (insertion==0) {
@@ -112,43 +120,50 @@ void Echo::setvolume(const unsigned char & Pvolume){
}
-void Echo::setpanning(const unsigned char & Ppanning){
+void Echo::setpanning(const unsigned char & Ppanning)
+{
this->Ppanning=Ppanning;
panning=(Ppanning+0.5)/127.0;
}
-void Echo::setdelay(const unsigned char & Pdelay){
+void Echo::setdelay(const unsigned char & Pdelay)
+{
delay.setmVal(Pdelay);
//this->Pdelay=Pdelay;
//delay=1+(int)(Pdelay/127.0*SAMPLE_RATE*1.5);//0 .. 1.5 sec
initdelays();
}
-void Echo::setlrdelay(const unsigned char & Plrdelay){
+void Echo::setlrdelay(const unsigned char & Plrdelay)
+{
REALTYPE tmp;
- this->Plrdelay=Plrdelay;
+ this->Plrdelay=Plrdelay;
tmp=(pow(2,fabs(Plrdelay-64.0)/64.0*9)-1.0)/1000.0*SAMPLE_RATE;
if (Plrdelay<64.0) tmp=-tmp;
lrdelay=(int) tmp;
initdelays();
}
-void Echo::setlrcross(const unsigned char & Plrcross){
+void Echo::setlrcross(const unsigned char & Plrcross)
+{
this->Plrcross=Plrcross;
lrcross=Plrcross/127.0*1.0;
}
-void Echo::setfb(const unsigned char & Pfb){
+void Echo::setfb(const unsigned char & Pfb)
+{
this->Pfb=Pfb;
fb=Pfb/128.0;
}
-void Echo::sethidamp(const unsigned char & Phidamp){
+void Echo::sethidamp(const unsigned char & Phidamp)
+{
this->Phidamp=Phidamp;
hidamp=1.0-Phidamp/127.0;
}
-void Echo::setpreset(unsigned char npreset){
+void Echo::setpreset(unsigned char npreset)
+{
/**\todo see if the preset array can be replaced with a struct or a class*/
const int PRESET_SIZE=7;
const int NUM_PRESETS=9;
@@ -170,7 +185,8 @@ void Echo::setpreset(unsigned char npreset){
//Panning Echo 3
{81,60,26,100,127,67,36},
//Feedback Echo
- {62,64,28,64,100,90,55}};
+ {62,64,28,64,100,90,55}
+ };
if (npreset>=NUM_PRESETS) npreset=NUM_PRESETS-1;
@@ -180,41 +196,57 @@ void Echo::setpreset(unsigned char npreset){
}
-void Echo::changepar(const int & npar,const unsigned char & value){
- switch (npar){
- case 0: setvolume(value);
- break;
- case 1: setpanning(value);
- break;
- case 2: setdelay(value);
- break;
- case 3: setlrdelay(value);
- break;
- case 4: setlrcross(value);
- break;
- case 5: setfb(value);
- break;
- case 6: sethidamp(value);
- break;
+void Echo::changepar(const int & npar,const unsigned char & value)
+{
+ switch (npar) {
+ case 0:
+ setvolume(value);
+ break;
+ case 1:
+ setpanning(value);
+ break;
+ case 2:
+ setdelay(value);
+ break;
+ case 3:
+ setlrdelay(value);
+ break;
+ case 4:
+ setlrcross(value);
+ break;
+ case 5:
+ setfb(value);
+ break;
+ case 6:
+ sethidamp(value);
+ break;
};
}
-unsigned char Echo::getpar(const int & npar)const{
- switch (npar){
- case 0: return(Pvolume);
- break;
- case 1: return(Ppanning);
- break;
- case 2: return(delay.getmVal());
- break;
- case 3: return(Plrdelay);
- break;
- case 4: return(Plrcross);
- break;
- case 5: return(Pfb);
- break;
- case 6: return(Phidamp);
- break;
+unsigned char Echo::getpar(const int & npar)const
+{
+ switch (npar) {
+ case 0:
+ return(Pvolume);
+ break;
+ case 1:
+ return(Ppanning);
+ break;
+ case 2:
+ return(delay.getmVal());
+ break;
+ case 3:
+ return(Plrdelay);
+ break;
+ case 4:
+ return(Plrcross);
+ break;
+ case 5:
+ return(Pfb);
+ break;
+ case 6:
+ return(Phidamp);
+ break;
};
return(0);// in case of bogus parameter number
}
diff --git a/src/Effects/Echo.h b/src/Effects/Echo.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Echo.h - Echo Effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -30,105 +30,106 @@
#include "../Controls/DelayCtl.h"
/**Echo Effect*/
-class Echo:public Effect{
- public:
-
- /**
- * The Constructor For Echo
- * @param insertion_ integer to determine if Echo is an insertion effect
- * or not
- * @param efxoutl_ Effect out Left Channel
- * @param efxoutr_ Effect out Right Channel
- * @return An initialized Echo Object
- */
- Echo(const int & insertion_,REALTYPE *const efxoutl_,REALTYPE *const efxoutr_);
-
- /**
- * The destructor
- */
- ~Echo();
-
- /**
- * Outputs the echo to efxoutl and efxoutr
- * @param smpsl Sample from Left channel
- * @param smpsr Sample from Right channel
- * \todo try to figure out if smpsl should be const *const
- * or not (It should be)
- */
- void out(REALTYPE *const smpsl,REALTYPE *const smpr);
-
- /**
- * Sets the state of Echo to the specified preset
- * @param npreset number of chosen preset
- */
- void setpreset(unsigned char npreset);
-
- /**
- * Sets the value of the chosen variable
- *
- * The possible parameters are:
- * -# Volume
- * -# Panning
- * -# Delay
- * -# L/R Delay
- * -# L/R Crossover
- * -# Feedback
- * -# Dampening
- * @param npar number of chosen parameter
- * @param value the new value
- */
- void changepar(const int & npar,const unsigned char & value);
-
- /**
- * Gets the specified parameter
- *
- * The possible parameters are
- * -# Volume
- * -# Panning
- * -# Delay
- * -# L/R Delay
- * -# L/R Crossover
- * -# Feedback
- * -# Dampening
- * @param npar number of chosen parameter
- * @return value of parameter
- */
- unsigned char getpar(const int & npar)const;
-
- int getnumparams();
-
- /**Zeros out the state of the Echo*/
- void cleanup();
-
- /**\todo This function needs to be implemented or the prototype should be removed*/
- void setdryonly();
- private:
- //Parameters
- char Pvolume;/**<#1 Volume or Dry/Wetness*/
- char Ppanning;/**<#2 Panning*/
- DelayCtl delay;/**<#3 Delay of the Echo*/
- char Plrdelay;/**<#4 L/R delay difference*/
- char Plrcross;/**<#5 L/R Mixing*/
- char Pfb;/**<#6Feedback*/
- char Phidamp;/**<#7Dampening of the Echo*/
-
- void setvolume(const unsigned char & Pvolume);
- void setpanning(const unsigned char & Ppanning);
- void setdelay(const unsigned char & Pdelay);
- void setlrdelay(const unsigned char & Plrdelay);
- void setlrcross(const unsigned char & Plrcross);
- void setfb(const unsigned char & Pfb);
- void sethidamp(const unsigned char & Phidamp);
-
- //Real Parameters
- REALTYPE panning,lrcross,fb,hidamp; //needs better names
- int dl,dr,lrdelay; //needs better names
-
- void initdelays();
- Stereo<AuSample> delaySample;
- Stereo<REALTYPE> old;
-
- int kl,kr;
+class Echo:public Effect
+{
+public:
+
+ /**
+ * The Constructor For Echo
+ * @param insertion_ integer to determine if Echo is an insertion effect
+ * or not
+ * @param efxoutl_ Effect out Left Channel
+ * @param efxoutr_ Effect out Right Channel
+ * @return An initialized Echo Object
+ */
+ Echo(const int & insertion_,REALTYPE *const efxoutl_,REALTYPE *const efxoutr_);
+
+ /**
+ * The destructor
+ */
+ ~Echo();
+
+ /**
+ * Outputs the echo to efxoutl and efxoutr
+ * @param smpsl Sample from Left channel
+ * @param smpsr Sample from Right channel
+ * \todo try to figure out if smpsl should be const *const
+ * or not (It should be)
+ */
+ void out(REALTYPE *const smpsl,REALTYPE *const smpr);
+
+ /**
+ * Sets the state of Echo to the specified preset
+ * @param npreset number of chosen preset
+ */
+ void setpreset(unsigned char npreset);
+
+ /**
+ * Sets the value of the chosen variable
+ *
+ * The possible parameters are:
+ * -# Volume
+ * -# Panning
+ * -# Delay
+ * -# L/R Delay
+ * -# L/R Crossover
+ * -# Feedback
+ * -# Dampening
+ * @param npar number of chosen parameter
+ * @param value the new value
+ */
+ void changepar(const int & npar,const unsigned char & value);
+
+ /**
+ * Gets the specified parameter
+ *
+ * The possible parameters are
+ * -# Volume
+ * -# Panning
+ * -# Delay
+ * -# L/R Delay
+ * -# L/R Crossover
+ * -# Feedback
+ * -# Dampening
+ * @param npar number of chosen parameter
+ * @return value of parameter
+ */
+ unsigned char getpar(const int & npar)const;
+
+ int getnumparams();
+
+ /**Zeros out the state of the Echo*/
+ void cleanup();
+
+ /**\todo This function needs to be implemented or the prototype should be removed*/
+ void setdryonly();
+private:
+ //Parameters
+ char Pvolume;/**<#1 Volume or Dry/Wetness*/
+ char Ppanning;/**<#2 Panning*/
+ DelayCtl delay;/**<#3 Delay of the Echo*/
+ char Plrdelay;/**<#4 L/R delay difference*/
+ char Plrcross;/**<#5 L/R Mixing*/
+ char Pfb;/**<#6Feedback*/
+ char Phidamp;/**<#7Dampening of the Echo*/
+
+ void setvolume(const unsigned char & Pvolume);
+ void setpanning(const unsigned char & Ppanning);
+ void setdelay(const unsigned char & Pdelay);
+ void setlrdelay(const unsigned char & Plrdelay);
+ void setlrcross(const unsigned char & Plrcross);
+ void setfb(const unsigned char & Pfb);
+ void sethidamp(const unsigned char & Phidamp);
+
+ //Real Parameters
+ REALTYPE panning,lrcross,fb,hidamp; //needs better names
+ int dl,dr,lrdelay; //needs better names
+
+ void initdelays();
+ Stereo<AuSample> delaySample;
+ Stereo<REALTYPE> old;
+
+ int kl,kr;
};
#endif
diff --git a/src/Effects/Effect.C b/src/Effects/Effect.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Effect.C - this class is inherited by the all effects(Reverb, Echo, ..)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -22,9 +22,9 @@
#include "Effect.h"
-
+
Effect::Effect(bool insertion_,REALTYPE *const efxoutl_,
REALTYPE *const efxoutr_,FilterParams *filterpars_,
const unsigned char & Ppreset_)
- :Ppreset(Ppreset_),efxoutl(efxoutl_),efxoutr(efxoutr_),
- filterpars(filterpars_),insertion(insertion_){}
+ :Ppreset(Ppreset_),efxoutl(efxoutl_),efxoutr(efxoutr_),
+ filterpars(filterpars_),insertion(insertion_) {}
diff --git a/src/Effects/Effect.h b/src/Effects/Effect.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Effect.h - this class is inherited by the all effects(Reverb, Echo, ..)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,70 +29,73 @@
/**this class is inherited by the all effects(Reverb, Echo, ..)*/
-class Effect{
- public:
- /**
- * Effect Constructor
- * @param insertion_ 1 when it is an insertion Effect and 0 when it
- * is not an insertion Effect
- * @param efxoutl_ Effect output buffer Left channel
- * @param efxoutr_ Effect output buffer Right channel
- * @param filterpars_ pointer to FilterParams array
- * @param Ppreset_ chosen preset
- * @return Initialized Effect object*/
- Effect(bool insertion_,REALTYPE *const efxoutl_,
- REALTYPE *const efxoutr_,FilterParams *filterpars_,
- const unsigned char & Ppreset_);
- /**Deconstructor
- *
- * Deconstructs the Effect and releases any resouces that it has
- * allocated for itself*/
- virtual ~Effect(){};
- /**
- * Choose a preset
- * @param npreset number of chosen preset*/
- virtual void setpreset(unsigned char npreset)=0;
- /**Change parameter npar to value
- * @param npar chosen parameter
- * @param value chosen new value*/
- virtual void changepar(const int &npar,const unsigned char &value)=0;
- /**Get the value of parameter npar
- * @param npar chosen parameter
- * @return the value of the parameter in an unsigned char or 0 if it
- * does not exist*/
- virtual unsigned char getpar(const int &npar)const=0;
- /**Output result of effect based on the given buffers
- *
- * This method should result in the effect generating its results
- * and placing them into the efxoutl and efxoutr buffers.
- * Every Effect should overide this method.
- *
- * @param smpsl Input buffer for the Left channel
- * @param smpsr Input buffer for the Right channel
- */
- virtual void out(REALTYPE *const smpsl,REALTYPE *const smpsr)=0;
- /**Reset the state of the effect*/
- virtual void cleanup(){};
- /**This is only used for EQ (for user interface)*/
- virtual REALTYPE getfreqresponse(REALTYPE freq){return (0);};
-
- unsigned char Ppreset;/**<Currently used preset*/
- REALTYPE *const efxoutl;/**<Effect out Left Channel*/
- REALTYPE *const efxoutr;/**<Effect out Right Channel*/
- /**\todo make efxoutl and efxoutr private and replace them with a StereoSample*/
-
- REALTYPE outvolume;/**<This is the volume of effect and is public because
+class Effect
+{
+public:
+ /**
+ * Effect Constructor
+ * @param insertion_ 1 when it is an insertion Effect and 0 when it
+ * is not an insertion Effect
+ * @param efxoutl_ Effect output buffer Left channel
+ * @param efxoutr_ Effect output buffer Right channel
+ * @param filterpars_ pointer to FilterParams array
+ * @param Ppreset_ chosen preset
+ * @return Initialized Effect object*/
+ Effect(bool insertion_,REALTYPE *const efxoutl_,
+ REALTYPE *const efxoutr_,FilterParams *filterpars_,
+ const unsigned char & Ppreset_);
+ /**Deconstructor
+ *
+ * Deconstructs the Effect and releases any resouces that it has
+ * allocated for itself*/
+ virtual ~Effect() {};
+ /**
+ * Choose a preset
+ * @param npreset number of chosen preset*/
+ virtual void setpreset(unsigned char npreset)=0;
+ /**Change parameter npar to value
+ * @param npar chosen parameter
+ * @param value chosen new value*/
+ virtual void changepar(const int &npar,const unsigned char &value)=0;
+ /**Get the value of parameter npar
+ * @param npar chosen parameter
+ * @return the value of the parameter in an unsigned char or 0 if it
+ * does not exist*/
+ virtual unsigned char getpar(const int &npar)const=0;
+ /**Output result of effect based on the given buffers
+ *
+ * This method should result in the effect generating its results
+ * and placing them into the efxoutl and efxoutr buffers.
+ * Every Effect should overide this method.
+ *
+ * @param smpsl Input buffer for the Left channel
+ * @param smpsr Input buffer for the Right channel
+ */
+ virtual void out(REALTYPE *const smpsl,REALTYPE *const smpsr)=0;
+ /**Reset the state of the effect*/
+ virtual void cleanup() {};
+ /**This is only used for EQ (for user interface)*/
+ virtual REALTYPE getfreqresponse(REALTYPE freq) {
+ return (0);
+ };
+
+ unsigned char Ppreset;/**<Currently used preset*/
+ REALTYPE *const efxoutl;/**<Effect out Left Channel*/
+ REALTYPE *const efxoutr;/**<Effect out Right Channel*/
+ /**\todo make efxoutl and efxoutr private and replace them with a StereoSample*/
+
+ REALTYPE outvolume;/**<This is the volume of effect and is public because
* it is needed in system effects.
* The out volume of such effects are always 1.0, so
* this setting tells me how is the volume to the
* Master Output only.*/
- REALTYPE volume;
+ REALTYPE volume;
- FilterParams *filterpars;/**<Parameters for filters used by Effect*/
- protected:
+ FilterParams *filterpars;/**<Parameters for filters used by Effect*/
+protected:
- const bool insertion;/**<If Effect is an insertion effect, insertion=1
+ const bool insertion;/**<If Effect is an insertion effect, insertion=1
*otherwise, it should be insertion=0*/
};
diff --git a/src/Effects/EffectLFO.C b/src/Effects/EffectLFO.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EffectLFO.C - Stereo LFO used by some effects
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,13 +26,15 @@
#include "EffectLFO.h"
-EffectLFO::EffectLFO(){
- xl=0.0;xr=0.0;
+EffectLFO::EffectLFO()
+{
+ xl=0.0;
+ xr=0.0;
Pfreq=40;
Prandomness=0;
PLFOtype=0;
Pstereo=96;
-
+
updateparams();
ampl1=(1-lfornd)+lfornd*RND;
@@ -41,20 +43,23 @@ EffectLFO::EffectLFO(){
ampr2=(1-lfornd)+lfornd*RND;
};
-EffectLFO::~EffectLFO(){
+EffectLFO::~EffectLFO()
+{
};
/*
* Update the changed parameters
*/
-void EffectLFO::updateparams(){
+void EffectLFO::updateparams()
+{
REALTYPE lfofreq=(pow(2,Pfreq/127.0*10.0)-1.0)*0.03;
incx=fabs(lfofreq)*(REALTYPE)SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE;
if (incx>0.49999999) incx=0.499999999; //Limit the Frequency
lfornd=Prandomness/127.0;
- if (lfornd<0.0) lfornd=0.0; else if (lfornd>1.0) lfornd=1.0;
+ if (lfornd<0.0) lfornd=0.0;
+ else if (lfornd>1.0) lfornd=1.0;
if (PLFOtype>1) PLFOtype=1;//this has to be updated if more lfo's are added
lfotype=PLFOtype;
@@ -66,16 +71,18 @@ void EffectLFO::updateparams(){
/*
* Compute the shape of the LFO
*/
-REALTYPE EffectLFO::getlfoshape(REALTYPE x){
+REALTYPE EffectLFO::getlfoshape(REALTYPE x)
+{
REALTYPE out;
- switch (lfotype){
- case 1: //EffectLFO_TRIANGLE
- if ((x>0.0)&&(x<0.25)) out=4.0*x;
- else if ((x>0.25)&&(x<0.75)) out=2-4*x;
- else out=4.0*x-4.0;
- break;
- /**\todo more to be added here; also ::updateparams() need to be updated (to allow more lfotypes)*/
- default:out=cos(x*2*PI);//EffectLFO_SINE
+ switch (lfotype) {
+ case 1: //EffectLFO_TRIANGLE
+ if ((x>0.0)&&(x<0.25)) out=4.0*x;
+ else if ((x>0.25)&&(x<0.75)) out=2-4*x;
+ else out=4.0*x-4.0;
+ break;
+ /**\todo more to be added here; also ::updateparams() need to be updated (to allow more lfotypes)*/
+ default:
+ out=cos(x*2*PI);//EffectLFO_SINE
};
return(out);
};
@@ -83,16 +90,17 @@ REALTYPE EffectLFO::getlfoshape(REALTYPE x){
/*
* LFO output
*/
-void EffectLFO::effectlfoout(REALTYPE *outl,REALTYPE *outr){
- REALTYPE out;
-
+void EffectLFO::effectlfoout(REALTYPE *outl,REALTYPE *outr)
+{
+ REALTYPE out;
+
out=getlfoshape(xl);
if ((lfotype==0)||(lfotype==1)) out*=(ampl1+xl*(ampl2-ampl1));
xl+=incx;
if (xl>1.0) {
- xl-=1.0;
- ampl1=ampl2;
- ampl2=(1.0-lfornd)+lfornd*RND;
+ xl-=1.0;
+ ampl1=ampl2;
+ ampl2=(1.0-lfornd)+lfornd*RND;
};
*outl=(out+1.0)*0.5;
@@ -100,9 +108,9 @@ void EffectLFO::effectlfoout(REALTYPE *outl,REALTYPE *outr){
if ((lfotype==0)||(lfotype==1)) out*=(ampr1+xr*(ampr2-ampr1));
xr+=incx;
if (xr>1.0) {
- xr-=1.0;
- ampr1=ampr2;
- ampr2=(1.0-lfornd)+lfornd*RND;
+ xr-=1.0;
+ ampr1=ampr2;
+ ampr2=(1.0-lfornd)+lfornd*RND;
};
*outr=(out+1.0)*0.5;
};
diff --git a/src/Effects/EffectLFO.h b/src/Effects/EffectLFO.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EffectLFO.h - Stereo LFO used by some effects
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,8 +26,9 @@
#include "../globals.h"
/**LFO for some of the Effect objects
* \todo see if this should inherit LFO*/
-class EffectLFO{
- public:
+class EffectLFO
+{
+public:
EffectLFO();
~EffectLFO();
void effectlfoout(REALTYPE *outl,REALTYPE *outr);
@@ -36,9 +37,9 @@ class EffectLFO{
unsigned char Prandomness;
unsigned char PLFOtype;
unsigned char Pstereo;//"64"=0
- private:
+private:
REALTYPE getlfoshape(REALTYPE x);
-
+
REALTYPE xl,xr;
REALTYPE incx;
REALTYPE ampl1,ampl2,ampr1,ampr2;//necessary for "randomness"
diff --git a/src/Effects/EffectMgr.C b/src/Effects/EffectMgr.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EffectMgr.C - Effect manager, an interface betwen the program and effects
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -23,10 +23,10 @@
#include "EffectMgr.h"
EffectMgr::EffectMgr(int insertion_,pthread_mutex_t *mutex_)
- :insertion(insertion_),
- efxoutl(new REALTYPE[SOUND_BUFFER_SIZE]),
- efxoutr(new REALTYPE[SOUND_BUFFER_SIZE]),
- filterpars(NULL),nefx(0),efx(NULL),mutex(mutex_),dryonly(false)
+ :insertion(insertion_),
+ efxoutl(new REALTYPE[SOUND_BUFFER_SIZE]),
+ efxoutr(new REALTYPE[SOUND_BUFFER_SIZE]),
+ filterpars(NULL),nefx(0),efx(NULL),mutex(mutex_),dryonly(false)
{
setpresettype("Peffect"); /**\todo Figure out what this is doing
* , as it might be another leaky abstraction.*/
@@ -36,9 +36,9 @@ EffectMgr::EffectMgr(int insertion_,pthread_mutex_t *mutex_)
// mutex=mutex_;
// efxoutl=new REALTYPE[SOUND_BUFFER_SIZE];
// efxoutr=new REALTYPE[SOUND_BUFFER_SIZE];
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]=0.0;
- efxoutr[i]=0.0;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]=0.0;
+ efxoutr[i]=0.0;
};
// filterpars=NULL;
// dryonly=false;
@@ -46,13 +46,15 @@ EffectMgr::EffectMgr(int insertion_,pthread_mutex_t *mutex_)
}
-EffectMgr::~EffectMgr(){
+EffectMgr::~EffectMgr()
+{
if (efx!=NULL) delete efx;
delete []efxoutl;
delete []efxoutr;
}
-void EffectMgr::defaults(){
+void EffectMgr::defaults()
+{
changeeffect(0);
setdryonly(false);
}
@@ -60,43 +62,64 @@ void EffectMgr::defaults(){
/*
* Change the effect
*/
-void EffectMgr::changeeffect(int nefx_){
+void EffectMgr::changeeffect(int nefx_)
+{
cleanup();
if (nefx==nefx_) return;
nefx=nefx_;
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]=0.0;
- efxoutr[i]=0.0;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]=0.0;
+ efxoutr[i]=0.0;
};
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)
+ 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)
};
-
+
if (efx!=NULL) filterpars=efx->filterpars;
}
/*
* Obtain the effect number
*/
-int EffectMgr::geteffect(){
+int EffectMgr::geteffect()
+{
return (nefx);
}
/*
* Cleanup the current effect
*/
-void EffectMgr::cleanup(){
+void EffectMgr::cleanup()
+{
if (efx!=NULL) efx->cleanup();
}
@@ -104,23 +127,26 @@ void EffectMgr::cleanup(){
/*
* Get the preset of the current effect
*/
-
-unsigned char EffectMgr::getpreset(){
+
+unsigned char EffectMgr::getpreset()
+{
if (efx!=NULL) return(efx->Ppreset);
- else return(0);
+ else return(0);
}
/*
* Change the preset of the current effect
*/
-void EffectMgr::changepreset_nolock(unsigned char npreset){
+void EffectMgr::changepreset_nolock(unsigned char npreset)
+{
if (efx!=NULL) efx->setpreset(npreset);
}
/*
* Change the preset of the current effect(with thread locking)
*/
-void EffectMgr::changepreset(unsigned char npreset){
+void EffectMgr::changepreset(unsigned char npreset)
+{
pthread_mutex_lock(mutex);
changepreset_nolock(npreset);
pthread_mutex_unlock(mutex);
@@ -128,9 +154,10 @@ void EffectMgr::changepreset(unsigned char npreset){
/*
- * Change a parameter of the current effect
+ * Change a parameter of the current effect
*/
-void EffectMgr::seteffectpar_nolock(int npar,unsigned char value){
+void EffectMgr::seteffectpar_nolock(int npar,unsigned char value)
+{
if (efx==NULL) return;
efx->changepar(npar,value);
}
@@ -138,7 +165,8 @@ void EffectMgr::seteffectpar_nolock(int npar,unsigned char value){
/*
* Change a parameter of the current effect (with thread locking)
*/
-void EffectMgr::seteffectpar(int npar,unsigned char value){
+void EffectMgr::seteffectpar(int npar,unsigned char value)
+{
pthread_mutex_lock(mutex);
seteffectpar_nolock(npar,value);
pthread_mutex_unlock(mutex);
@@ -147,7 +175,8 @@ void EffectMgr::seteffectpar(int npar,unsigned char value){
/*
* Get a parameter of the current effect
*/
-unsigned char EffectMgr::geteffectpar(int npar){
+unsigned char EffectMgr::geteffectpar(int npar)
+{
if (efx==NULL) return(0);
return(efx->getpar(npar));
}
@@ -156,139 +185,147 @@ unsigned char EffectMgr::geteffectpar(int npar){
/*
* Apply the effect
*/
-void EffectMgr::out(REALTYPE *smpsl,REALTYPE *smpsr){
+void EffectMgr::out(REALTYPE *smpsl,REALTYPE *smpsr)
+{
int i;
- if (efx==NULL){
- if (insertion==0)
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- smpsl[i]=0.0;smpsr[i]=0.0;
- efxoutl[i]=0.0;efxoutr[i]=0.0;
- };
- return;
+ if (efx==NULL) {
+ if (insertion==0)
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ smpsl[i]=0.0;
+ smpsr[i]=0.0;
+ efxoutl[i]=0.0;
+ efxoutr[i]=0.0;
+ };
+ return;
};
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- smpsl[i]+=denormalkillbuf[i];
- smpsr[i]+=denormalkillbuf[i];
- efxoutl[i]=0.0;
- efxoutr[i]=0.0;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ smpsl[i]+=denormalkillbuf[i];
+ smpsr[i]+=denormalkillbuf[i];
+ efxoutl[i]=0.0;
+ efxoutr[i]=0.0;
};
efx->out(smpsl,smpsr);
-
+
REALTYPE volume=efx->volume;
-
- if (nefx==7){//this is need only for the EQ effect
- /**\todo figure out why*/
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- smpsl[i]=efxoutl[i];
- smpsr[i]=efxoutr[i];
- };
- return;
+
+ if (nefx==7) {//this is need only for the EQ effect
+ /**\todo figure out why*/
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ smpsl[i]=efxoutl[i];
+ smpsr[i]=efxoutr[i];
+ };
+ return;
};
-
+
//Insertion effect
if (insertion!=0) {
REALTYPE v1,v2;
- if (volume<0.5) {
- v1=1.0;
- v2=volume*2.0;
- } else {
- v1=(1.0-volume)*2.0;
- v2=1.0;
- };
- 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
- 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
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- smpsl[i]=smpsl[i]*v1+efxoutl[i]*v2;
- smpsr[i]=smpsr[i]*v1+efxoutr[i]*v2;
- };
- };
+ if (volume<0.5) {
+ v1=1.0;
+ v2=volume*2.0;
+ } else {
+ v1=(1.0-volume)*2.0;
+ v2=1.0;
+ };
+ 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
+ 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
+ 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
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]*=2.0*volume;
- efxoutr[i]*=2.0*volume;
- smpsl[i]=efxoutl[i];
- smpsr[i]=efxoutr[i];
- };
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]*=2.0*volume;
+ efxoutr[i]*=2.0*volume;
+ smpsl[i]=efxoutl[i];
+ smpsr[i]=efxoutr[i];
+ };
};
-
+
}
/*
* Get the effect volume for the system effect
*/
-REALTYPE EffectMgr::sysefxgetvolume(){
+REALTYPE EffectMgr::sysefxgetvolume()
+{
if (efx==NULL) return (1.0);
- else return(efx->outvolume);
+ else return(efx->outvolume);
}
/*
* Get the EQ response
*/
-REALTYPE EffectMgr::getEQfreqresponse(REALTYPE freq){
+REALTYPE EffectMgr::getEQfreqresponse(REALTYPE freq)
+{
if (nefx==7) return(efx->getfreqresponse(freq));
- else return(0.0);
+ else return(0.0);
}
-void EffectMgr::setdryonly(bool value){
+void EffectMgr::setdryonly(bool value)
+{
dryonly=value;
}
-void EffectMgr::add2XML(XMLwrapper *xml){
+void EffectMgr::add2XML(XMLwrapper *xml)
+{
xml->addpar("type",geteffect());
if ((efx==NULL)||(geteffect()==0)) return;
xml->addpar("preset",efx->Ppreset);
xml->beginbranch("EFFECT_PARAMETERS");
- for (int n=0;n<128;n++){ /**\todo evaluate who should oversee saving
+ for (int n=0;n<128;n++) { /**\todo evaluate who should oversee saving
* and loading of parameters*/
- int par=geteffectpar(n);
- if (par==0) continue;
- xml->beginbranch("par_no",n);
- xml->addpar("par",par);
- xml->endbranch();
- };
- if (filterpars!=NULL){
- xml->beginbranch("FILTER");
- filterpars->add2XML(xml);
- xml->endbranch();
- };
+ int par=geteffectpar(n);
+ if (par==0) continue;
+ xml->beginbranch("par_no",n);
+ xml->addpar("par",par);
+ xml->endbranch();
+ };
+ if (filterpars!=NULL) {
+ xml->beginbranch("FILTER");
+ filterpars->add2XML(xml);
+ xml->endbranch();
+ };
xml->endbranch();
}
-void EffectMgr::getfromXML(XMLwrapper *xml){
+void EffectMgr::getfromXML(XMLwrapper *xml)
+{
changeeffect(xml->getpar127("type",geteffect()));
if ((efx==NULL)||(geteffect()==0)) return;
-
+
efx->Ppreset=xml->getpar127("preset",efx->Ppreset);
- if (xml->enterbranch("EFFECT_PARAMETERS")){
- for (int n=0;n<128;n++){
- seteffectpar_nolock(n,0);//erase effect parameter
- if (xml->enterbranch("par_no",n)==0) continue;
-
- int par=geteffectpar(n);
- seteffectpar_nolock(n,xml->getpar127("par",par));
- xml->exitbranch();
- };
- if (filterpars!=NULL){
- if (xml->enterbranch("FILTER")){
- filterpars->getfromXML(xml);
- xml->exitbranch();
- };
- };
- xml->exitbranch();
+ if (xml->enterbranch("EFFECT_PARAMETERS")) {
+ for (int n=0;n<128;n++) {
+ seteffectpar_nolock(n,0);//erase effect parameter
+ if (xml->enterbranch("par_no",n)==0) continue;
+
+ int par=geteffectpar(n);
+ seteffectpar_nolock(n,xml->getpar127("par",par));
+ xml->exitbranch();
+ };
+ 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
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EffectMgr.h - Effect manager, an interface betwen the program and effects
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -39,63 +39,64 @@
/**Effect manager, an interface betwen the program and effects*/
-class EffectMgr:public Presets{
- public:
- EffectMgr(int insertion_,pthread_mutex_t *mutex_);
- ~EffectMgr();
+class EffectMgr:public Presets
+{
+public:
+ EffectMgr(int insertion_,pthread_mutex_t *mutex_);
+ ~EffectMgr();
+
+ void add2XML(XMLwrapper *xml);
+ void defaults();
+ void getfromXML(XMLwrapper *xml);
- void add2XML(XMLwrapper *xml);
- void defaults();
- void getfromXML(XMLwrapper *xml);
+ void out(REALTYPE *smpsl,REALTYPE *smpsr);
- void out(REALTYPE *smpsl,REALTYPE *smpsr);
+ void setdryonly(bool value);
- void setdryonly(bool value);
-
- /**get the output(to speakers) volume of the systemeffect*/
- REALTYPE sysefxgetvolume();
+ /**get the output(to speakers) volume of the systemeffect*/
+ REALTYPE sysefxgetvolume();
- void cleanup();/**<cleanup the effect*/
+ void cleanup();/**<cleanup the effect*/
- /**change effect to the given int
+ /**change effect to the given int
* @param nefx_ the number of the effect*/
- void changeeffect(int nefx_);
- /**Get the number of the effect
- * @return the number*/
- int geteffect();
- /**
- * Change the preset to the given one
- * @param npreset number of the chosen preset
- */
- void changepreset(unsigned char npreset);
- /**
- * Change the preset to the given one without locking the thread
- * @param npreset number of the chosen preset
- */
- void changepreset_nolock(unsigned char npreset);
- /**
- * Get the current preset
- * @return the current preset*/
- unsigned char getpreset();
- /**sets the effect par*/
- void seteffectpar(int npar,unsigned char value);
- /**<sets the effect par without thread lock*/
- 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*/
- REALTYPE *efxoutl,*efxoutr;
-
- /**used by UI
+ void changeeffect(int nefx_);
+ /**Get the number of the effect
+ * @return the number*/
+ int geteffect();
+ /**
+ * Change the preset to the given one
+ * @param npreset number of the chosen preset
+ */
+ void changepreset(unsigned char npreset);
+ /**
+ * Change the preset to the given one without locking the thread
+ * @param npreset number of the chosen preset
+ */
+ void changepreset_nolock(unsigned char npreset);
+ /**
+ * Get the current preset
+ * @return the current preset*/
+ unsigned char getpreset();
+ /**sets the effect par*/
+ void seteffectpar(int npar,unsigned char value);
+ /**<sets the effect par without thread lock*/
+ 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*/
+ REALTYPE *efxoutl,*efxoutr;
+
+ /**used by UI
* \todo needs to be decoupled*/
- REALTYPE getEQfreqresponse(REALTYPE freq);
-
- FilterParams *filterpars;
-
- private:
- int nefx;
- Effect *efx;
- pthread_mutex_t *mutex;
- bool dryonly;
+ REALTYPE getEQfreqresponse(REALTYPE freq);
+
+ FilterParams *filterpars;
+
+private:
+ int nefx;
+ Effect *efx;
+ pthread_mutex_t *mutex;
+ bool dryonly;
};
#endif
diff --git a/src/Effects/Phaser.C b/src/Effects/Phaser.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Phaser.C - Phaser effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,13 +25,14 @@
#define PHASER_LFO_SHAPE 2
Phaser::Phaser(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_)
- :Effect(insertion_,efxoutl_,efxoutr_,NULL,0),old(1),oldgain(0.0)
+ :Effect(insertion_,efxoutl_,efxoutr_,NULL,0),old(1),oldgain(0.0)
{
setpreset(Ppreset);
cleanup();
};
-Phaser::~Phaser(){
+Phaser::~Phaser()
+{
//if (oldl!=NULL) delete [] oldl;
//if (oldr!=NULL) delete [] oldr;
};
@@ -40,7 +41,8 @@ Phaser::~Phaser(){
/*
* Effect output
*/
-void Phaser::out(REALTYPE *smpsl,REALTYPE *smpsr){
+void Phaser::out(REALTYPE *smpsl,REALTYPE *smpsr)
+{
int i,j;
REALTYPE lfol,lfor,lgain,rgain,tmp;
@@ -53,58 +55,62 @@ void Phaser::out(REALTYPE *smpsl,REALTYPE *smpsr){
lgain=1.0-phase*(1.0-depth)-(1.0-phase)*lgain*depth;
rgain=1.0-phase*(1.0-depth)-(1.0-phase)*rgain*depth;
-
- if (lgain>1.0) lgain=1.0;else if (lgain<0.0) lgain=0.0;
- if (rgain>1.0) rgain=1.0;else if (rgain<0.0) rgain=0.0;
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- REALTYPE x=(REALTYPE) i /SOUND_BUFFER_SIZE;
- REALTYPE x1=1.0-x;
- REALTYPE gl=lgain*x+oldgain.left()*x1;
- REALTYPE gr=rgain*x+oldgain.right()*x1;
- REALTYPE inl=smpsl[i]*panning+fbl;
- REALTYPE inr=smpsr[i]*(1.0-panning)+fbr;
-
- //Left channel
- for (j=0;j<Pstages*2;j++){//Phasing routine
- tmp=old.left()[j];
- old.left()[j]=gl*tmp+inl;
- inl=tmp-gl*old.left()[j];
- };
- //Right channel
- for (j=0;j<Pstages*2;j++){//Phasing routine
- tmp=old.right()[j];
- old.right()[j]=gr*tmp+inr;
- inr=tmp-gr*old.right()[j];
- };
- //Left/Right crossing
- REALTYPE l=inl;
- REALTYPE r=inr;
- inl=l*(1.0-lrcross)+r*lrcross;
- inr=r*(1.0-lrcross)+l*lrcross;
-
- fbl=inl*fb;
- fbr=inr*fb;
- efxoutl[i]=inl;
- efxoutr[i]=inr;
+
+ if (lgain>1.0) lgain=1.0;
+ else if (lgain<0.0) lgain=0.0;
+ if (rgain>1.0) rgain=1.0;
+ else if (rgain<0.0) rgain=0.0;
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE x=(REALTYPE) i /SOUND_BUFFER_SIZE;
+ REALTYPE x1=1.0-x;
+ REALTYPE gl=lgain*x+oldgain.left()*x1;
+ REALTYPE gr=rgain*x+oldgain.right()*x1;
+ REALTYPE inl=smpsl[i]*panning+fbl;
+ REALTYPE inr=smpsr[i]*(1.0-panning)+fbr;
+
+ //Left channel
+ for (j=0;j<Pstages*2;j++) {//Phasing routine
+ tmp=old.left()[j];
+ old.left()[j]=gl*tmp+inl;
+ inl=tmp-gl*old.left()[j];
+ };
+ //Right channel
+ for (j=0;j<Pstages*2;j++) {//Phasing routine
+ tmp=old.right()[j];
+ old.right()[j]=gr*tmp+inr;
+ inr=tmp-gr*old.right()[j];
+ };
+ //Left/Right crossing
+ REALTYPE l=inl;
+ REALTYPE r=inr;
+ inl=l*(1.0-lrcross)+r*lrcross;
+ inr=r*(1.0-lrcross)+l*lrcross;
+
+ fbl=inl*fb;
+ fbr=inr*fb;
+ efxoutl[i]=inl;
+ efxoutr[i]=inr;
};
oldgain=Stereo<REALTYPE>(lgain,rgain);
if (Poutsub!=0)
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
efxoutl[i]*= -1.0;
efxoutr[i]*= -1.0;
- };
+ };
};
/*
* Cleanup the effect
*/
-void Phaser::cleanup(){
- fbl=0.0;fbr=0.0;
+void Phaser::cleanup()
+{
+ fbl=0.0;
+ fbr=0.0;
//oldlgain=0.0;
//oldrgain=0.0;
oldgain=Stereo<REALTYPE>(0.0);
@@ -119,35 +125,41 @@ void Phaser::cleanup(){
/*
* Parameter control
*/
-void Phaser::setdepth(const unsigned char &Pdepth){
+void Phaser::setdepth(const unsigned char &Pdepth)
+{
this->Pdepth=Pdepth;
depth=(Pdepth/127.0);
};
-void Phaser::setfb(const unsigned char &Pfb){
+void Phaser::setfb(const unsigned char &Pfb)
+{
this->Pfb=Pfb;
fb=(Pfb-64.0)/64.1;
};
-void Phaser::setvolume(const unsigned char &Pvolume){
+void Phaser::setvolume(const unsigned char &Pvolume)
+{
this->Pvolume=Pvolume;
outvolume=Pvolume/127.0;
if (insertion==0) volume=1.0;
else volume=outvolume;
};
-void Phaser::setpanning(const unsigned char &Ppanning){
+void Phaser::setpanning(const unsigned char &Ppanning)
+{
this->Ppanning=Ppanning;
panning=Ppanning/127.0;
};
-void Phaser::setlrcross(const unsigned char &Plrcross){
+void Phaser::setlrcross(const unsigned char &Plrcross)
+{
this->Plrcross=Plrcross;
lrcross=Plrcross/127.0;
};
-void Phaser::setstages(const unsigned char &Pstages){
+void Phaser::setstages(const unsigned char &Pstages)
+{
//if (oldl!=NULL) delete [] oldl;
//if (oldr!=NULL) delete [] oldr;
if (Pstages>=MAX_PHASER_STAGES) this->Pstages=MAX_PHASER_STAGES-1;
@@ -158,96 +170,126 @@ void Phaser::setstages(const unsigned char &Pstages){
cleanup();
};
-void Phaser::setphase(const unsigned char &Pphase){
+void Phaser::setphase(const unsigned char &Pphase)
+{
this->Pphase=Pphase;
phase=(Pphase/127.0);
};
-void Phaser::setpreset(unsigned char npreset){
+void Phaser::setpreset(unsigned char npreset)
+{
const int PRESET_SIZE=12;
const int NUM_PRESETS=6;
unsigned char presets[NUM_PRESETS][PRESET_SIZE]={
- //Phaser1
- {64,64,36,0,0,64,110,64,1,0,0,20},
- //Phaser2
- {64,64,35,0,0,88,40,64,3,0,0,20},
- //Phaser3
- {64,64,31,0,0,66,68,107,2,0,0,20},
- //Phaser4
- {39,64,22,0,0,66,67,10,5,0,1,20},
- //Phaser5
- {64,64,20,0,1,110,67,78,10,0,0,20},
- //Phaser6
- {64,64,53,100,0,58,37,78,3,0,0,20}};
+ //Phaser1
+ {64,64,36,0,0,64,110,64,1,0,0,20},
+ //Phaser2
+ {64,64,35,0,0,88,40,64,3,0,0,20},
+ //Phaser3
+ {64,64,31,0,0,66,68,107,2,0,0,20},
+ //Phaser4
+ {39,64,22,0,0,66,67,10,5,0,1,20},
+ //Phaser5
+ {64,64,20,0,1,110,67,78,10,0,0,20},
+ //Phaser6
+ {64,64,53,100,0,58,37,78,3,0,0,20}
+ };
if (npreset>=NUM_PRESETS) npreset=NUM_PRESETS-1;
for (int n=0;n<PRESET_SIZE;n++) changepar(n,presets[npreset][n]);
Ppreset=npreset;
};
-void Phaser::changepar(const int &npar,const unsigned char &value){
- switch(npar){
- case 0: setvolume(value);
- break;
- case 1: setpanning(value);
- break;
- case 2: lfo.Pfreq=value;
- lfo.updateparams();
- break;
- case 3: lfo.Prandomness=value;
- lfo.updateparams();
- break;
- case 4: lfo.PLFOtype=value;
- lfo.updateparams();
- break;
- case 5: lfo.Pstereo=value;
- lfo.updateparams();
- break;
- case 6: setdepth(value);
- break;
- case 7: setfb(value);
- break;
- case 8: setstages(value);
- break;
- case 9: setlrcross(value);
- break;
- case 10:if (value>1) Poutsub=1;
- else Poutsub=value;
- break;
- case 11:setphase(value);
- break;
+void Phaser::changepar(const int &npar,const unsigned char &value)
+{
+ switch (npar) {
+ case 0:
+ setvolume(value);
+ break;
+ case 1:
+ setpanning(value);
+ break;
+ case 2:
+ lfo.Pfreq=value;
+ lfo.updateparams();
+ break;
+ case 3:
+ lfo.Prandomness=value;
+ lfo.updateparams();
+ break;
+ case 4:
+ lfo.PLFOtype=value;
+ lfo.updateparams();
+ break;
+ case 5:
+ lfo.Pstereo=value;
+ lfo.updateparams();
+ break;
+ case 6:
+ setdepth(value);
+ break;
+ case 7:
+ setfb(value);
+ break;
+ case 8:
+ setstages(value);
+ break;
+ case 9:
+ setlrcross(value);
+ break;
+ case 10:
+ if (value>1) Poutsub=1;
+ else Poutsub=value;
+ break;
+ case 11:
+ setphase(value);
+ break;
};
};
-unsigned char Phaser::getpar(const int &npar)const{
- switch (npar){
- case 0: return(Pvolume);
+unsigned char Phaser::getpar(const int &npar)const
+{
+ switch (npar) {
+ case 0:
+ return(Pvolume);
break;
- case 1: return(Ppanning);
+ case 1:
+ return(Ppanning);
break;
- case 2: return(lfo.Pfreq);
+ case 2:
+ return(lfo.Pfreq);
break;
- case 3: return(lfo.Prandomness);
+ case 3:
+ return(lfo.Prandomness);
break;
- case 4: return(lfo.PLFOtype);
+ case 4:
+ return(lfo.PLFOtype);
break;
- case 5: return(lfo.Pstereo);
+ case 5:
+ return(lfo.Pstereo);
break;
- case 6: return(Pdepth);
+ case 6:
+ return(Pdepth);
break;
- case 7: return(Pfb);
+ case 7:
+ return(Pfb);
break;
- case 8: return(Pstages);
+ case 8:
+ return(Pstages);
break;
- case 9: return(Plrcross);
+ case 9:
+ return(Plrcross);
break;
- case 10:return(Poutsub);
+ case 10:
+ return(Poutsub);
break;
- case 11:return(Pphase);
+ case 11:
+ return(Pphase);
break;
- default:return (0);
+ default:
+ return (0);
};
-
+
};
diff --git a/src/Effects/Phaser.h b/src/Effects/Phaser.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Phaser.h - Phaser effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -30,45 +30,46 @@
#define MAX_PHASER_STAGES 12
/**Phaser Effect*/
-class Phaser:public Effect {
- public:
- Phaser(const int &insetion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
- ~Phaser();
- void out(REALTYPE *smpsl,REALTYPE *smpsr);
- void setpreset(unsigned char npreset);
- void changepar(const int &npar,const unsigned char &value);
- unsigned char getpar(const int &npar)const;
- void cleanup();
- void setdryonly();
-
- private:
- //Parametrii Phaser
- EffectLFO lfo;/**<lfo-ul Phaser*/
- unsigned char Pvolume;
- unsigned char Ppanning;
- unsigned char Pdepth;/**<the depth of the Phaser*/
- unsigned char Pfb;/**<feedback*/
- unsigned char Plrcross;/**<feedback*/
- unsigned char Pstages;
- unsigned char Poutsub;/**<if I wish to substract the output instead of the adding it*/
- unsigned char Pphase;
-
- //Control Parametrii
- void setvolume(const unsigned char &Pvolume);
- void setpanning(const unsigned char &Ppanning);
- void setdepth(const unsigned char &Pdepth);
- void setfb(const unsigned char &Pfb);
- void setlrcross(const unsigned char &Plrcross);
- void setstages(const unsigned char &Pstages);
- void setphase(const unsigned char &Pphase);
-
- //Internal Values
- //int insertion; //inherited from Effect
- REALTYPE panning,fb,depth,lrcross,fbl,fbr,phase;
- //REALTYPE *oldl,*oldr;
- Stereo<AuSample> old;
- //REALTYPE oldlgain,oldrgain;
- Stereo<REALTYPE> oldgain;
+class Phaser:public Effect
+{
+public:
+ Phaser(const int &insetion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
+ ~Phaser();
+ void out(REALTYPE *smpsl,REALTYPE *smpsr);
+ void setpreset(unsigned char npreset);
+ void changepar(const int &npar,const unsigned char &value);
+ unsigned char getpar(const int &npar)const;
+ void cleanup();
+ void setdryonly();
+
+private:
+ //Parametrii Phaser
+ EffectLFO lfo;/**<lfo-ul Phaser*/
+ unsigned char Pvolume;
+ unsigned char Ppanning;
+ unsigned char Pdepth;/**<the depth of the Phaser*/
+ unsigned char Pfb;/**<feedback*/
+ unsigned char Plrcross;/**<feedback*/
+ unsigned char Pstages;
+ unsigned char Poutsub;/**<if I wish to substract the output instead of the adding it*/
+ unsigned char Pphase;
+
+ //Control Parametrii
+ void setvolume(const unsigned char &Pvolume);
+ void setpanning(const unsigned char &Ppanning);
+ void setdepth(const unsigned char &Pdepth);
+ void setfb(const unsigned char &Pfb);
+ void setlrcross(const unsigned char &Plrcross);
+ void setstages(const unsigned char &Pstages);
+ void setphase(const unsigned char &Pphase);
+
+ //Internal Values
+ //int insertion; //inherited from Effect
+ REALTYPE panning,fb,depth,lrcross,fbl,fbr,phase;
+ //REALTYPE *oldl,*oldr;
+ Stereo<AuSample> old;
+ //REALTYPE oldlgain,oldrgain;
+ Stereo<REALTYPE> oldgain;
};
#endif
diff --git a/src/Effects/Reverb.C b/src/Effects/Reverb.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Reverb.C - Reverberation effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,39 +26,42 @@
/**\todo: EarlyReflections,Prdelay,Perbalance */
Reverb::Reverb(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_)
- :Effect(insertion_,efxoutl_,efxoutr_,NULL,0)
+ :Effect(insertion_,efxoutl_,efxoutr_,NULL,0)
{
inputbuf=new REALTYPE[SOUND_BUFFER_SIZE];
//defaults
- Pvolume=48;
- Ppan=64;
+ Pvolume=48;
+ Ppan=64;
Ptime=64;
- Pidelay=40;
- Pidelayfb=0;
- Prdelay=0;
+ Pidelay=40;
+ Pidelayfb=0;
+ Prdelay=0;
Plpf=127;
- Phpf=0;
- Perbalance=64;
+ Phpf=0;
+ Perbalance=64;
Plohidamp=80;
Ptype=1;
- Proomsize=64;roomsize=1.0;rs=1.0;
+ Proomsize=64;
+ roomsize=1.0;
+ rs=1.0;
for (int i=0;i<REV_COMBS*2;i++) {
- comblen[i]=800+(int)(RND*1400);
- combk[i]=0;
- lpcomb[i]=0;
- combfb[i]=-0.97;
- comb[i]=NULL;
- };
+ comblen[i]=800+(int)(RND*1400);
+ combk[i]=0;
+ lpcomb[i]=0;
+ combfb[i]=-0.97;
+ comb[i]=NULL;
+ };
for (int i=0;i<REV_APS*2;i++) {
- aplen[i]=500+(int)(RND*500);
- apk[i]=0;
- ap[i]=NULL;
+ aplen[i]=500+(int)(RND*500);
+ apk[i]=0;
+ ap[i]=NULL;
};
-
- lpf=NULL;hpf=NULL;//no filter
+
+ lpf=NULL;
+ hpf=NULL;//no filter
idelay=NULL;
setpreset(Ppreset);
@@ -66,7 +69,8 @@ Reverb::Reverb(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_)
};
-Reverb::~Reverb(){
+Reverb::~Reverb()
+{
int i;
if (idelay!=NULL) delete []idelay;
if (hpf!=NULL) delete hpf;
@@ -81,14 +85,15 @@ Reverb::~Reverb(){
/*
* Cleanup the effect
*/
-void Reverb::cleanup(){
+void Reverb::cleanup()
+{
int i,j;
- for (i=0;i<REV_COMBS*2;i++){
+ for (i=0;i<REV_COMBS*2;i++) {
lpcomb[i]=0.0;
for (j=0;j<comblen[i];j++) comb[i][j]=0.0;
};
- for (i=0;i<REV_APS*2;i++)
+ for (i=0;i<REV_APS*2;i++)
for (j=0;j<aplen[i];j++) ap[i][j]=0.0;
if (idelay!=NULL) for (i=0;i<idelaylen;i++) idelay[i]=0.0;
@@ -101,61 +106,64 @@ void Reverb::cleanup(){
/*
* Process one channel; 0=left,1=right
*/
-void Reverb::processmono(int ch,REALTYPE *output){
+void Reverb::processmono(int ch,REALTYPE *output)
+{
int i,j;
REALTYPE fbout,tmp;
/**\todo: implement the high part from lohidamp*/
-
- for (j=REV_COMBS*ch;j<REV_COMBS*(ch+1);j++){
- int ck=combk[j];
- int comblength=comblen[j];
- REALTYPE lpcombj=lpcomb[j];
+ for (j=REV_COMBS*ch;j<REV_COMBS*(ch+1);j++) {
+
+ int ck=combk[j];
+ int comblength=comblen[j];
+ REALTYPE lpcombj=lpcomb[j];
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
fbout=comb[j][ck]*combfb[j];
fbout=fbout*(1.0-lohifb)+lpcombj*lohifb;
- lpcombj=fbout;
+ lpcombj=fbout;
- comb[j][ck]=inputbuf[i]+fbout;
- output[i]+=fbout;
+ comb[j][ck]=inputbuf[i]+fbout;
+ output[i]+=fbout;
- if ((++ck)>=comblength) ck=0;
- };
+ if ((++ck)>=comblength) ck=0;
+ };
- combk[j]=ck;
- lpcomb[j]=lpcombj;
+ combk[j]=ck;
+ lpcomb[j]=lpcombj;
};
- for (j=REV_APS*ch;j<REV_APS*(1+ch);j++){
- int ak=apk[j];
- int aplength=aplen[j];
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- tmp=ap[j][ak];
- ap[j][ak]=0.7*tmp+output[i];
- output[i]=tmp-0.7*ap[j][ak];
- if ((++ak)>=aplength) ak=0;
- };
- apk[j]=ak;
+ for (j=REV_APS*ch;j<REV_APS*(1+ch);j++) {
+ int ak=apk[j];
+ int aplength=aplen[j];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmp=ap[j][ak];
+ ap[j][ak]=0.7*tmp+output[i];
+ output[i]=tmp-0.7*ap[j][ak];
+ if ((++ak)>=aplength) ak=0;
+ };
+ apk[j]=ak;
};
};
/*
* Effect output
*/
-void Reverb::out(REALTYPE *smps_l, REALTYPE *smps_r){
+void Reverb::out(REALTYPE *smps_l, REALTYPE *smps_r)
+{
int i;
if ((Pvolume==0)&&(insertion!=0)) return;
-
+
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- inputbuf[i]=(smps_l[i]+smps_r[i])/2.0;
- //Initial delay r
- if (idelay!=NULL){
+ inputbuf[i]=(smps_l[i]+smps_r[i])/2.0;
+ //Initial delay r
+ if (idelay!=NULL) {
REALTYPE tmp=inputbuf[i]+idelay[idelayk]*idelayfb;
inputbuf[i]=idelay[idelayk];
- idelay[idelayk]=tmp;
- idelayk++;if (idelayk>=idelaylen) idelayk=0;
- };
+ idelay[idelayk]=tmp;
+ idelayk++;
+ if (idelayk>=idelaylen) idelayk=0;
+ };
};
if (lpf!=NULL) lpf->filterout(inputbuf);
@@ -163,15 +171,16 @@ void Reverb::out(REALTYPE *smps_l, REALTYPE *smps_r){
processmono(0,efxoutl);//left
processmono(1,efxoutr);//right
-
+
REALTYPE lvol=rs/REV_COMBS*pan;
REALTYPE rvol=rs/REV_COMBS*(1.0-pan);
- if (insertion!=0){
- lvol*=2;rvol*=2;
+ if (insertion!=0) {
+ lvol*=2;
+ rvol*=2;
};
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- efxoutl[i]*=lvol;
- efxoutr[i]*=rvol;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ efxoutl[i]*=lvol;
+ efxoutr[i]*=rvol;
};
};
@@ -179,188 +188,198 @@ void Reverb::out(REALTYPE *smps_l, REALTYPE *smps_r){
/*
* Parameter control
*/
-void Reverb::setvolume(const unsigned char &Pvolume){
+void Reverb::setvolume(const unsigned char &Pvolume)
+{
this->Pvolume=Pvolume;
if (insertion==0) {
- outvolume=pow(0.01,(1.0-Pvolume/127.0))*4.0;
- volume=1.0;
+ outvolume=pow(0.01,(1.0-Pvolume/127.0))*4.0;
+ volume=1.0;
} else {
- volume=outvolume=Pvolume/127.0;
- if (Pvolume==0) cleanup();
+ volume=outvolume=Pvolume/127.0;
+ if (Pvolume==0) cleanup();
};
};
-void Reverb::setpan(const unsigned char &Ppan){
+void Reverb::setpan(const unsigned char &Ppan)
+{
this->Ppan=Ppan;
pan=(REALTYPE)Ppan/127.0;
};
-void Reverb::settime(const unsigned char &Ptime){
+void Reverb::settime(const unsigned char &Ptime)
+{
int i;
REALTYPE t;
this->Ptime=Ptime;
t=pow(60.0,(REALTYPE)Ptime/127.0)-0.97;
- for (i=0;i<REV_COMBS*2;i++){
- combfb[i]=-exp((REALTYPE)comblen[i]/(REALTYPE)SAMPLE_RATE*log(0.001)/t);
- //the feedback is negative because it removes the DC
+ for (i=0;i<REV_COMBS*2;i++) {
+ combfb[i]=-exp((REALTYPE)comblen[i]/(REALTYPE)SAMPLE_RATE*log(0.001)/t);
+ //the feedback is negative because it removes the DC
};
};
-void Reverb::setlohidamp(unsigned char Plohidamp){
- REALTYPE x;
-
+void Reverb::setlohidamp(unsigned char Plohidamp)
+{
+ REALTYPE x;
+
if (Plohidamp<64) Plohidamp=64;//remove this when the high part from lohidamp will be added
-
+
this->Plohidamp=Plohidamp;
if (Plohidamp==64) {
- lohidamptype=0;
- lohifb=0.0;
+ lohidamptype=0;
+ lohifb=0.0;
} else {
if (Plohidamp<64) lohidamptype=1;
- if (Plohidamp>64) lohidamptype=2;
+ if (Plohidamp>64) lohidamptype=2;
x=fabs((REALTYPE)(Plohidamp-64)/64.1);
lohifb=x*x;
};
};
-void Reverb::setidelay(const unsigned char &Pidelay){
+void Reverb::setidelay(const unsigned char &Pidelay)
+{
REALTYPE delay;
this->Pidelay=Pidelay;
delay=pow(50*Pidelay/127.0,2)-1.0;
-
+
if (idelay!=NULL) delete []idelay;
idelay=NULL;
-
+
idelaylen=(int) (SAMPLE_RATE*delay/1000);
if (idelaylen>1) {
- idelayk=0;
- idelay=new REALTYPE[idelaylen];
+ idelayk=0;
+ idelay=new REALTYPE[idelaylen];
for (int i=0;i<idelaylen;i++) idelay[i]=0.0;
};
};
-void Reverb::setidelayfb(const unsigned char &Pidelayfb){
+void Reverb::setidelayfb(const unsigned char &Pidelayfb)
+{
this->Pidelayfb=Pidelayfb;
idelayfb=Pidelayfb/128.0;
};
-void Reverb::sethpf(const unsigned char &Phpf){
+void Reverb::sethpf(const unsigned char &Phpf)
+{
this->Phpf=Phpf;
if (Phpf==0) {//No HighPass
if (hpf!=NULL) delete hpf;
hpf=NULL;
- }
- else{
+ } else {
REALTYPE fr=exp(pow(Phpf/127.0,0.5)*log(10000.0))+20.0;
- if (hpf==NULL) hpf=new AnalogFilter(3,fr,1,0);
- else hpf->setfreq(fr);
+ if (hpf==NULL) hpf=new AnalogFilter(3,fr,1,0);
+ else hpf->setfreq(fr);
};
};
-void Reverb::setlpf(const unsigned char &Plpf){
+void Reverb::setlpf(const unsigned char &Plpf)
+{
this->Plpf=Plpf;
if (Plpf==127) {//No LowPass
if (lpf!=NULL) delete lpf;
lpf=NULL;
- }
- else{
+ } else {
REALTYPE fr=exp(pow(Plpf/127.0,0.5)*log(25000.0))+40;
- if (lpf==NULL) lpf=new AnalogFilter(2,fr,1,0);
- else lpf->setfreq(fr);
+ if (lpf==NULL) lpf=new AnalogFilter(2,fr,1,0);
+ else lpf->setfreq(fr);
};
};
-void Reverb::settype(unsigned char Ptype){
+void Reverb::settype(unsigned char Ptype)
+{
const int NUM_TYPES=2;
int combtunings[NUM_TYPES][REV_COMBS]={
- //this is unused (for random)
- {0,0,0,0,0,0,0,0},
- //Freeverb by Jezar at Dreampoint
- {1116,1188,1277,1356,1422,1491,1557,1617}
+ //this is unused (for random)
+ {0,0,0,0,0,0,0,0},
+ //Freeverb by Jezar at Dreampoint
+ {1116,1188,1277,1356,1422,1491,1557,1617}
};
int aptunings[NUM_TYPES][REV_APS]={
- //this is unused (for random)
- {0,0,0,0},
- //Freeverb by Jezar at Dreampoint
- {225,341,441,556}
+ //this is unused (for random)
+ {0,0,0,0},
+ //Freeverb by Jezar at Dreampoint
+ {225,341,441,556}
};
if (Ptype>=NUM_TYPES) Ptype=NUM_TYPES-1;
this->Ptype=Ptype;
-
+
REALTYPE tmp;
for (int i=0;i<REV_COMBS*2;i++) {
- if (Ptype==0) tmp=800.0+(int)(RND*1400.0);
+ if (Ptype==0) tmp=800.0+(int)(RND*1400.0);
else tmp=combtunings[Ptype][i%REV_COMBS];
- tmp*=roomsize;
- if (i>REV_COMBS) tmp+=23.0;
- tmp*=SAMPLE_RATE/44100.0;//adjust the combs according to the samplerate
- if (tmp<10) tmp=10;
-
- comblen[i]=(int) tmp;
- combk[i]=0;
- lpcomb[i]=0;
- if (comb[i]!=NULL) delete []comb[i];
- comb[i]=new REALTYPE[comblen[i]];
- };
+ tmp*=roomsize;
+ if (i>REV_COMBS) tmp+=23.0;
+ tmp*=SAMPLE_RATE/44100.0;//adjust the combs according to the samplerate
+ if (tmp<10) tmp=10;
+
+ comblen[i]=(int) tmp;
+ combk[i]=0;
+ lpcomb[i]=0;
+ if (comb[i]!=NULL) delete []comb[i];
+ comb[i]=new REALTYPE[comblen[i]];
+ };
for (int i=0;i<REV_APS*2;i++) {
- if (Ptype==0) tmp=500+(int)(RND*500);
+ if (Ptype==0) tmp=500+(int)(RND*500);
else tmp=aptunings[Ptype][i%REV_APS];
- tmp*=roomsize;
- if (i>REV_APS) tmp+=23.0;
- tmp*=SAMPLE_RATE/44100.0;//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 REALTYPE[aplen[i]];
+ tmp*=roomsize;
+ if (i>REV_APS) tmp+=23.0;
+ tmp*=SAMPLE_RATE/44100.0;//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 REALTYPE[aplen[i]];
};
settime(Ptime);
cleanup();
};
-void Reverb::setroomsize(const unsigned char &Proomsize){
+void Reverb::setroomsize(const unsigned char &Proomsize)
+{
this->Proomsize=Proomsize;
if (Proomsize==0) this->Proomsize=64;//this is because the older versions consider roomsize=0
roomsize=(this->Proomsize-64.0)/64.0;
if (roomsize>0.0) roomsize*=2.0;
roomsize=pow(10.0,roomsize);
rs=sqrt(roomsize);
- settype(Ptype);
+ settype(Ptype);
};
-void Reverb::setpreset(unsigned char npreset){
+void Reverb::setpreset(unsigned char npreset)
+{
const int PRESET_SIZE=12;
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},
+ //Cathedral1
+ {80,64,63,24,0,0,0,85,5,83,1,64},
//Cathedral2
- {80,64,69,35,0,0,0,127,0,71,0,64},
- //Cathedral3
- {80,64,69,24,0,0,0,127,75,78,1,85},
- //Hall1
- {90,64,51,10,0,0,0,127,21,78,1,64},
- //Hall2
- {90,64,53,20,0,0,0,127,75,71,1,64},
- //Room1
- {100,64,33,0,0,0,0,127,0,106,0,30},
- //Room2
- {100,64,21,26,0,0,0,62,0,77,1,45},
- //Basement
- {110,64,14,0,0,0,0,127,5,71,0,25},
- //Tunnel
- {85,80,84,20,42,0,0,51,0,78,1,105},
- //Echoed1
- {95,64,26,60,71,0,0,114,0,64,1,64},
- //Echoed2
- {90,64,40,88,71,0,0,114,0,88,1,64},
- //VeryLong1
- {90,64,93,15,0,0,0,114,0,77,0,95},
- //VeryLong2
- {90,64,111,30,0,0,0,114,90,74,1,80}};
+ {80,64,69,35,0,0,0,127,0,71,0,64},
+ //Cathedral3
+ {80,64,69,24,0,0,0,127,75,78,1,85},
+ //Hall1
+ {90,64,51,10,0,0,0,127,21,78,1,64},
+ //Hall2
+ {90,64,53,20,0,0,0,127,75,71,1,64},
+ //Room1
+ {100,64,33,0,0,0,0,127,0,106,0,30},
+ //Room2
+ {100,64,21,26,0,0,0,62,0,77,1,45},
+ //Basement
+ {110,64,14,0,0,0,0,127,5,71,0,25},
+ //Tunnel
+ {85,80,84,20,42,0,0,51,0,78,1,105},
+ //Echoed1
+ {95,64,26,60,71,0,0,114,0,64,1,64},
+ //Echoed2
+ {90,64,40,88,71,0,0,114,0,88,1,64},
+ //VeryLong1
+ {90,64,93,15,0,0,0,114,0,77,0,95},
+ //VeryLong2
+ {90,64,111,30,0,0,0,114,90,74,1,80}
+ };
if (npreset>=NUM_PRESETS) npreset=NUM_PRESETS-1;
for (int n=0;n<PRESET_SIZE;n++) changepar(n,presets[npreset][n]);
@@ -369,60 +388,82 @@ void Reverb::setpreset(unsigned char npreset){
};
-void Reverb::changepar(const int &npar,const unsigned char &value){
- switch (npar){
- case 0: setvolume(value);
+void Reverb::changepar(const int &npar,const unsigned char &value)
+{
+ switch (npar) {
+ case 0:
+ setvolume(value);
break;
- case 1: setpan(value);
+ case 1:
+ setpan(value);
break;
- case 2: settime(value);
+ case 2:
+ settime(value);
break;
- case 3: setidelay(value);
+ case 3:
+ setidelay(value);
break;
- case 4: setidelayfb(value);
+ case 4:
+ setidelayfb(value);
break;
// case 5: setrdelay(value);
// break;
// case 6: seterbalance(value);
// break;
- case 7: setlpf(value);
+ case 7:
+ setlpf(value);
break;
- case 8: sethpf(value);
+ case 8:
+ sethpf(value);
break;
- case 9: setlohidamp(value);
+ case 9:
+ setlohidamp(value);
break;
- case 10:settype(value);
+ case 10:
+ settype(value);
break;
- case 11:setroomsize(value);
+ case 11:
+ setroomsize(value);
break;
};
};
-unsigned char Reverb::getpar(const int &npar)const{
- switch (npar){
- case 0: return(Pvolume);
+unsigned char Reverb::getpar(const int &npar)const
+{
+ switch (npar) {
+ case 0:
+ return(Pvolume);
break;
- case 1: return(Ppan);
+ case 1:
+ return(Ppan);
break;
- case 2: return(Ptime);
+ case 2:
+ return(Ptime);
break;
- case 3: return(Pidelay);
+ case 3:
+ return(Pidelay);
break;
- case 4: return(Pidelayfb);
+ case 4:
+ return(Pidelayfb);
break;
// case 5: return(Prdelay);
// break;
// case 6: return(Perbalance);
// break;
- case 7: return(Plpf);
+ case 7:
+ return(Plpf);
break;
- case 8: return(Phpf);
+ case 8:
+ return(Phpf);
break;
- case 9: return(Plohidamp);
+ case 9:
+ return(Plohidamp);
break;
- case 10:return(Ptype);
+ case 10:
+ return(Ptype);
break;
- case 11:return(Proomsize);
+ case 11:
+ return(Proomsize);
break;
};
return(0);//in case of bogus "parameter"
diff --git a/src/Effects/Reverb.h b/src/Effects/Reverb.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Reverb.h - Reverberation effect
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -32,18 +32,19 @@
#define REV_APS 4
/**Creates Reverberation Effects*/
-class Reverb:public Effect {
- public:
+class Reverb:public Effect
+{
+public:
Reverb(const int &insertion_,REALTYPE *efxoutl_,REALTYPE *efxoutr_);
~Reverb();
void out(REALTYPE *smps_l,REALTYPE *smps_r);
void cleanup();
- void setpreset(unsigned char npreset);
+ void setpreset(unsigned char npreset);
void changepar(const int &npar,const unsigned char &value);
unsigned char getpar(const int &npar)const;
- private:
+private:
//Parametrii
/**Amount of the reverb*/
unsigned char Pvolume;
@@ -93,32 +94,32 @@ class Reverb:public Effect {
void setlpf(const unsigned char &Plpf);
void settype( unsigned char Ptype);
void setroomsize(const unsigned char &Proomsize);
-
+
REALTYPE pan,erbalance;
- //Parametrii 2
+ //Parametrii 2
int lohidamptype;/**<0=disable,1=highdamp(lowpass),2=lowdamp(highpass)*/
int idelaylen,rdelaylen;
int idelayk;
REALTYPE lohifb,idelayfb,roomsize,rs;//rs is used to "normalise" the volume according to the roomsize
- int comblen[REV_COMBS*2];
+ int comblen[REV_COMBS*2];
int aplen[REV_APS*2];
-
+
//Internal Variables
-
+
REALTYPE *comb[REV_COMBS*2];
-
+
int combk[REV_COMBS*2];
REALTYPE combfb[REV_COMBS*2];/**<feedback-ul fiecarui filtru "comb"*/
REALTYPE lpcomb[REV_COMBS*2];/**<pentru Filtrul LowPass*/
REALTYPE *ap[REV_APS*2];
-
+
int apk[REV_APS*2];
-
+
REALTYPE *idelay;
AnalogFilter *lpf,*hpf;//filters
REALTYPE *inputbuf;
-
+
void processmono(int ch,REALTYPE *output);
};
diff --git a/src/Input/ALSAMidiIn.C b/src/Input/ALSAMidiIn.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
ALSAMidiIn.C - Midi input for ALSA (this creates an ALSA virtual port)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -22,25 +22,27 @@
#include "ALSAMidiIn.h"
-ALSAMidiIn::ALSAMidiIn(){
+ALSAMidiIn::ALSAMidiIn()
+{
int alsaport;
inputok=false;
midi_handle=NULL;
-
+
if (snd_seq_open(&midi_handle,"default",SND_SEQ_OPEN_INPUT,0)!=0) return;
-
+
snd_seq_set_client_name(midi_handle,"ZynAddSubFX");//thanks to Frank Neumann
alsaport = snd_seq_create_simple_port(midi_handle,"ZynAddSubFX"
- ,SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE
- ,SND_SEQ_PORT_TYPE_SYNTH);
+ ,SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE
+ ,SND_SEQ_PORT_TYPE_SYNTH);
if (alsaport<0) return;
inputok=true;
};
-ALSAMidiIn::~ALSAMidiIn(){
+ALSAMidiIn::~ALSAMidiIn()
+{
if (midi_handle) snd_seq_close(midi_handle);
};
@@ -48,51 +50,53 @@ ALSAMidiIn::~ALSAMidiIn(){
/*
* Get the midi command,channel and parameters
*/
-void ALSAMidiIn::getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams){
+void ALSAMidiIn::getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams)
+{
snd_seq_event_t *midievent=NULL;
cmdtype=MidiNull;
- if (inputok==false){
- /* The input is broken. We need to block for a while anyway so other
- non-RT threads get a chance to run. */
- sleep(1);
- return;
+ if (inputok==false) {
+ /* The input is broken. We need to block for a while anyway so other
+ non-RT threads get a chance to run. */
+ sleep(1);
+ return;
};
-
+
snd_seq_event_input(midi_handle,&midievent);
-
+
if (midievent==NULL) return;
- switch (midievent->type){
- case SND_SEQ_EVENT_NOTEON:
- cmdtype=MidiNoteON;
- cmdchan=midievent->data.note.channel;
- cmdparams[0]=midievent->data.note.note;
- cmdparams[1]=midievent->data.note.velocity;
- break;
- case SND_SEQ_EVENT_NOTEOFF:
- cmdtype=MidiNoteOFF;
- cmdchan=midievent->data.note.channel;
- cmdparams[0]=midievent->data.note.note;
- break;
- case SND_SEQ_EVENT_PITCHBEND:
- cmdtype=MidiController;
- cmdchan=midievent->data.control.channel;
- cmdparams[0]=C_pitchwheel;//Pitch Bend
- cmdparams[1]=midievent->data.control.value;
- break;
- case SND_SEQ_EVENT_CONTROLLER:
- cmdtype=MidiController;
- cmdchan=midievent->data.control.channel;
- cmdparams[0]=getcontroller(midievent->data.control.param);
- cmdparams[1]=midievent->data.control.value;
- //fprintf(stderr,"t=%d val=%d\n",midievent->data.control.param,midievent->data.control.value);
- break;
-
+ switch (midievent->type) {
+ case SND_SEQ_EVENT_NOTEON:
+ cmdtype=MidiNoteON;
+ cmdchan=midievent->data.note.channel;
+ cmdparams[0]=midievent->data.note.note;
+ cmdparams[1]=midievent->data.note.velocity;
+ break;
+ case SND_SEQ_EVENT_NOTEOFF:
+ cmdtype=MidiNoteOFF;
+ cmdchan=midievent->data.note.channel;
+ cmdparams[0]=midievent->data.note.note;
+ break;
+ case SND_SEQ_EVENT_PITCHBEND:
+ cmdtype=MidiController;
+ cmdchan=midievent->data.control.channel;
+ cmdparams[0]=C_pitchwheel;//Pitch Bend
+ cmdparams[1]=midievent->data.control.value;
+ break;
+ case SND_SEQ_EVENT_CONTROLLER:
+ cmdtype=MidiController;
+ cmdchan=midievent->data.control.channel;
+ cmdparams[0]=getcontroller(midievent->data.control.param);
+ cmdparams[1]=midievent->data.control.value;
+ //fprintf(stderr,"t=%d val=%d\n",midievent->data.control.param,midievent->data.control.value);
+ break;
+
};
};
-int ALSAMidiIn::getalsaid() {
+int ALSAMidiIn::getalsaid()
+{
if (midi_handle) {
snd_seq_client_info_t* seq_info;
snd_seq_client_info_malloc(&seq_info);
diff --git a/src/Input/ALSAMidiIn.h b/src/Input/ALSAMidiIn.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
ALSAMidiIn.h - Midi input for ALSA (this creates an ALSA virtual port)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,20 +28,21 @@
/**Midi input for ALSA (this creates an ALSA virtual port)*/
-class ALSAMidiIn:public MidiIn{
- public:
- /**Constructor*/
- ALSAMidiIn();
- /**Destructor*/
- ~ALSAMidiIn();
-
- void getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams);
- /**Get the ALSA id
- * @return ALSA id*/
- int getalsaid();
-
- private:
- snd_seq_t *midi_handle;
+class ALSAMidiIn:public MidiIn
+{
+public:
+ /**Constructor*/
+ ALSAMidiIn();
+ /**Destructor*/
+ ~ALSAMidiIn();
+
+ void getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams);
+ /**Get the ALSA id
+ * @return ALSA id*/
+ int getalsaid();
+
+private:
+ snd_seq_t *midi_handle;
};
#endif
diff --git a/src/Input/MidiIn.C b/src/Input/MidiIn.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
MidiIn.C - This class is inherited by all the Midi input classes
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -23,52 +23,73 @@
#include "../globals.h"
#include "MidiIn.h"
-int MidiIn::getcontroller(unsigned char b){
+int MidiIn::getcontroller(unsigned char b)
+{
/**\todo there might be a better way to do this*/
int ctl=C_NULL;
- switch (b){
- case 1:ctl=C_modwheel;//Modulation Wheel
- break;
- case 7:ctl=C_volume;//Volume
- break;
- case 10:ctl=C_panning;//Panning
- break;
- case 11:ctl=C_expression;//Expression
- break;
- case 64:ctl=C_sustain;//Sustain pedal
- break;
- case 65:ctl=C_portamento;//Portamento
- break;
- case 71:ctl=C_filterq;//Filter Q (Sound Timbre)
- break;
- case 74:ctl=C_filtercutoff;//Filter Cutoff (Brightness)
- break;
- case 75:ctl=C_bandwidth;//BandWidth
- break;
- case 76:ctl=C_fmamp;//FM amplitude
- break;
- case 77:ctl=C_resonance_center;//Resonance Center Frequency
- break;
- case 78:ctl=C_resonance_bandwidth;//Resonance Bandwith
- break;
- case 120:ctl=C_allsoundsoff;//All Sounds OFF
- break;
- case 121:ctl=C_resetallcontrollers;//Reset All Controllers
- break;
- case 123:ctl=C_allnotesoff;//All Notes OFF
- break;
- //RPN and NRPN
- case 0x06:ctl=C_dataentryhi;//Data Entry (Coarse)
- break;
- case 0x26:ctl=C_dataentrylo;//Data Entry (Fine)
- break;
- case 99:ctl=C_nrpnhi;//NRPN (Coarse)
- break;
- case 98:ctl=C_nrpnlo;//NRPN (Fine)
- break;
- default:ctl=C_NULL;//unknown controller
- //fprintf(stderr,"Controller=%d , par=%d\n",midievent->data.control.param,cmdparams[1]);
- break;
- };
+ switch (b) {
+ case 1:
+ ctl=C_modwheel;//Modulation Wheel
+ break;
+ case 7:
+ ctl=C_volume;//Volume
+ break;
+ case 10:
+ ctl=C_panning;//Panning
+ break;
+ case 11:
+ ctl=C_expression;//Expression
+ break;
+ case 64:
+ ctl=C_sustain;//Sustain pedal
+ break;
+ case 65:
+ ctl=C_portamento;//Portamento
+ break;
+ case 71:
+ ctl=C_filterq;//Filter Q (Sound Timbre)
+ break;
+ case 74:
+ ctl=C_filtercutoff;//Filter Cutoff (Brightness)
+ break;
+ case 75:
+ ctl=C_bandwidth;//BandWidth
+ break;
+ case 76:
+ ctl=C_fmamp;//FM amplitude
+ break;
+ case 77:
+ ctl=C_resonance_center;//Resonance Center Frequency
+ break;
+ case 78:
+ ctl=C_resonance_bandwidth;//Resonance Bandwith
+ break;
+ case 120:
+ ctl=C_allsoundsoff;//All Sounds OFF
+ break;
+ case 121:
+ ctl=C_resetallcontrollers;//Reset All Controllers
+ break;
+ case 123:
+ ctl=C_allnotesoff;//All Notes OFF
+ break;
+ //RPN and NRPN
+ case 0x06:
+ ctl=C_dataentryhi;//Data Entry (Coarse)
+ break;
+ case 0x26:
+ ctl=C_dataentrylo;//Data Entry (Fine)
+ break;
+ case 99:
+ ctl=C_nrpnhi;//NRPN (Coarse)
+ break;
+ case 98:
+ ctl=C_nrpnlo;//NRPN (Fine)
+ break;
+ default:
+ ctl=C_NULL;//unknown controller
+ //fprintf(stderr,"Controller=%d , par=%d\n",midievent->data.control.param,cmdparams[1]);
+ break;
+ };
return(ctl);
};
diff --git a/src/Input/MidiIn.h b/src/Input/MidiIn.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
MidiIn.h - This class is inherited by all the Midi input classes
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,22 +25,23 @@
#include "../globals.h"
-enum MidiCmdType{MidiNull,MidiNoteOFF,MidiNoteON,MidiController};
+enum MidiCmdType {MidiNull,MidiNoteOFF,MidiNoteON,MidiController};
#define MP_MAX_BYTES 4000 //in case of loooong SYS_EXes
/**This class is inherited by all the Midi input classes*/
-class MidiIn{
- public:
- /**Get the command,channel and parameters of the MIDI
- *
- * \todo make pure virtual
- * @param cmdtype the referece to the variable that will store the type
- * @param cmdchan the channel for the event
- * @param parameters for the event*/
- virtual void getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams)=0;
- int getcontroller(unsigned char b);
- protected:
- bool inputok;/**<1 if I can read midi bytes from input ports*/
+class MidiIn
+{
+public:
+ /**Get the command,channel and parameters of the MIDI
+ *
+ * \todo make pure virtual
+ * @param cmdtype the referece to the variable that will store the type
+ * @param cmdchan the channel for the event
+ * @param parameters for the event*/
+ virtual void getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams)=0;
+ int getcontroller(unsigned char b);
+protected:
+ bool inputok;/**<1 if I can read midi bytes from input ports*/
};
#endif
diff --git a/src/Input/NULLMidiIn.C b/src/Input/NULLMidiIn.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
NULLMidiIn.C - a dummy Midi port
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -22,17 +22,20 @@
#include "NULLMidiIn.h"
-NULLMidiIn::NULLMidiIn(){
+NULLMidiIn::NULLMidiIn()
+{
};
-NULLMidiIn::~NULLMidiIn(){
+NULLMidiIn::~NULLMidiIn()
+{
};
/*
* Get the midi command,channel and parameters
* It returns MidiNull because it is a dummy driver
*/
-void NULLMidiIn::getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,unsigned char *cmdparams){
+void NULLMidiIn::getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,unsigned char *cmdparams)
+{
cmdtype=MidiNull;
};
diff --git a/src/Input/NULLMidiIn.h b/src/Input/NULLMidiIn.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
NULLMidiIn.h - a dummy Midi port
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,20 +27,21 @@
/**a dummy Midi port*/
-class NULLMidiIn:public MidiIn{
- public:
- /**Dummy Constructor
- * \todo see if the default constructor would work here*/
- NULLMidiIn();
- /**Dummy Destructor
- * \todo see if the default destructor would work here*/
- ~NULLMidiIn();
- /**Get the midi command,channel and parameters
- * It returns MidiNull because it is a dummy driver
- */
- void getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,unsigned char *cmdparams);
-
- private:
+class NULLMidiIn:public MidiIn
+{
+public:
+ /**Dummy Constructor
+ * \todo see if the default constructor would work here*/
+ NULLMidiIn();
+ /**Dummy Destructor
+ * \todo see if the default destructor would work here*/
+ ~NULLMidiIn();
+ /**Get the midi command,channel and parameters
+ * It returns MidiNull because it is a dummy driver
+ */
+ void getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,unsigned char *cmdparams);
+
+private:
};
#endif
diff --git a/src/Input/OSSMidiIn.C b/src/Input/OSSMidiIn.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
OSSMidiIn.C - Midi input for Open Sound System
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -31,31 +31,35 @@
#include "OSSMidiIn.h"
#include "../Misc/Util.h"
-OSSMidiIn::OSSMidiIn(){
+OSSMidiIn::OSSMidiIn()
+{
inputok=false;
midi_handle=open(config.cfg.LinuxOSSSeqInDev,O_RDONLY,0);
if (midi_handle!=-1) inputok=true;
-
+
lastmidicmd=0;
cmdtype=0;
cmdchan=0;
};
-OSSMidiIn::~OSSMidiIn(){
+OSSMidiIn::~OSSMidiIn()
+{
close(midi_handle);
};
-unsigned char OSSMidiIn::readbyte(){
+unsigned char OSSMidiIn::readbyte()
+{
unsigned char tmp[4];
read(midi_handle,&tmp[0],1);
- while (tmp[0]!=SEQ_MIDIPUTC){
+ while (tmp[0]!=SEQ_MIDIPUTC) {
read(midi_handle,&tmp[0],4);
}
return(tmp[1]);
};
-unsigned char OSSMidiIn::getmidibyte(){
+unsigned char OSSMidiIn::getmidibyte()
+{
unsigned char b;
do {
b=readbyte();
@@ -66,14 +70,15 @@ unsigned char OSSMidiIn::getmidibyte(){
/*
* Get the midi command,channel and parameters
*/
-void OSSMidiIn::getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams){
+void OSSMidiIn::getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams)
+{
unsigned char tmp,i;
if (inputok==false) {
cmdtype=MidiNull;
return;
}
i=0;
- if (lastmidicmd==0){//asteapta prima data pana cand vine prima comanda midi
+ if (lastmidicmd==0) {//asteapta prima data pana cand vine prima comanda midi
while (tmp<0x80) tmp=getmidibyte();
lastmidicmd=tmp;
}
@@ -85,26 +90,26 @@ void OSSMidiIn::getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdp
tmp=getmidibyte();
}
- if ((lastmidicmd>=0x80)&&(lastmidicmd<=0x8f)){//Note OFF
+ if ((lastmidicmd>=0x80)&&(lastmidicmd<=0x8f)) {//Note OFF
cmdtype=MidiNoteOFF;
cmdchan=lastmidicmd%16;
cmdparams[0]=tmp;//note number
}
- if ((lastmidicmd>=0x90)&&(lastmidicmd<=0x9f)){//Note ON
+ if ((lastmidicmd>=0x90)&&(lastmidicmd<=0x9f)) {//Note ON
cmdtype=MidiNoteON;
cmdchan=lastmidicmd%16;
- cmdparams[0]=tmp;//note number
+ cmdparams[0]=tmp;//note number
cmdparams[1]=getmidibyte();//velocity
if (cmdparams[1]==0) cmdtype=MidiNoteOFF;//if velocity==0 then is note off
}
- if ((lastmidicmd>=0xB0)&&(lastmidicmd<=0xBF)){//Controllers
+ if ((lastmidicmd>=0xB0)&&(lastmidicmd<=0xBF)) {//Controllers
cmdtype=MidiController;
cmdchan=lastmidicmd%16;
cmdparams[0]=getcontroller(tmp);
cmdparams[1]=getmidibyte();
}
- if ((lastmidicmd>=0xE0)&&(lastmidicmd<=0xEF)){//Pitch Wheel
+ if ((lastmidicmd>=0xE0)&&(lastmidicmd<=0xEF)) {//Pitch Wheel
cmdtype=MidiController;
cmdchan=lastmidicmd%16;
cmdparams[0]=C_pitchwheel;
diff --git a/src/Input/OSSMidiIn.h b/src/Input/OSSMidiIn.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
OSSMidiIn.h - Midi input for Open Sound System
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,21 +25,22 @@
#include "MidiIn.h"
-class OSSMidiIn:public MidiIn{
- public:
- OSSMidiIn();
- ~OSSMidiIn();
- unsigned char getmidibyte();
- unsigned char readbyte();
-
- //Midi parser
- void getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams);
- unsigned char cmdtype;//the Message Type (noteon,noteof,sysex..)
- unsigned char cmdchan;//the channel number
-
- private:
- int midi_handle;
- unsigned char lastmidicmd;//last byte (>=80) received from the Midi
+class OSSMidiIn:public MidiIn
+{
+public:
+ OSSMidiIn();
+ ~OSSMidiIn();
+ unsigned char getmidibyte();
+ unsigned char readbyte();
+
+ //Midi parser
+ void getmidicmd(MidiCmdType &cmdtype,unsigned char &cmdchan,int *cmdparams);
+ unsigned char cmdtype;//the Message Type (noteon,noteof,sysex..)
+ unsigned char cmdchan;//the channel number
+
+private:
+ int midi_handle;
+ unsigned char lastmidicmd;//last byte (>=80) received from the Midi
};
diff --git a/src/Input/WINMidiIn.C b/src/Input/WINMidiIn.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
WINMidiIn.C - Midi input for Windows
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -35,49 +35,53 @@ HMIDIIN winmidiinhandle;
MidiIn midictl;//used to convert the controllers to ZynAddSubFX controllers
void CALLBACK WinMidiInProc(HMIDIIN hMidiIn,UINT wMsg,DWORD dwInstance,
- DWORD dwParam1,DWORD dwParam2){
- int midicommand=MidiNull;
- if (wMsg==MIM_DATA){
- int cmd,par1,par2;
- cmd=dwParam1&0xff;
- if (cmd==0xfe) return;
- par1=(dwParam1>>8)&0xff;
- par2=dwParam1>>16;
- //printf("%x %x %x\n",cmd,par1,par2);fflush(stdout);
- int cmdchan=cmd&0x0f;
- int cmdtype=(cmd>>4)&0x0f;
+ DWORD dwParam1,DWORD dwParam2)
+{
+ int midicommand=MidiNull;
+ if (wMsg==MIM_DATA) {
+ int cmd,par1,par2;
+ cmd=dwParam1&0xff;
+ if (cmd==0xfe) return;
+ par1=(dwParam1>>8)&0xff;
+ par2=dwParam1>>16;
+ //printf("%x %x %x\n",cmd,par1,par2);fflush(stdout);
+ int cmdchan=cmd&0x0f;
+ int cmdtype=(cmd>>4)&0x0f;
- int tmp=0;
- pthread_mutex_lock(&winmaster->mutex);
- switch(cmdtype){
- case(0x8)://noteon
- winmaster->NoteOff(cmdchan,par1);
- break;
- case(0x9)://noteoff
- winmaster->NoteOn(cmdchan,par1,par2&0xff);
- break;
- case(0xb)://controller
- winmaster->SetController(cmdchan,midictl.getcontroller(par1),par2&0xff);
- break;
- case(0xe)://pitch wheel
- tmp=(par1+par2*(long int) 128)-8192;
- winmaster->SetController(cmdchan,C_pitchwheel,tmp);
- break;
- default:break;
- };
- pthread_mutex_unlock(&winmaster->mutex);
+ int tmp=0;
+ pthread_mutex_lock(&winmaster->mutex);
+ switch (cmdtype) {
+ case(0x8)://noteon
+ winmaster->NoteOff(cmdchan,par1);
+ break;
+ case(0x9)://noteoff
+ winmaster->NoteOn(cmdchan,par1,par2&0xff);
+ break;
+ case(0xb)://controller
+ winmaster->SetController(cmdchan,midictl.getcontroller(par1),par2&0xff);
+ break;
+ case(0xe)://pitch wheel
+ tmp=(par1+par2*(long int) 128)-8192;
+ winmaster->SetController(cmdchan,C_pitchwheel,tmp);
+ break;
+ default:
+ break;
+ };
+ pthread_mutex_unlock(&winmaster->mutex);
- };
+ };
};
-void InitWinMidi(Master *master_){
- winmaster=master_;
-
- long int result=midiInOpen(&winmidiinhandle,config.cfg.WindowsMidiInId,(DWORD)WinMidiInProc,0,CALLBACK_FUNCTION);
- result=midiInStart(winmidiinhandle);
+void InitWinMidi(Master *master_)
+{
+ winmaster=master_;
+
+ long int result=midiInOpen(&winmidiinhandle,config.cfg.WindowsMidiInId,(DWORD)WinMidiInProc,0,CALLBACK_FUNCTION);
+ result=midiInStart(winmidiinhandle);
};
-void StopWinMidi(){
- midiInStop(winmidiinhandle);
- midiInClose(winmidiinhandle);
+void StopWinMidi()
+{
+ midiInStop(winmidiinhandle);
+ midiInClose(winmidiinhandle);
};
diff --git a/src/Input/WINMidiIn.h b/src/Input/WINMidiIn.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
WINMidiIn.h - Midi input for Windows
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
diff --git a/src/Misc/Bank.C b/src/Misc/Bank.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
- Bank.h - Instrument Bank
+
+ Bank.h - Instrument Bank
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -39,37 +39,39 @@
//if this file exists into a directory, this make the directory to be considered as a bank, even if it not contains a instrument file
#define FORCE_BANK_DIR_FILE ".bankdir"
-Bank::Bank(){
+Bank::Bank()
+{
ZERO(defaultinsname,PART_MAX_NAME_LEN);
snprintf(defaultinsname,PART_MAX_NAME_LEN,"%s"," ");
-
- for (int i=0;i<BANK_SIZE;i++){
- ins[i].used=false;
- ins[i].filename=NULL;
- ins[i].info.PADsynth_used=false;
+
+ for (int i=0;i<BANK_SIZE;i++) {
+ ins[i].used=false;
+ ins[i].filename=NULL;
+ ins[i].info.PADsynth_used=false;
};
dirname=NULL;
clearbank();
- for (int i=0;i<MAX_NUM_BANKS;i++){
- banks[i].dir=NULL;
- banks[i].name=NULL;
+ for (int i=0;i<MAX_NUM_BANKS;i++) {
+ banks[i].dir=NULL;
+ banks[i].name=NULL;
};
-
+
bankfiletitle=dirname;
loadbank(config.cfg.currentBankDir);
};
-Bank::~Bank(){
- for (int i=0;i<MAX_NUM_BANKS;i++){
- if (banks[i].dir!=NULL) delete []banks[i].dir;
- if (banks[i].name!=NULL) delete []banks[i].name;
+Bank::~Bank()
+{
+ for (int i=0;i<MAX_NUM_BANKS;i++) {
+ if (banks[i].dir!=NULL) delete []banks[i].dir;
+ if (banks[i].name!=NULL) delete []banks[i].name;
};
clearbank();
@@ -78,7 +80,8 @@ Bank::~Bank(){
/*
* Get the name of an instrument from the bank
*/
-char *Bank::getname (unsigned int ninstrument){
+char *Bank::getname (unsigned int ninstrument)
+{
if (emptyslot(ninstrument)) return (defaultinsname);
return (ins[ninstrument].name);
};
@@ -86,7 +89,8 @@ char *Bank::getname (unsigned int ninstrument){
/*
* Get the numbered name of an instrument from the bank
*/
-char *Bank::getnamenumbered (unsigned int ninstrument){
+char *Bank::getnamenumbered (unsigned int ninstrument)
+{
if (emptyslot(ninstrument)) return (defaultinsname);
snprintf(tmpinsname[ninstrument],PART_MAX_NAME_LEN+15,"%d. %s",ninstrument+1,getname(ninstrument));
return(tmpinsname[ninstrument]);
@@ -95,28 +99,29 @@ char *Bank::getnamenumbered (unsigned int ninstrument){
/*
* Changes the name of an instrument (and the filename)
*/
-void Bank::setname(unsigned int ninstrument,const char *newname,int newslot){
+void Bank::setname(unsigned int ninstrument,const char *newname,int newslot)
+{
if (emptyslot(ninstrument)) return;
-
+
char newfilename[1000+1],tmpfilename[100+1];
-
+
ZERO(newfilename,1001);
ZERO(tmpfilename,101);
if (newslot>=0) snprintf(tmpfilename,100,"%4d-%s",newslot+1,newname);
- else snprintf(tmpfilename,100,"%4d-%s",ninstrument+1,newname);
-
+ else snprintf(tmpfilename,100,"%4d-%s",ninstrument+1,newname);
+
//add the zeroes at the start of filename
for (int i=0;i<4;i++) if (tmpfilename[i]==' ') tmpfilename[i]='0';
//make the filenames legal
for (int i=0;i<(int) strlen(tmpfilename);i++) {
- char c=tmpfilename[i];
- if ((c>='0')&&(c<='9')) continue;
- if ((c>='A')&&(c<='Z')) continue;
- if ((c>='a')&&(c<='z')) continue;
- if ((c=='-')||(c==' ')) continue;
-
- tmpfilename[i]='_';
+ char c=tmpfilename[i];
+ if ((c>='0')&&(c<='9')) continue;
+ if ((c>='A')&&(c<='Z')) continue;
+ if ((c>='a')&&(c<='z')) continue;
+ if ((c=='-')||(c==' ')) continue;
+
+ tmpfilename[i]='_';
};
snprintf(newfilename,1000,"%s/%s.xiz",dirname,tmpfilename);
@@ -128,39 +133,42 @@ void Bank::setname(unsigned int ninstrument,const char *newname,int newslot){
ins[ninstrument].filename=new char[strlen(newfilename)+5];
snprintf(ins[ninstrument].filename,strlen(newfilename)+1,"%s",newfilename);
snprintf(ins[ninstrument].name,PART_MAX_NAME_LEN,"%s",&tmpfilename[5]);
-
+
};
/*
* Check if there is no instrument on a slot from the bank
*/
-int Bank::emptyslot(unsigned int ninstrument){
+int Bank::emptyslot(unsigned int ninstrument)
+{
if (ninstrument>=BANK_SIZE) return (1);
if (ins[ninstrument].filename==NULL) return(1);
if (ins[ninstrument].used) return (0);
- else return(1);
+ else return(1);
};
/*
* Removes the instrument from the bank
*/
-void Bank::clearslot(unsigned int ninstrument){
+void Bank::clearslot(unsigned int ninstrument)
+{
if (emptyslot(ninstrument)) return;
-
+
// printf("remove %s \n",ins[ninstrument].filename);////////////////////////
-
-
+
+
remove(ins[ninstrument].filename);
deletefrombank(ninstrument);
};
/*
- * Save the instrument to a slot
+ * Save the instrument to a slot
*/
-void Bank::savetoslot(unsigned int ninstrument,Part *part){
+void Bank::savetoslot(unsigned int ninstrument,Part *part)
+{
clearslot(ninstrument);
-
+
const int maxfilename=200;
char tmpfilename[maxfilename+20];
ZERO(tmpfilename,maxfilename+20);
@@ -169,16 +177,16 @@ void Bank::savetoslot(unsigned int ninstrument,Part *part){
//add the zeroes at the start of filename
for (int i=0;i<4;i++) if (tmpfilename[i]==' ') tmpfilename[i]='0';
-
+
//make the filenames legal
for (int i=0;i<(int)strlen(tmpfilename);i++) {
- char c=tmpfilename[i];
- if ((c>='0')&&(c<='9')) continue;
- if ((c>='A')&&(c<='Z')) continue;
- if ((c>='a')&&(c<='z')) continue;
- if ((c=='-')||(c==' ')) continue;
-
- tmpfilename[i]='_';
+ char c=tmpfilename[i];
+ if ((c>='0')&&(c<='9')) continue;
+ if ((c>='A')&&(c<='Z')) continue;
+ if ((c>='a')&&(c<='z')) continue;
+ if ((c=='-')||(c==' ')) continue;
+
+ tmpfilename[i]='_';
};
strncat(tmpfilename,".xiz",maxfilename+10);
@@ -192,29 +200,31 @@ void Bank::savetoslot(unsigned int ninstrument,Part *part){
remove(filename);
part->saveXML(filename);
addtobank(ninstrument,tmpfilename,(char *) part->Pname);
-
+
delete[]filename;
};
/*
* Loads the instrument from the bank
*/
-void Bank::loadfromslot(unsigned int ninstrument,Part *part){
+void Bank::loadfromslot(unsigned int ninstrument,Part *part)
+{
if (emptyslot(ninstrument)) return;
-
+
part->defaultsinstrument();
// printf("load: %s\n",ins[ninstrument].filename);
-
+
part->loadXMLinstrument(ins[ninstrument].filename);
-
+
};
/*
* Makes current a bank directory
*/
-int Bank::loadbank(const char *bankdirname){
+int Bank::loadbank(const char *bankdirname)
+{
DIR *dir=opendir(bankdirname);
clearbank();
@@ -223,59 +233,59 @@ int Bank::loadbank(const char *bankdirname){
if (dirname!=NULL) delete[]dirname;
dirname=new char[strlen(bankdirname)+1];
snprintf(dirname,strlen(bankdirname)+1,"%s",bankdirname);
-
+
bankfiletitle=dirname;
- // printf("loadbank %s/\n",bankdirname);
+ // printf("loadbank %s/\n",bankdirname);
struct dirent *fn;
-
- while ((fn=readdir(dir))){
- const char *filename= fn->d_name;
-
- //sa verific daca e si extensia dorita
- if (strstr(filename,INSTRUMENT_EXTENSION)==NULL) continue;
-
- //verify if the name is like this NNNN-name (where N is a digit)
- int no=0;
- unsigned int startname=0;
-
- for (unsigned int i=0;i<4;i++) {
- if (strlen(filename)<=i) break;
-
- if ((filename[i]>='0')&&(filename[i]<='9')) {
- no=no*10+(filename[i]-'0');
- startname++;
- };
- };
-
-
- if ((startname+1)<strlen(filename)) startname++;//to take out the "-"
-
- char name[PART_MAX_NAME_LEN+1];
- ZERO(name,PART_MAX_NAME_LEN+1);
- snprintf(name,PART_MAX_NAME_LEN,"%s",filename);
-
- //remove the file extension
- for (int i=strlen(name)-1;i>=2;i--){
- if (name[i]=='.') {
- name[i]='\0';
- break;
- };
- };
-
- if (no!=0){//the instrument position in the bank is found
- addtobank(no-1,filename,&name[startname]);
- } else {
- addtobank(-1,filename,name);
- };
-
+
+ while ((fn=readdir(dir))) {
+ const char *filename= fn->d_name;
+
+ //sa verific daca e si extensia dorita
+ if (strstr(filename,INSTRUMENT_EXTENSION)==NULL) continue;
+
+ //verify if the name is like this NNNN-name (where N is a digit)
+ int no=0;
+ unsigned int startname=0;
+
+ for (unsigned int i=0;i<4;i++) {
+ if (strlen(filename)<=i) break;
+
+ if ((filename[i]>='0')&&(filename[i]<='9')) {
+ no=no*10+(filename[i]-'0');
+ startname++;
+ };
+ };
+
+
+ if ((startname+1)<strlen(filename)) startname++;//to take out the "-"
+
+ char name[PART_MAX_NAME_LEN+1];
+ ZERO(name,PART_MAX_NAME_LEN+1);
+ snprintf(name,PART_MAX_NAME_LEN,"%s",filename);
+
+ //remove the file extension
+ for (int i=strlen(name)-1;i>=2;i--) {
+ if (name[i]=='.') {
+ name[i]='\0';
+ break;
+ };
+ };
+
+ if (no!=0) {//the instrument position in the bank is found
+ addtobank(no-1,filename,&name[startname]);
+ } else {
+ addtobank(-1,filename,name);
+ };
+
};
-
-
+
+
closedir(dir);
if (dirname!=NULL) {
- sprintf(config.cfg.currentBankDir,"%s",dirname);
+ sprintf(config.cfg.currentBankDir,"%s",dirname);
};
return(0);
@@ -284,14 +294,15 @@ int Bank::loadbank(const char *bankdirname){
/*
* Makes a new bank, put it on a file and makes it current bank
*/
-int Bank::newbank(const char *newbankdirname){
+int Bank::newbank(const char *newbankdirname)
+{
int result;
char tmpfilename[MAX_STRING_SIZE];
char bankdir[MAX_STRING_SIZE];
snprintf(bankdir,MAX_STRING_SIZE,"%s",config.cfg.bankRootDirList[0]);
- if (((bankdir[strlen(bankdir)-1])!='/')&&((bankdir[strlen(bankdir)-1])!='\\')){
- strncat(bankdir,"/",MAX_STRING_SIZE);
+ if (((bankdir[strlen(bankdir)-1])!='/')&&((bankdir[strlen(bankdir)-1])!='\\')) {
+ strncat(bankdir,"/",MAX_STRING_SIZE);
};
strncat(bankdir,newbankdirname,MAX_STRING_SIZE);
#ifdef OS_WINDOWS
@@ -305,57 +316,62 @@ int Bank::newbank(const char *newbankdirname){
// printf("%s\n",tmpfilename);
FILE *tmpfile=fopen(tmpfilename,"w+");
fclose(tmpfile);
-
+
return(loadbank(bankdir));
};
/*
* Check if the bank is locked (i.e. the file opened was readonly)
*/
-int Bank::locked(){
+int Bank::locked()
+{
return(dirname==NULL);
};
/*
* Swaps a slot with another
*/
-void Bank::swapslot(unsigned int n1, unsigned int n2){
+void Bank::swapslot(unsigned int n1, unsigned int n2)
+{
if ((n1==n2)||(locked())) return;
if (emptyslot(n1)&&(emptyslot(n2))) return;
- if (emptyslot(n1)){//change n1 to n2 in order to make
- int tmp=n2;n2=n1;n1=tmp;
+ if (emptyslot(n1)) {//change n1 to n2 in order to make
+ int tmp=n2;
+ n2=n1;
+ n1=tmp;
};
-
- if (emptyslot(n2)){//this is just a movement from slot1 to slot2
- setname(n1,getname(n1),n2);
- ins[n2]=ins[n1];
- ins[n1].used=false;
- ins[n1].name[0]='\0';
- ins[n1].filename=NULL;
- ins[n1].info.PADsynth_used=0;
+
+ if (emptyslot(n2)) {//this is just a movement from slot1 to slot2
+ setname(n1,getname(n1),n2);
+ ins[n2]=ins[n1];
+ ins[n1].used=false;
+ ins[n1].name[0]='\0';
+ ins[n1].filename=NULL;
+ ins[n1].info.PADsynth_used=0;
} else {//if both slots are used
- if (strcmp(ins[n1].name,ins[n2].name)==0){//change the name of the second instrument if the name are equal
- strncat(ins[n2].name,"2",PART_MAX_NAME_LEN);
- };
- setname(n1,getname(n1),n2);
- setname(n2,getname(n2),n1);
- ins_t tmp;
- tmp.used=true;
- strcpy(tmp.name,ins[n2].name);
- char *tmpfilename=ins[n2].filename;
- bool padsynth_used=ins[n2].info.PADsynth_used;
-
- ins[n2]=ins[n1];
- strcpy(ins[n1].name,tmp.name);
- ins[n1].filename=tmpfilename;
- ins[n1].info.PADsynth_used=padsynth_used;
+ if (strcmp(ins[n1].name,ins[n2].name)==0) {//change the name of the second instrument if the name are equal
+ strncat(ins[n2].name,"2",PART_MAX_NAME_LEN);
+ };
+ setname(n1,getname(n1),n2);
+ setname(n2,getname(n2),n1);
+ ins_t tmp;
+ tmp.used=true;
+ strcpy(tmp.name,ins[n2].name);
+ char *tmpfilename=ins[n2].filename;
+ bool padsynth_used=ins[n2].info.PADsynth_used;
+
+ ins[n2]=ins[n1];
+ strcpy(ins[n1].name,tmp.name);
+ ins[n1].filename=tmpfilename;
+ ins[n1].info.PADsynth_used=padsynth_used;
};
-
+
};
//a helper function that compares 2 banks[] arrays
-int Bank_compar(const void *a,const void *b){
+int Bank_compar(const void *a,const void *b)
+{
struct Bank::bankstruct *bank1= (Bank::bankstruct *)a;
struct Bank::bankstruct *bank2= (Bank::bankstruct *)b;
if (((bank1->name)==NULL)||((bank2->name)==NULL)) return(0);
@@ -369,156 +385,160 @@ int Bank_compar(const void *a,const void *b){
* Re-scan for directories containing instrument banks
*/
-void Bank::rescanforbanks(){
- for (int i=0;i<MAX_NUM_BANKS;i++){
- if (banks[i].dir!=NULL) delete []banks[i].dir;
- if (banks[i].name!=NULL) delete []banks[i].name;
- banks[i].dir=NULL;
- banks[i].name=NULL;
+void Bank::rescanforbanks()
+{
+ for (int i=0;i<MAX_NUM_BANKS;i++) {
+ if (banks[i].dir!=NULL) delete []banks[i].dir;
+ if (banks[i].name!=NULL) delete []banks[i].name;
+ banks[i].dir=NULL;
+ banks[i].name=NULL;
};
for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) if (config.cfg.bankRootDirList[i]!=NULL) scanrootdir(config.cfg.bankRootDirList[i]);
//sort the banks
- for (int j=0;j<MAX_NUM_BANKS-1;j++){
- for (int i=j+1;i<MAX_NUM_BANKS;i++){
- if (Bank_compar(&banks[i],&banks[j])) {
- char *tmpname=banks[i].name;
- char *tmpdir=banks[i].dir;
-
- banks[i].name=banks[j].name;
- banks[i].dir=banks[j].dir;
-
- banks[j].name=tmpname;
- banks[j].dir=tmpdir;
-
- };
- };
+ for (int j=0;j<MAX_NUM_BANKS-1;j++) {
+ for (int i=j+1;i<MAX_NUM_BANKS;i++) {
+ if (Bank_compar(&banks[i],&banks[j])) {
+ char *tmpname=banks[i].name;
+ char *tmpdir=banks[i].dir;
+
+ banks[i].name=banks[j].name;
+ banks[i].dir=banks[j].dir;
+
+ banks[j].name=tmpname;
+ banks[j].dir=tmpdir;
+
+ };
+ };
};
-
+
//remove duplicate bank names
int dupl=0;
- for (int j=0;j<MAX_NUM_BANKS-1;j++){
- for (int i=j+1;i<MAX_NUM_BANKS;i++){
- if ((banks[i].name==NULL)||(banks[j].name==NULL)) continue;
- if (strcmp(banks[i].name,banks[j].name)==0) {//add a [1] to the first bankname and [n] to others
- char *tmpname=banks[i].name;
- banks[i].name=new char[strlen(tmpname)+100];
- sprintf(banks[i].name,"%s[%d]",tmpname,dupl+2);
- delete[]tmpname;
-
- if (dupl==0){
- char *tmpname=banks[j].name;
- banks[j].name=new char[strlen(tmpname)+100];
- sprintf(banks[j].name,"%s[1]",tmpname);
- delete[]tmpname;
- };
-
- dupl++;
- } else dupl=0;
- };
+ for (int j=0;j<MAX_NUM_BANKS-1;j++) {
+ for (int i=j+1;i<MAX_NUM_BANKS;i++) {
+ if ((banks[i].name==NULL)||(banks[j].name==NULL)) continue;
+ if (strcmp(banks[i].name,banks[j].name)==0) {//add a [1] to the first bankname and [n] to others
+ char *tmpname=banks[i].name;
+ banks[i].name=new char[strlen(tmpname)+100];
+ sprintf(banks[i].name,"%s[%d]",tmpname,dupl+2);
+ delete[]tmpname;
+
+ if (dupl==0) {
+ char *tmpname=banks[j].name;
+ banks[j].name=new char[strlen(tmpname)+100];
+ sprintf(banks[j].name,"%s[1]",tmpname);
+ delete[]tmpname;
+ };
+
+ dupl++;
+ } else dupl=0;
+ };
};
-
-};
+
+};
// private stuff
-void Bank::scanrootdir(char *rootdir){
+void Bank::scanrootdir(char *rootdir)
+{
// printf("Scanning root dir:%s\n",rootdir);
DIR *dir=opendir(rootdir);
if (dir==NULL) return;
const int maxdirsize=1000;
struct {
- char dir[maxdirsize];
- char name[maxdirsize];
+ char dir[maxdirsize];
+ char name[maxdirsize];
}bank;
const char *separator="/";
if (strlen(rootdir)) {
- char tmp=rootdir[strlen(rootdir)-1];
- if ((tmp=='/') || (tmp=='\\')) separator="";
+ char tmp=rootdir[strlen(rootdir)-1];
+ if ((tmp=='/') || (tmp=='\\')) separator="";
};
-
+
struct dirent *fn;
- while ((fn=readdir(dir))){
- const char *dirname=fn->d_name;
- if (dirname[0]=='.') continue;
-
- snprintf(bank.dir,maxdirsize,"%s%s%s/",rootdir,separator,dirname);
- snprintf(bank.name,maxdirsize,"%s",dirname);
- //find out if the directory contains at least 1 instrument
- bool isbank=false;
-
- DIR *d=opendir(bank.dir);
- if (d==NULL) continue;
-
- struct dirent *fname;
-
- while((fname=readdir(d))){
- if ((strstr(fname->d_name,INSTRUMENT_EXTENSION)!=NULL)||
- (strstr(fname->d_name,FORCE_BANK_DIR_FILE)!=NULL)){
- isbank=true;
- break;//aici as putea pune in loc de break un update la un counter care imi arata nr. de instrumente din bank
- };
- };
-
- closedir(d);
-
- if (isbank) {
- int pos=-1;
- for (int i=1;i<MAX_NUM_BANKS;i++){ //banks[0] e liber intotdeauna
- if (banks[i].name==NULL) {
- pos=i;
- break;
- };
- };
-
- if (pos>=0) {
- banks[pos].name=new char[maxdirsize];
- banks[pos].dir=new char[maxdirsize];
- snprintf(banks[pos].name,maxdirsize,"%s",bank.name);
- snprintf(banks[pos].dir,maxdirsize,"%s",bank.dir);
- };
-
- };
-
+ while ((fn=readdir(dir))) {
+ const char *dirname=fn->d_name;
+ if (dirname[0]=='.') continue;
+
+ snprintf(bank.dir,maxdirsize,"%s%s%s/",rootdir,separator,dirname);
+ snprintf(bank.name,maxdirsize,"%s",dirname);
+ //find out if the directory contains at least 1 instrument
+ bool isbank=false;
+
+ DIR *d=opendir(bank.dir);
+ if (d==NULL) continue;
+
+ struct dirent *fname;
+
+ while ((fname=readdir(d))) {
+ if ((strstr(fname->d_name,INSTRUMENT_EXTENSION)!=NULL)||
+ (strstr(fname->d_name,FORCE_BANK_DIR_FILE)!=NULL)) {
+ isbank=true;
+ break;//aici as putea pune in loc de break un update la un counter care imi arata nr. de instrumente din bank
+ };
+ };
+
+ closedir(d);
+
+ if (isbank) {
+ int pos=-1;
+ for (int i=1;i<MAX_NUM_BANKS;i++) { //banks[0] e liber intotdeauna
+ if (banks[i].name==NULL) {
+ pos=i;
+ break;
+ };
+ };
+
+ if (pos>=0) {
+ banks[pos].name=new char[maxdirsize];
+ banks[pos].dir=new char[maxdirsize];
+ snprintf(banks[pos].name,maxdirsize,"%s",bank.name);
+ snprintf(banks[pos].dir,maxdirsize,"%s",bank.dir);
+ };
+
+ };
+
};
closedir(dir);
-
+
};
-void Bank::clearbank(){
+void Bank::clearbank()
+{
for (int i=0;i<BANK_SIZE;i++) deletefrombank(i);
if (dirname!=NULL) delete[]dirname;
bankfiletitle=NULL;
dirname=NULL;
};
-int Bank::addtobank(int pos, const char *filename, const char* name){
- if ((pos>=0)&&(pos<BANK_SIZE)){
- if (ins[pos].used) pos=-1;//force it to find a new free position
+int Bank::addtobank(int pos, const char *filename, const char* name)
+{
+ if ((pos>=0)&&(pos<BANK_SIZE)) {
+ if (ins[pos].used) pos=-1;//force it to find a new free position
} else if (pos>=BANK_SIZE) pos=-1;
-
-
+
+
if (pos<0) {//find a free position
- for (int i=BANK_SIZE-1;i>=0;i--)
- if (!ins[i].used) {
- pos=i;
- break;
- };
-
+ 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
- // printf("%s %d\n",filename,pos);
+ // printf("%s %d\n",filename,pos);
deletefrombank(pos);
-
+
ins[pos].used=true;
snprintf(ins[pos].name,PART_MAX_NAME_LEN,"%s",name);
@@ -530,33 +550,35 @@ int Bank::addtobank(int pos, const char *filename, const char* name){
snprintf(ins[pos].filename,len+1,"%s/%s",dirname,filename);
//see if PADsynth is used
- if (config.cfg.CheckPADsynth){
- XMLwrapper *xml=new XMLwrapper();
- xml->checkfileinformation(ins[pos].filename);
-
- ins[pos].info.PADsynth_used=xml->information.PADsynth_used;
- delete xml;
+ if (config.cfg.CheckPADsynth) {
+ XMLwrapper *xml=new XMLwrapper();
+ xml->checkfileinformation(ins[pos].filename);
+
+ ins[pos].info.PADsynth_used=xml->information.PADsynth_used;
+ delete xml;
} else ins[pos].info.PADsynth_used=false;
-
+
return(0);
};
-bool Bank::isPADsynth_used(unsigned int ninstrument){
+bool Bank::isPADsynth_used(unsigned int ninstrument)
+{
if (config.cfg.CheckPADsynth==0) return(0);
- else return(ins[ninstrument].info.PADsynth_used);
+ else return(ins[ninstrument].info.PADsynth_used);
};
-void Bank::deletefrombank(int pos){
+void Bank::deletefrombank(int pos)
+{
if ((pos<0)||(pos>=BANK_SIZE)) return;
ins[pos].used=false;
ZERO(ins[pos].name,PART_MAX_NAME_LEN+1);
if (ins[pos].filename!=NULL) {
- delete []ins[pos].filename;
- ins[pos].filename=NULL;
+ delete []ins[pos].filename;
+ ins[pos].filename=NULL;
};
-
+
ZERO(tmpinsname[pos],PART_MAX_NAME_LEN+20);
-
+
};
diff --git a/src/Misc/Bank.h b/src/Misc/Bank.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Bank.C - Instrument Bank
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -36,71 +36,72 @@
/**The instrument Bank
* \todo add in strings to replace char* */
-class Bank{
- public:
- /**Constructor*/
- Bank();
- ~Bank();
- char *getname(unsigned int ninstrument);
- char *getnamenumbered(unsigned int ninstrument);
- void setname(unsigned int ninstrument,const char *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 0 if the slot is not empty or 1 if the slot is empty
- * \todo start using bool before facepalm*/
- int emptyslot(unsigned int ninstrument);
-
- /**Empties out the selected slot*/
- void clearslot(unsigned int ninstrument);
- /**Saves the given Part to slot*/
- void savetoslot(unsigned int ninstrument,Part *part);
- /**Loads the given slot into a Part*/
- void loadfromslot(unsigned int ninstrument,Part *part);
-
- /**Swaps Slots*/
- void swapslot(unsigned int n1,unsigned int n2);
-
- int loadbank(const char *bankdirname);
- int newbank(const char *newbankdirname);
-
- char *bankfiletitle; //this is shown on the UI of the bank (the title of the window)
- int locked();
-
- void rescanforbanks();
-
- struct bankstruct{
- char *dir;
- char *name;
- };
-
- bankstruct banks[MAX_NUM_BANKS];
-
- private:
-
- //it adds a filename to the bank
- //if pos is -1 it try to find a position
- //returns -1 if the bank is full, or 0 if the instrument was added
- int addtobank(int pos,const char* filename,const char* name);
-
- void deletefrombank(int pos);
-
- void clearbank();
-
- char defaultinsname[PART_MAX_NAME_LEN];
- char tmpinsname[BANK_SIZE][PART_MAX_NAME_LEN+20];//this keeps the numbered names
-
- struct ins_t{
- bool used;
- char name[PART_MAX_NAME_LEN+1];
- char *filename;
- struct{
- bool PADsynth_used;
- } info;
- }ins[BANK_SIZE];
-
- char *dirname;
-
- void scanrootdir(char *rootdir);//scans a root dir for banks
+class Bank
+{
+public:
+ /**Constructor*/
+ Bank();
+ ~Bank();
+ char *getname(unsigned int ninstrument);
+ char *getnamenumbered(unsigned int ninstrument);
+ void setname(unsigned int ninstrument,const char *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 0 if the slot is not empty or 1 if the slot is empty
+ * \todo start using bool before facepalm*/
+ int emptyslot(unsigned int ninstrument);
+
+ /**Empties out the selected slot*/
+ void clearslot(unsigned int ninstrument);
+ /**Saves the given Part to slot*/
+ void savetoslot(unsigned int ninstrument,Part *part);
+ /**Loads the given slot into a Part*/
+ void loadfromslot(unsigned int ninstrument,Part *part);
+
+ /**Swaps Slots*/
+ void swapslot(unsigned int n1,unsigned int n2);
+
+ int loadbank(const char *bankdirname);
+ int newbank(const char *newbankdirname);
+
+ char *bankfiletitle; //this is shown on the UI of the bank (the title of the window)
+ int locked();
+
+ void rescanforbanks();
+
+ struct bankstruct {
+ char *dir;
+ char *name;
+ };
+
+ bankstruct banks[MAX_NUM_BANKS];
+
+private:
+
+ //it adds a filename to the bank
+ //if pos is -1 it try to find a position
+ //returns -1 if the bank is full, or 0 if the instrument was added
+ int addtobank(int pos,const char* filename,const char* name);
+
+ void deletefrombank(int pos);
+
+ void clearbank();
+
+ char defaultinsname[PART_MAX_NAME_LEN];
+ char tmpinsname[BANK_SIZE][PART_MAX_NAME_LEN+20];//this keeps the numbered names
+
+ struct ins_t {
+ bool used;
+ char name[PART_MAX_NAME_LEN+1];
+ char *filename;
+ struct {
+ bool PADsynth_used;
+ } info;
+ }ins[BANK_SIZE];
+
+ char *dirname;
+
+ void scanrootdir(char *rootdir);//scans a root dir for banks
};
#endif
diff --git a/src/Misc/Config.C b/src/Misc/Config.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Config.C - Configuration file functions
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -32,16 +32,18 @@
#include "Config.h"
#include "XMLwrapper.h"
-Config::Config(){
+Config::Config()
+{
};
-void Config::init(){
+void Config::init()
+{
maxstringsize=MAX_STRING_SIZE;//for ui
//defaults
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];
@@ -52,7 +54,7 @@ void Config::init(){
cfg.WindowsWaveOutId=0;
cfg.WindowsMidiInId=0;
-
+
cfg.BankUIAutoClose=0;
cfg.DumpNotesToFile=0;
cfg.DumpAppend=1;
@@ -64,7 +66,8 @@ void Config::init(){
cfg.UserInterfaceMode=0;
cfg.VirKeybLayout=1;
- winwavemax=1;winmidimax=1;
+ winwavemax=1;
+ winmidimax=1;
//try to find out how many input midi devices are there
#ifdef WINMIDIIN
winmidimax=midiInGetNumDevs();
@@ -72,107 +75,108 @@ void Config::init(){
#endif
winmididevices=new winmidionedevice[winmidimax];
for (int i=0;i<winmidimax;i++) {
- winmididevices[i].name=new char[MAX_STRING_SIZE];
- for (int j=0;j<MAX_STRING_SIZE;j++) winmididevices[i].name[j]='\0';
+ winmididevices[i].name=new char[MAX_STRING_SIZE];
+ for (int j=0;j<MAX_STRING_SIZE;j++) winmididevices[i].name[j]='\0';
};
//get the midi input devices name
#ifdef WINMIDIIN
MIDIINCAPS midiincaps;
- for (int i=0;i<winmidimax;i++){
- if (! midiInGetDevCaps(i,&midiincaps,sizeof(MIDIINCAPS)))
- snprintf(winmididevices[i].name,MAX_STRING_SIZE,"%s",midiincaps.szPname);
+ for (int i=0;i<winmidimax;i++) {
+ if (! midiInGetDevCaps(i,&midiincaps,sizeof(MIDIINCAPS)))
+ snprintf(winmididevices[i].name,MAX_STRING_SIZE,"%s",midiincaps.szPname);
};
#endif
for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) cfg.bankRootDirList[i]=NULL;
cfg.currentBankDir=new char[MAX_STRING_SIZE];
sprintf(cfg.currentBankDir,"./testbnk");
-
+
for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) cfg.presetsDirList[i]=NULL;
-
+
char filename[MAX_STRING_SIZE];
getConfigFileName(filename,MAX_STRING_SIZE);
readConfig(filename);
- if (cfg.bankRootDirList[0]==NULL){
+ if (cfg.bankRootDirList[0]==NULL) {
#if defined(OS_LINUX)
- //banks
+ //banks
cfg.bankRootDirList[0]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[0],"~/banks");
+ sprintf(cfg.bankRootDirList[0],"~/banks");
- cfg.bankRootDirList[1]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[1],"./");
+ cfg.bankRootDirList[1]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.bankRootDirList[1],"./");
cfg.bankRootDirList[2]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[2],"/usr/share/zynaddsubfx/banks");
+ sprintf(cfg.bankRootDirList[2],"/usr/share/zynaddsubfx/banks");
+
+ cfg.bankRootDirList[3]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.bankRootDirList[3],"/usr/local/share/zynaddsubfx/banks");
- cfg.bankRootDirList[3]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[3],"/usr/local/share/zynaddsubfx/banks");
+ cfg.bankRootDirList[4]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.bankRootDirList[4],"../banks");
- cfg.bankRootDirList[4]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[4],"../banks");
+ cfg.bankRootDirList[5]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.bankRootDirList[5],"banks");
- cfg.bankRootDirList[5]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[5],"banks");
-
#else
- //banks
- cfg.bankRootDirList[0]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[0],"./");
+ //banks
+ cfg.bankRootDirList[0]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.bankRootDirList[0],"./");
#ifdef VSTAUDIOOUT
- cfg.bankRootDirList[1]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[1],"c:/Program Files/ZynAddSubFX/banks");
+ cfg.bankRootDirList[1]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.bankRootDirList[1],"c:/Program Files/ZynAddSubFX/banks");
#else
- cfg.bankRootDirList[1]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[1],"../banks");
+ cfg.bankRootDirList[1]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.bankRootDirList[1],"../banks");
#endif
- cfg.bankRootDirList[2]=new char[MAX_STRING_SIZE];
- sprintf(cfg.bankRootDirList[2],"banks");
-
+ cfg.bankRootDirList[2]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.bankRootDirList[2],"banks");
+
#endif
};
- if (cfg.presetsDirList[0]==NULL){
+ if (cfg.presetsDirList[0]==NULL) {
#if defined(OS_LINUX)
- //presets
- cfg.presetsDirList[0]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[0],"./");
+ //presets
+ cfg.presetsDirList[0]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.presetsDirList[0],"./");
+
+ cfg.presetsDirList[1]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.presetsDirList[1],"../presets");
- cfg.presetsDirList[1]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[1],"../presets");
+ cfg.presetsDirList[2]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.presetsDirList[2],"presets");
- cfg.presetsDirList[2]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[2],"presets");
-
cfg.presetsDirList[3]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[3],"/usr/share/zynaddsubfx/presets");
+ sprintf(cfg.presetsDirList[3],"/usr/share/zynaddsubfx/presets");
+
+ cfg.presetsDirList[4]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.presetsDirList[4],"/usr/local/share/zynaddsubfx/presets");
- cfg.presetsDirList[4]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[4],"/usr/local/share/zynaddsubfx/presets");
-
#else
- //presets
- cfg.presetsDirList[0]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[0],"./");
+ //presets
+ cfg.presetsDirList[0]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.presetsDirList[0],"./");
#ifdef VSTAUDIOOUT
- cfg.presetsDirList[1]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[1],"c:/Program Files/ZynAddSubFX/presets");
+ cfg.presetsDirList[1]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.presetsDirList[1],"c:/Program Files/ZynAddSubFX/presets");
#else
- cfg.presetsDirList[1]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[1],"../presets");
+ cfg.presetsDirList[1]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.presetsDirList[1],"../presets");
#endif
- cfg.presetsDirList[2]=new char[MAX_STRING_SIZE];
- sprintf(cfg.presetsDirList[2],"presets");
+ cfg.presetsDirList[2]=new char[MAX_STRING_SIZE];
+ sprintf(cfg.presetsDirList[2],"presets");
#endif
};
};
-Config::~Config(){
+Config::~Config()
+{
delete [] cfg.LinuxOSSWaveOutDev;
delete [] cfg.LinuxOSSSeqInDev;
@@ -183,78 +187,82 @@ Config::~Config(){
};
-void Config::save(){
+void Config::save()
+{
char filename[MAX_STRING_SIZE];
getConfigFileName(filename,MAX_STRING_SIZE);
saveConfig(filename);
};
-void Config::clearbankrootdirlist(){
+void Config::clearbankrootdirlist()
+{
for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) {
- if (cfg.bankRootDirList[i]==NULL) delete(cfg.bankRootDirList[i]);
- cfg.bankRootDirList[i]=NULL;
+ if (cfg.bankRootDirList[i]==NULL) delete(cfg.bankRootDirList[i]);
+ cfg.bankRootDirList[i]=NULL;
};
};
-void Config::clearpresetsdirlist(){
+void Config::clearpresetsdirlist()
+{
for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) {
- if (cfg.presetsDirList[i]==NULL) delete(cfg.presetsDirList[i]);
- cfg.presetsDirList[i]=NULL;
+ if (cfg.presetsDirList[i]==NULL) delete(cfg.presetsDirList[i]);
+ cfg.presetsDirList[i]=NULL;
};
};
-void Config::readConfig(const char *filename){
+void Config::readConfig(const char *filename)
+{
XMLwrapper *xmlcfg=new XMLwrapper();
if (xmlcfg->loadXMLfile(filename)<0) return;
- if (xmlcfg->enterbranch("CONFIGURATION")){
- cfg.SampleRate=xmlcfg->getpar("sample_rate",cfg.SampleRate,4000,1024000);
- cfg.SoundBufferSize=xmlcfg->getpar("sound_buffer_size",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.BankUIAutoClose=xmlcfg->getpar("bank_window_auto_close",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);
- xmlcfg->getparstr("dump_file",cfg.DumpFile,MAX_STRING_SIZE);
-
- cfg.GzipCompression=xmlcfg->getpar("gzip_compression",cfg.GzipCompression,0,9);
-
- xmlcfg->getparstr("bank_current",cfg.currentBankDir,MAX_STRING_SIZE);
- cfg.Interpolation=xmlcfg->getpar("interpolation",cfg.Interpolation,0,1);
-
- cfg.CheckPADsynth=xmlcfg->getpar("check_pad_synth",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);
-
- //get bankroot dirs
- for (int i=0;i<MAX_BANK_ROOT_DIRS;i++){
- if (xmlcfg->enterbranch("BANKROOT",i)){
- cfg.bankRootDirList[i]=new char[MAX_STRING_SIZE];
- xmlcfg->getparstr("bank_root",cfg.bankRootDirList[i],MAX_STRING_SIZE);
- xmlcfg->exitbranch();
- };
- };
-
- //get preset root dirs
- for (int i=0;i<MAX_BANK_ROOT_DIRS;i++){
- if (xmlcfg->enterbranch("PRESETSROOT",i)){
- cfg.presetsDirList[i]=new char[MAX_STRING_SIZE];
- xmlcfg->getparstr("presets_root",cfg.presetsDirList[i],MAX_STRING_SIZE);
- xmlcfg->exitbranch();
- };
- };
-
- //linux stuff
- xmlcfg->getparstr("linux_oss_wave_out_dev",cfg.LinuxOSSWaveOutDev,MAX_STRING_SIZE);
- xmlcfg->getparstr("linux_oss_seq_in_dev",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);
-
- xmlcfg->exitbranch();
+ if (xmlcfg->enterbranch("CONFIGURATION")) {
+ cfg.SampleRate=xmlcfg->getpar("sample_rate",cfg.SampleRate,4000,1024000);
+ cfg.SoundBufferSize=xmlcfg->getpar("sound_buffer_size",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.BankUIAutoClose=xmlcfg->getpar("bank_window_auto_close",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);
+ xmlcfg->getparstr("dump_file",cfg.DumpFile,MAX_STRING_SIZE);
+
+ cfg.GzipCompression=xmlcfg->getpar("gzip_compression",cfg.GzipCompression,0,9);
+
+ xmlcfg->getparstr("bank_current",cfg.currentBankDir,MAX_STRING_SIZE);
+ cfg.Interpolation=xmlcfg->getpar("interpolation",cfg.Interpolation,0,1);
+
+ cfg.CheckPADsynth=xmlcfg->getpar("check_pad_synth",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);
+
+ //get bankroot dirs
+ for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) {
+ if (xmlcfg->enterbranch("BANKROOT",i)) {
+ cfg.bankRootDirList[i]=new char[MAX_STRING_SIZE];
+ xmlcfg->getparstr("bank_root",cfg.bankRootDirList[i],MAX_STRING_SIZE);
+ xmlcfg->exitbranch();
+ };
+ };
+
+ //get preset root dirs
+ for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) {
+ if (xmlcfg->enterbranch("PRESETSROOT",i)) {
+ cfg.presetsDirList[i]=new char[MAX_STRING_SIZE];
+ xmlcfg->getparstr("presets_root",cfg.presetsDirList[i],MAX_STRING_SIZE);
+ xmlcfg->exitbranch();
+ };
+ };
+
+ //linux stuff
+ xmlcfg->getparstr("linux_oss_wave_out_dev",cfg.LinuxOSSWaveOutDev,MAX_STRING_SIZE);
+ xmlcfg->getparstr("linux_oss_seq_in_dev",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);
+
+ xmlcfg->exitbranch();
};
delete(xmlcfg);
@@ -262,64 +270,66 @@ void Config::readConfig(const char *filename){
};
-void Config::saveConfig(const char *filename){
+void Config::saveConfig(const char *filename)
+{
XMLwrapper *xmlcfg=new XMLwrapper();
-
- xmlcfg->beginbranch("CONFIGURATION");
-
- xmlcfg->addpar("sample_rate",cfg.SampleRate);
- xmlcfg->addpar("sound_buffer_size",cfg.SoundBufferSize);
- xmlcfg->addpar("oscil_size",cfg.OscilSize);
- xmlcfg->addpar("swap_stereo",cfg.SwapStereo);
- xmlcfg->addpar("bank_window_auto_close",cfg.BankUIAutoClose);
- xmlcfg->addpar("dump_notes_to_file",cfg.DumpNotesToFile);
- xmlcfg->addpar("dump_append",cfg.DumpAppend);
- xmlcfg->addparstr("dump_file",cfg.DumpFile);
+ xmlcfg->beginbranch("CONFIGURATION");
+
+ xmlcfg->addpar("sample_rate",cfg.SampleRate);
+ xmlcfg->addpar("sound_buffer_size",cfg.SoundBufferSize);
+ xmlcfg->addpar("oscil_size",cfg.OscilSize);
+ xmlcfg->addpar("swap_stereo",cfg.SwapStereo);
+ xmlcfg->addpar("bank_window_auto_close",cfg.BankUIAutoClose);
- xmlcfg->addpar("gzip_compression",cfg.GzipCompression);
+ xmlcfg->addpar("dump_notes_to_file",cfg.DumpNotesToFile);
+ xmlcfg->addpar("dump_append",cfg.DumpAppend);
+ xmlcfg->addparstr("dump_file",cfg.DumpFile);
- xmlcfg->addpar("check_pad_synth",cfg.CheckPADsynth);
+ xmlcfg->addpar("gzip_compression",cfg.GzipCompression);
- xmlcfg->addparstr("bank_current",cfg.currentBankDir);
+ xmlcfg->addpar("check_pad_synth",cfg.CheckPADsynth);
- xmlcfg->addpar("user_interface_mode",cfg.UserInterfaceMode);
- xmlcfg->addpar("virtual_keyboard_layout",cfg.VirKeybLayout);
+ xmlcfg->addparstr("bank_current",cfg.currentBankDir);
+ xmlcfg->addpar("user_interface_mode",cfg.UserInterfaceMode);
+ xmlcfg->addpar("virtual_keyboard_layout",cfg.VirKeybLayout);
- for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) if (cfg.bankRootDirList[i]!=NULL) {
- xmlcfg->beginbranch("BANKROOT",i);
- xmlcfg->addparstr("bank_root",cfg.bankRootDirList[i]);
- xmlcfg->endbranch();
- };
- for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) if (cfg.presetsDirList[i]!=NULL) {
- xmlcfg->beginbranch("PRESETSROOT",i);
- xmlcfg->addparstr("presets_root",cfg.presetsDirList[i]);
- xmlcfg->endbranch();
- };
+ for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) if (cfg.bankRootDirList[i]!=NULL) {
+ xmlcfg->beginbranch("BANKROOT",i);
+ xmlcfg->addparstr("bank_root",cfg.bankRootDirList[i]);
+ xmlcfg->endbranch();
+ };
- xmlcfg->addpar("interpolation",cfg.Interpolation);
+ for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) if (cfg.presetsDirList[i]!=NULL) {
+ xmlcfg->beginbranch("PRESETSROOT",i);
+ xmlcfg->addparstr("presets_root",cfg.presetsDirList[i]);
+ xmlcfg->endbranch();
+ };
- //linux stuff
- xmlcfg->addparstr("linux_oss_wave_out_dev",cfg.LinuxOSSWaveOutDev);
- xmlcfg->addparstr("linux_oss_seq_in_dev",cfg.LinuxOSSSeqInDev);
-
- //windows stuff
- xmlcfg->addpar("windows_wave_out_id",cfg.WindowsWaveOutId);
- xmlcfg->addpar("windows_midi_in_id",cfg.WindowsMidiInId);
+ xmlcfg->addpar("interpolation",cfg.Interpolation);
+
+ //linux stuff
+ xmlcfg->addparstr("linux_oss_wave_out_dev",cfg.LinuxOSSWaveOutDev);
+ xmlcfg->addparstr("linux_oss_seq_in_dev",cfg.LinuxOSSSeqInDev);
+
+ //windows stuff
+ xmlcfg->addpar("windows_wave_out_id",cfg.WindowsWaveOutId);
+ xmlcfg->addpar("windows_midi_in_id",cfg.WindowsMidiInId);
xmlcfg->endbranch();
-
+
int tmp=cfg.GzipCompression;
cfg.GzipCompression=0;
xmlcfg->saveXMLfile(filename);
cfg.GzipCompression=tmp;
-
+
delete(xmlcfg);
};
-void Config::getConfigFileName(char *name, int namesize){
+void Config::getConfigFileName(char *name, int namesize)
+{
name[0]=0;
#ifdef OS_LINUX
snprintf(name,namesize,"%s%s",getenv("HOME"),"/.zynaddsubfxXML.cfg");
diff --git a/src/Misc/Config.h b/src/Misc/Config.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Config.h - Configuration file functions
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,44 +27,45 @@
#define MAX_BANK_ROOT_DIRS 100
/**Configuration file functions*/
-class Config{
- public:
- /** Constructor*/
- Config();
- /** Destructor*/
- ~Config();
- struct {
- char *LinuxOSSWaveOutDev,*LinuxOSSSeqInDev;
- int SampleRate,SoundBufferSize,OscilSize,SwapStereo;
- int WindowsWaveOutId,WindowsMidiInId;
- int BankUIAutoClose;
- int DumpNotesToFile,DumpAppend;
- int GzipCompression;
- int Interpolation;
- char *DumpFile;
- char *bankRootDirList[MAX_BANK_ROOT_DIRS],*currentBankDir;
- char *presetsDirList[MAX_BANK_ROOT_DIRS];
- int CheckPADsynth;
- int UserInterfaceMode;
- int VirKeybLayout;
- } cfg;
- int winwavemax,winmidimax;//number of wave/midi devices on Windows
- int maxstringsize;
-
- struct winmidionedevice{
- char *name;
- };
- winmidionedevice *winmididevices;
+class Config
+{
+public:
+ /** Constructor*/
+ Config();
+ /** Destructor*/
+ ~Config();
+ struct {
+ char *LinuxOSSWaveOutDev,*LinuxOSSSeqInDev;
+ int SampleRate,SoundBufferSize,OscilSize,SwapStereo;
+ int WindowsWaveOutId,WindowsMidiInId;
+ int BankUIAutoClose;
+ int DumpNotesToFile,DumpAppend;
+ int GzipCompression;
+ int Interpolation;
+ char *DumpFile;
+ char *bankRootDirList[MAX_BANK_ROOT_DIRS],*currentBankDir;
+ char *presetsDirList[MAX_BANK_ROOT_DIRS];
+ int CheckPADsynth;
+ int UserInterfaceMode;
+ int VirKeybLayout;
+ } cfg;
+ int winwavemax,winmidimax;//number of wave/midi devices on Windows
+ int maxstringsize;
+
+ struct winmidionedevice {
+ char *name;
+ };
+ winmidionedevice *winmididevices;
+
+ void clearbankrootdirlist();
+ void clearpresetsdirlist();
+ void init();
+ void save();
- void clearbankrootdirlist();
- void clearpresetsdirlist();
- void init();
- void save();
-
- private:
- void readConfig(const char *filename);
- void saveConfig(const char *filename);
- void getConfigFileName(char *name,int namesize);
+private:
+ void readConfig(const char *filename);
+ void saveConfig(const char *filename);
+ void getConfigFileName(char *name,int namesize);
};
#endif
diff --git a/src/Misc/Control.h b/src/Misc/Control.h
@@ -1,13 +1,13 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Control.h - Defines a variable that can be controled from a frontend
Copyright (C) 2009 Harald Hvaal
Author: Harald Hvaal
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,74 +27,74 @@
class Control
{
- public:
- /**
- * The parent is the logical owner of this control. Parent should only
- * be null for the root node.
- * The id is a string uniquely identifying this control within the
- * context of the parent control. No spaces or dots are allowed in this
- * id.
- * Children id's are denoted by <parent-id>.<children-id>, so that one
- * can refer to any control in the hierarchy by separating them with
- * dots. Example: Main.AddSynth.FrequencyLFO.Amplitude
- */
- Control(Control *parent, string id);
-
- /**
- * Will recursively get the XML representation for all the subcontrols.
- * Used for saving to file and copy-pasting settings
- */
- string getXMLRepresentation();
-
- /**
- * Set the value of this (and possibly subcomponents as well) based on
- * a xml description.
- */
- void restoreFromXML(string xml);
-
- /**
- * Register a controluser. This will cause this user to be notified
- * whenever the contents of the control changes.
- */
- void registerControlUser(ControlUser *user);
-
- /**
- * This should return a string representation of the controls internal
- * value
- */
- virtual string getStringRepresentation() = 0;
+public:
+ /**
+ * The parent is the logical owner of this control. Parent should only
+ * be null for the root node.
+ * The id is a string uniquely identifying this control within the
+ * context of the parent control. No spaces or dots are allowed in this
+ * id.
+ * Children id's are denoted by <parent-id>.<children-id>, so that one
+ * can refer to any control in the hierarchy by separating them with
+ * dots. Example: Main.AddSynth.FrequencyLFO.Amplitude
+ */
+ Control(Control *parent, string id);
+
+ /**
+ * Will recursively get the XML representation for all the subcontrols.
+ * Used for saving to file and copy-pasting settings
+ */
+ string getXMLRepresentation();
+
+ /**
+ * Set the value of this (and possibly subcomponents as well) based on
+ * a xml description.
+ */
+ void restoreFromXML(string xml);
+
+ /**
+ * Register a controluser. This will cause this user to be notified
+ * whenever the contents of the control changes.
+ */
+ void registerControlUser(ControlUser *user);
+
+ /**
+ * This should return a string representation of the controls internal
+ * value
+ */
+ virtual string getStringRepresentation() = 0;
};
class FloatControl : public Control
{
- public:
- /**
- * Set the value of this control. If the ControlUser variable is set,
- * then this user will not be updated with the new value. This is to
- * avoid setting a value being set back to the source that set it
- * (which would be redundant, or possibly causing infinite setValue
- * loops).
- * NOTE: this function is thread-safe (using a mutex internally)
- */
- void setValue(float value, ControlUser *user = NULL);
-
- /**
- * Reimplemented from Control
- */
- virtual string getStringRepresentation();
-
- float value();
+public:
+ /**
+ * Set the value of this control. If the ControlUser variable is set,
+ * then this user will not be updated with the new value. This is to
+ * avoid setting a value being set back to the source that set it
+ * (which would be redundant, or possibly causing infinite setValue
+ * loops).
+ * NOTE: this function is thread-safe (using a mutex internally)
+ */
+ void setValue(float value, ControlUser *user = NULL);
+
+ /**
+ * Reimplemented from Control
+ */
+ virtual string getStringRepresentation();
+
+ float value();
};
class ControlUser
{
- public:
- /**
- * Pure virtual method, to notify the controluser that the value has
- * been changed internally, and needs to be read again.
- */
- virtual void controlUpdated(Control *control) = 0;
+public:
+ /**
+ * Pure virtual method, to notify the controluser that the value has
+ * been changed internally, and needs to be read again.
+ */
+ virtual void controlUpdated(Control *control) = 0;
};
#endif /* _CONTROL_H_ */
diff --git a/src/Misc/Dump.C b/src/Misc/Dump.C
@@ -1,13 +1,13 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Dump.C - It dumps the notes to a text file
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,72 +26,80 @@
Dump dump;
-Dump::Dump(){
+Dump::Dump()
+{
file=NULL;
tick=0;
k=0;
keyspressed=0;
};
-Dump::~Dump(){
+Dump::~Dump()
+{
if (file!=NULL) {
- double duration=(double)tick*(double) SOUND_BUFFER_SIZE/(double) SAMPLE_RATE;
- fprintf(file,"\n# statistics: duration = %d seconds; keyspressed = %d\n\n\n\n",(int) duration,keyspressed);
- fclose(file);
+ double duration=(double)tick*(double) SOUND_BUFFER_SIZE/(double) SAMPLE_RATE;
+ fprintf(file,"\n# statistics: duration = %d seconds; keyspressed = %d\n\n\n\n",(int) duration,keyspressed);
+ fclose(file);
};
};
-void Dump::startnow(){
+void Dump::startnow()
+{
if (file!=NULL) return;//the file is already open
- if (config.cfg.DumpNotesToFile!=0){
- if (config.cfg.DumpAppend!=0) file=fopen(config.cfg.DumpFile,"a");
- else file=fopen(config.cfg.DumpFile,"w");
- if (file==NULL) return;
- if (config.cfg.DumpAppend!=0) fprintf(file,"%s","#************************************\n");
-
- time_t tm=time(NULL);
-
- fprintf(file,"#date/time = %s\n",ctime(&tm));
- fprintf(file,"#1 tick = %g milliseconds\n",SOUND_BUFFER_SIZE*1000.0/SAMPLE_RATE);
- fprintf(file,"SAMPLERATE = %d\n",SAMPLE_RATE);
- fprintf(file,"TICKSIZE = %d #samples\n",SOUND_BUFFER_SIZE);
- fprintf(file,"\n\nSTART\n");
+ if (config.cfg.DumpNotesToFile!=0) {
+ if (config.cfg.DumpAppend!=0) file=fopen(config.cfg.DumpFile,"a");
+ else file=fopen(config.cfg.DumpFile,"w");
+ if (file==NULL) return;
+ if (config.cfg.DumpAppend!=0) fprintf(file,"%s","#************************************\n");
+
+ time_t tm=time(NULL);
+
+ fprintf(file,"#date/time = %s\n",ctime(&tm));
+ fprintf(file,"#1 tick = %g milliseconds\n",SOUND_BUFFER_SIZE*1000.0/SAMPLE_RATE);
+ fprintf(file,"SAMPLERATE = %d\n",SAMPLE_RATE);
+ fprintf(file,"TICKSIZE = %d #samples\n",SOUND_BUFFER_SIZE);
+ fprintf(file,"\n\nSTART\n");
};
};
-void Dump::inctick(){
+void Dump::inctick()
+{
tick++;
};
-void Dump::dumpnote(char chan,char note, char vel){
+void Dump::dumpnote(char chan,char note, char vel)
+{
if (file==NULL) return;
if (note==0) return;
if (vel==0) fprintf(file,"n %d -> %d %d \n",tick,chan,note);//note off
- else fprintf(file,"N %d -> %d %d %d \n",tick,chan,note,vel);//note on
-
+ else fprintf(file,"N %d -> %d %d %d \n",tick,chan,note,vel);//note on
+
if (vel!=0) keyspressed++;
#ifndef JACKAUDIOOUT
if (k++>25) {
- fflush(file);
- k=0;
+ fflush(file);
+ k=0;
};
#endif
};
-void Dump::dumpcontroller(char chan,unsigned int type,int par){
+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;
+ 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;
};
#ifndef JACKAUDIOOUT
if (k++>25) {
- fflush(file);
- k=0;
+ fflush(file);
+ k=0;
};
#endif
};
diff --git a/src/Misc/Dump.h b/src/Misc/Dump.h
@@ -1,13 +1,13 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Dump.h - It dumps the notes to a text file
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,35 +28,36 @@
* \todo see if this object should have knowledge about the file
* that it will write to
* \todo upgrade from stdio to iostream*/
-class Dump{
- public:
- /**Constructor*/
- Dump();
- /**Destructor
- * Closes the dumpfile*/
- ~Dump();
- /**Open dumpfile and prepare it for dumps
- * \todo see if this fits better in the constructor*/
- void startnow();
- /**Tick the timestamp*/
- void inctick();
- /**Dump Note to dumpfile
- * @param chan The channel of the note
- * @param note The note
- * @param vel The velocity of the note*/
- void dumpnote(char chan,char note, char vel);
- /** Dump the Controller
- * @param chan The channel of the Controller
- * @param type The type
- * @param par The value of the controller
- * \todo figure out what type is exactly meaning*/
- void dumpcontroller(char chan,unsigned int type,int par);
-
- private:
- FILE *file;
- int tick;
- int k;//This appears to be a constant used to flush the file
- //periodically when JACK is used
- int keyspressed;
+class Dump
+{
+public:
+ /**Constructor*/
+ Dump();
+ /**Destructor
+ * Closes the dumpfile*/
+ ~Dump();
+ /**Open dumpfile and prepare it for dumps
+ * \todo see if this fits better in the constructor*/
+ void startnow();
+ /**Tick the timestamp*/
+ void inctick();
+ /**Dump Note to dumpfile
+ * @param chan The channel of the note
+ * @param note The note
+ * @param vel The velocity of the note*/
+ void dumpnote(char chan,char note, char vel);
+ /** Dump the Controller
+ * @param chan The channel of the Controller
+ * @param type The type
+ * @param par The value of the controller
+ * \todo figure out what type is exactly meaning*/
+ void dumpcontroller(char chan,unsigned int type,int par);
+
+private:
+ FILE *file;
+ int tick;
+ int k;//This appears to be a constant used to flush the file
+ //periodically when JACK is used
+ int keyspressed;
};
#endif
diff --git a/src/Misc/LASHClient.C b/src/Misc/LASHClient.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
LASHClient.C - LASH support
Copyright (C) 2006-2009 Lars Luthman
Author: Lars Luthman
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,74 +26,79 @@
#include "LASHClient.h"
-LASHClient::LASHClient(int* argc, char*** argv) {
- client = lash_init(lash_extract_args(argc, argv), "ZynAddSubFX",
- LASH_Config_File, LASH_PROTOCOL(2, 0));
+LASHClient::LASHClient(int* argc, char*** argv)
+{
+ client = lash_init(lash_extract_args(argc, argv), "ZynAddSubFX",
+ LASH_Config_File, LASH_PROTOCOL(2, 0));
}
-void LASHClient::setalsaid(int id) {
- if (lash_enabled(client)) {
- if (id != -1)
- lash_alsa_client_id(client, id);
- }
+void LASHClient::setalsaid(int id)
+{
+ if (lash_enabled(client)) {
+ if (id != -1)
+ lash_alsa_client_id(client, id);
+ }
}
-void LASHClient::setjackname(const char* name) {
- if (lash_enabled(client)) {
- if (name != NULL){
- lash_jack_client_name(client, name);
+void LASHClient::setjackname(const char* name)
+{
+ if (lash_enabled(client)) {
+ if (name != NULL) {
+ lash_jack_client_name(client, name);
- lash_event_t *event = lash_event_new_with_type(LASH_Client_Name);
- lash_event_set_string(event, name);
- lash_send_event(client, event);
- }
- }
+ lash_event_t *event = lash_event_new_with_type(LASH_Client_Name);
+ lash_event_set_string(event, name);
+ lash_send_event(client, event);
+ }
+ }
}
-LASHClient::Event LASHClient::checkevents(std::string& filename) {
-
- if (!lash_enabled(client))
- return NoEvent;
-
- Event received = NoEvent;
- lash_event_t* event;
- while (event = lash_get_event(client)) {
-
- // save
- if (lash_event_get_type(event) == LASH_Save_File) {
- std::cerr<<"LASH event: LASH_Save_File"<<std::endl;
- filename = std::string(lash_event_get_string(event)) + "/master.xmz";
- received = Save;
- break;
- }
-
- // restore
- else if (lash_event_get_type(event) == LASH_Restore_File) {
- std::cerr<<"LASH event: LASH_Restore_File"<<std::endl;
- filename = std::string(lash_event_get_string(event)) + "/master.xmz";
- received = Restore;
- break;
- }
-
- // quit
- else if (lash_event_get_type(event) == LASH_Quit) {
- std::cerr<<"LASH event: LASH_Quit"<<std::endl;
- received = Quit;
- break;
+LASHClient::Event LASHClient::checkevents(std::string& filename)
+{
+
+ if (!lash_enabled(client))
+ return NoEvent;
+
+ Event received = NoEvent;
+ lash_event_t* event;
+ while (event = lash_get_event(client)) {
+
+ // save
+ if (lash_event_get_type(event) == LASH_Save_File) {
+ std::cerr<<"LASH event: LASH_Save_File"<<std::endl;
+ filename = std::string(lash_event_get_string(event)) + "/master.xmz";
+ received = Save;
+ break;
+ }
+
+ // restore
+ else if (lash_event_get_type(event) == LASH_Restore_File) {
+ std::cerr<<"LASH event: LASH_Restore_File"<<std::endl;
+ filename = std::string(lash_event_get_string(event)) + "/master.xmz";
+ received = Restore;
+ break;
+ }
+
+ // quit
+ else if (lash_event_get_type(event) == LASH_Quit) {
+ std::cerr<<"LASH event: LASH_Quit"<<std::endl;
+ received = Quit;
+ break;
+ }
+
+ lash_event_destroy(event);
}
-
- lash_event_destroy(event);
- }
- return received;
+ return received;
}
-void LASHClient::confirmevent(Event event) {
- if (event == Save)
- lash_send_event(client, lash_event_new_with_type(LASH_Save_File));
- else if (event == Restore)
- lash_send_event(client, lash_event_new_with_type(LASH_Restore_File));
+void LASHClient::confirmevent(Event event)
+{
+ if (event == Save)
+ lash_send_event(client, lash_event_new_with_type(LASH_Save_File));
+ else 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
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
LASHClient.h - LASH support
Copyright (C) 2006-2009 Lars Luthman
Author: Lars Luthman
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -31,33 +31,34 @@
* the LASH daemon.
* \todo fix indentation nonconformism
* \todo see why there is no destructor*/
-class LASHClient {
- public:
- /**Enum to represent the LASH events that are currently handled*/
- enum Event {
- Save,
- Restore,
- Quit,
- NoEvent
- };
+class LASHClient
+{
+public:
+ /**Enum to represent the LASH events that are currently handled*/
+ enum Event {
+ Save,
+ Restore,
+ Quit,
+ NoEvent
+ };
+
+ /** Constructor
+ * @param argc number of arguments
+ * @param argv the text arguments*/
+ LASHClient(int* argc, char*** argv);
+
+ /**set the ALSA id
+ * @param id new ALSA id*/
+ void setalsaid(int id);
+ /**Set the JACK name
+ * @param name the new name*/
+ void setjackname(const char* name);
+ Event checkevents(std::string& filename);
+ void confirmevent(Event event);
+
+private:
- /** Constructor
- * @param argc number of arguments
- * @param argv the text arguments*/
- LASHClient(int* argc, char*** argv);
-
- /**set the ALSA id
- * @param id new ALSA id*/
- void setalsaid(int id);
- /**Set the JACK name
- * @param name the new name*/
- void setjackname(const char* name);
- Event checkevents(std::string& filename);
- void confirmevent(Event event);
-
- private:
-
- lash_client_t* client;
+ lash_client_t* client;
};
diff --git a/src/Misc/Master.C b/src/Misc/Master.C
@@ -1,13 +1,13 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Master.C - It sends Midi Messages to Parts, receives samples from parts,
process them with system/insertion effects and mix them
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,9 +29,10 @@
#include <unistd.h>
-Master::Master(){
+Master::Master()
+{
swaplr=0;
-
+
pthread_mutex_init(&mutex,NULL);
fft=new FFTwrapper(OSCIL_SIZE);
@@ -39,77 +40,80 @@ Master::Master(){
tmpmixr=new REALTYPE[SOUND_BUFFER_SIZE];
audiooutl=new REALTYPE[SOUND_BUFFER_SIZE];
audiooutr=new REALTYPE[SOUND_BUFFER_SIZE];
-
+
ksoundbuffersample=-1;//this is only time when this is -1; this means that the GetAudioOutSamples was never called
ksoundbuffersamplelow=0.0;
- oldsamplel=0.0;oldsampler=0.0;
+ oldsamplel=0.0;
+ oldsampler=0.0;
shutup=0;
for (int npart=0;npart<NUM_MIDI_PARTS;npart++) {
- vuoutpeakpart[npart]=1e-9;
- fakepeakpart[npart]=0;
+ vuoutpeakpart[npart]=1e-9;
+ fakepeakpart[npart]=0;
};
-
+
for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
- audiooutl[i]=0.0;
- audiooutr[i]=0.0;
+ audiooutl[i]=0.0;
+ audiooutr[i]=0.0;
};
for (int npart=0;npart<NUM_MIDI_PARTS;npart++)
- part[npart]=new Part(µtonal,fft,&mutex);
-
+ part[npart]=new Part(µtonal,fft,&mutex);
- //Insertion Effects init
+
+ //Insertion Effects init
for (int nefx=0;nefx<NUM_INS_EFX;nefx++)
- insefx[nefx]=new EffectMgr(1,&mutex);
+ insefx[nefx]=new EffectMgr(1,&mutex);
- //System Effects init
+ //System Effects init
for (int nefx=0;nefx<NUM_SYS_EFX;nefx++) {
- sysefx[nefx]=new EffectMgr(0,&mutex);
+ sysefx[nefx]=new EffectMgr(0,&mutex);
};
-
+
defaults();
};
-void Master::defaults(){
- volume=1.0;
- setPvolume(80);
- setPkeyshift(64);
-
- for (int npart=0;npart<NUM_MIDI_PARTS;npart++){
- part[npart]->defaults();
- part[npart]->Prcvchn=npart%NUM_MIDI_CHANNELS;
- };
-
- partonoff(0,1);//enable the first part
-
- for (int nefx=0;nefx<NUM_INS_EFX;nefx++) {
- insefx[nefx]->defaults();
- Pinsparts[nefx]=-1;
- };
-
- //System Effects init
- for (int nefx=0;nefx<NUM_SYS_EFX;nefx++) {
- sysefx[nefx]->defaults();
- for (int npart=0;npart<NUM_MIDI_PARTS;npart++){
- //if (nefx==0) setPsysefxvol(npart,nefx,64);
- //else
- setPsysefxvol(npart,nefx,0);
- };
- for (int nefxto=0;nefxto<NUM_SYS_EFX;nefxto++)
- setPsysefxsend(nefx,nefxto,0);
- };
+void Master::defaults()
+{
+ volume=1.0;
+ setPvolume(80);
+ setPkeyshift(64);
+
+ for (int npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ part[npart]->defaults();
+ part[npart]->Prcvchn=npart%NUM_MIDI_CHANNELS;
+ };
+
+ partonoff(0,1);//enable the first part
+
+ for (int nefx=0;nefx<NUM_INS_EFX;nefx++) {
+ insefx[nefx]->defaults();
+ Pinsparts[nefx]=-1;
+ };
+
+ //System Effects init
+ for (int nefx=0;nefx<NUM_SYS_EFX;nefx++) {
+ sysefx[nefx]->defaults();
+ for (int npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ //if (nefx==0) setPsysefxvol(npart,nefx,64);
+ //else
+ setPsysefxvol(npart,nefx,0);
+ };
+ for (int nefxto=0;nefxto<NUM_SYS_EFX;nefxto++)
+ setPsysefxsend(nefx,nefxto,0);
+ };
// sysefx[0]->changeeffect(1);
- microtonal.defaults();
- ShutUp();
+ microtonal.defaults();
+ ShutUp();
};
/*
* Note On Messages (velocity=0 for NoteOff)
*/
-void Master::NoteOn(unsigned char chan,unsigned char note,unsigned char velocity){
+void Master::NoteOn(unsigned char chan,unsigned char note,unsigned char velocity)
+{
dump.dumpnote(chan,note,velocity);
noteon(chan,note,velocity);
@@ -118,82 +122,87 @@ void Master::NoteOn(unsigned char chan,unsigned char note,unsigned char velocity
/*
* Internal Note On (velocity=0 for NoteOff)
*/
-void Master::noteon(unsigned char chan,unsigned char note,unsigned char velocity){
+void Master::noteon(unsigned char chan,unsigned char note,unsigned char velocity)
+{
int npart;
- if (velocity!=0){
- for (npart=0;npart<NUM_MIDI_PARTS;npart++){
- if (chan==part[npart]->Prcvchn){
- fakepeakpart[npart]=velocity*2;
- if (part[npart]->Penabled!=0) part[npart]->NoteOn(note,velocity,keyshift);
- };
- };
- }else{
- this->NoteOff(chan,note);
- };
- HDDRecorder.triggernow();
+ if (velocity!=0) {
+ for (npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ if (chan==part[npart]->Prcvchn) {
+ fakepeakpart[npart]=velocity*2;
+ if (part[npart]->Penabled!=0) part[npart]->NoteOn(note,velocity,keyshift);
+ };
+ };
+ } else {
+ this->NoteOff(chan,note);
+ };
+ HDDRecorder.triggernow();
};
/*
* Note Off Messages
*/
-void Master::NoteOff(unsigned char chan,unsigned char note){
+void Master::NoteOff(unsigned char chan,unsigned char note)
+{
dump.dumpnote(chan,note,0);
-
+
noteoff(chan,note);
};
/*
* Internal Note Off
*/
-void Master::noteoff(unsigned char chan,unsigned char note){
+void Master::noteoff(unsigned char chan,unsigned char note)
+{
int npart;
- for (npart=0;npart<NUM_MIDI_PARTS;npart++){
- if ((chan==part[npart]->Prcvchn) && (part[npart]->Penabled!=0))
- part[npart]->NoteOff(note);
+ for (npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ if ((chan==part[npart]->Prcvchn) && (part[npart]->Penabled!=0))
+ part[npart]->NoteOff(note);
};
};
/*
- * Controllers
+ * Controllers
*/
-void Master::SetController(unsigned char chan,unsigned int type,int par){
+void Master::SetController(unsigned char chan,unsigned int type,int par)
+{
dump.dumpcontroller(chan,type,par);
-
+
setcontroller(chan,type,par);
};
/*
- * Internal Controllers
+ * Internal Controllers
*/
-void Master::setcontroller(unsigned char chan,unsigned int type,int par){
+void Master::setcontroller(unsigned char chan,unsigned int type,int par)
+{
if ((type==C_dataentryhi)||(type==C_dataentrylo)||
- (type==C_nrpnhi)||(type==C_nrpnlo)){//Process RPN and NRPN by the Master (ignore the chan)
- ctl.setparameternumber(type,par);
-
- int parhi=-1,parlo=-1,valhi=-1,vallo=-1;
- 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;
-
- };
- };
+ (type==C_nrpnhi)||(type==C_nrpnlo)) {//Process RPN and NRPN by the Master (ignore the chan)
+ ctl.setparameternumber(type,par);
+
+ int parhi=-1,parlo=-1,valhi=-1,vallo=-1;
+ 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;
+
+ };
+ };
} else {//other controllers
- for (int npart=0;npart<NUM_MIDI_PARTS;npart++){//Send the controller to all part assigned to the channel
- if ((chan==part[npart]->Prcvchn) && (part[npart]->Penabled!=0))
- part[npart]->SetController(type,par);
- };
-
- if(type==C_allsoundsoff) { //cleanup insertion/system FX
+ for (int npart=0;npart<NUM_MIDI_PARTS;npart++) {//Send the controller to all part assigned to the channel
+ if ((chan==part[npart]->Prcvchn) && (part[npart]->Penabled!=0))
+ part[npart]->SetController(type,par);
+ };
+
+ if (type==C_allsoundsoff) { //cleanup insertion/system FX
for (int nefx=0;nefx<NUM_SYS_EFX;++nefx) {
sysefx[nefx]->cleanup();
}
@@ -208,171 +217,173 @@ void Master::setcontroller(unsigned char chan,unsigned int type,int par){
/*
* Enable/Disable a part
*/
-void Master::partonoff(int npart,int what){
+void Master::partonoff(int npart,int what)
+{
if (npart>=NUM_MIDI_PARTS) return;
- if (what==0){//disable part
- fakepeakpart[npart]=0;
- part[npart]->Penabled=0;
- part[npart]->cleanup();
- for (int nefx=0;nefx<NUM_INS_EFX;nefx++){
- if (Pinsparts[nefx]==npart) {
- insefx[nefx]->cleanup();
- };
- };
+ if (what==0) {//disable part
+ fakepeakpart[npart]=0;
+ part[npart]->Penabled=0;
+ part[npart]->cleanup();
+ for (int nefx=0;nefx<NUM_INS_EFX;nefx++) {
+ if (Pinsparts[nefx]==npart) {
+ insefx[nefx]->cleanup();
+ };
+ };
} else {//enabled
- part[npart]->Penabled=1;
- fakepeakpart[npart]=0;
+ part[npart]->Penabled=1;
+ fakepeakpart[npart]=0;
};
};
/*
* Master audio out (the final sound)
*/
-void Master::AudioOut(REALTYPE *outl,REALTYPE *outr){
+void Master::AudioOut(REALTYPE *outl,REALTYPE *outr)
+{
int i,npart,nefx;
-/* //test!!!!!!!!!!!!! se poate bloca aici (mutex)
- if (seq.play){
- int type,par1,par2,again,midichan;
- int ntrack=1;
-// do{
- again=seq.getevent(ntrack,&midichan,&type,&par1,&par2);
- if (type>0) {
-// printf("aaa\n");
-
- if (type==1){//note_on or note_off
- if (par2!=0) NoteOn(midichan,par1,par2);
- else NoteOff(midichan,par1);
- };
- };
-// } while (again);
- };
-*/
+ /* //test!!!!!!!!!!!!! se poate bloca aici (mutex)
+ if (seq.play){
+ int type,par1,par2,again,midichan;
+ int ntrack=1;
+ // do{
+ again=seq.getevent(ntrack,&midichan,&type,&par1,&par2);
+ if (type>0) {
+ // printf("aaa\n");
+
+ if (type==1){//note_on or note_off
+ if (par2!=0) NoteOn(midichan,par1,par2);
+ else NoteOff(midichan,par1);
+ };
+ };
+ // } while (again);
+ };
+ */
// printf("zzzz\n");
//Swaps the Left channel with Right Channel (if it is asked for)
- if (swaplr!=0){
+ if (swaplr!=0) {
REALTYPE *tmp=outl;
- outl=outr;
- outr=tmp;
+ outl=outr;
+ outr=tmp;
};
-
- //clean up the output samples
+
+ //clean up the output samples
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- outl[i]=0.0;
- outr[i]=0.0;
+ outl[i]=0.0;
+ outr[i]=0.0;
};
//Compute part samples and store them part[npart]->partoutl,partoutr
for (npart=0;npart<NUM_MIDI_PARTS;npart++)
- if (part[npart]->Penabled!=0) part[npart]->ComputePartSmps();
-
- //Insertion effects
- for (nefx=0;nefx<NUM_INS_EFX;nefx++){
- if (Pinsparts[nefx]>=0) {
- int efxpart=Pinsparts[nefx];
- if (part[efxpart]->Penabled!=0)
- insefx[nefx]->out(part[efxpart]->partoutl,part[efxpart]->partoutr);
- };
- };
-
-
+ if (part[npart]->Penabled!=0) part[npart]->ComputePartSmps();
+
+ //Insertion effects
+ for (nefx=0;nefx<NUM_INS_EFX;nefx++) {
+ if (Pinsparts[nefx]>=0) {
+ int efxpart=Pinsparts[nefx];
+ if (part[efxpart]->Penabled!=0)
+ insefx[nefx]->out(part[efxpart]->partoutl,part[efxpart]->partoutr);
+ };
+ };
+
+
//Apply the part volumes and pannings (after insertion effects)
- for (npart=0;npart<NUM_MIDI_PARTS;npart++){
- if (part[npart]->Penabled==0) continue;
-
- REALTYPE newvol_l=part[npart]->volume;
- REALTYPE newvol_r=part[npart]->volume;
- REALTYPE oldvol_l=part[npart]->oldvolumel;
- REALTYPE oldvol_r=part[npart]->oldvolumer;
- REALTYPE pan=part[npart]->panning;
- if (pan<0.5) newvol_l*=pan*2.0;
- else newvol_r*=(1.0-pan)*2.0;
-
- if (ABOVE_AMPLITUDE_THRESHOLD(oldvol_l,newvol_l)||
- ABOVE_AMPLITUDE_THRESHOLD(oldvol_r,newvol_r)){//the volume or the panning has changed and needs interpolation
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- REALTYPE vol_l=INTERPOLATE_AMPLITUDE(oldvol_l,newvol_l,i,SOUND_BUFFER_SIZE);
- REALTYPE vol_r=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 {
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {//the volume did not changed
- part[npart]->partoutl[i]*=newvol_l;
- part[npart]->partoutr[i]*=newvol_r;
- };
- };
+ for (npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ if (part[npart]->Penabled==0) continue;
+
+ REALTYPE newvol_l=part[npart]->volume;
+ REALTYPE newvol_r=part[npart]->volume;
+ REALTYPE oldvol_l=part[npart]->oldvolumel;
+ REALTYPE oldvol_r=part[npart]->oldvolumer;
+ REALTYPE pan=part[npart]->panning;
+ if (pan<0.5) newvol_l*=pan*2.0;
+ else newvol_r*=(1.0-pan)*2.0;
+
+ if (ABOVE_AMPLITUDE_THRESHOLD(oldvol_l,newvol_l)||
+ ABOVE_AMPLITUDE_THRESHOLD(oldvol_r,newvol_r)) {//the volume or the panning has changed and needs interpolation
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE vol_l=INTERPOLATE_AMPLITUDE(oldvol_l,newvol_l,i,SOUND_BUFFER_SIZE);
+ REALTYPE vol_r=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 {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {//the volume did not changed
+ part[npart]->partoutl[i]*=newvol_l;
+ part[npart]->partoutr[i]*=newvol_r;
+ };
+ };
};
//System effects
- for (nefx=0;nefx<NUM_SYS_EFX;nefx++){
- if (sysefx[nefx]->geteffect()==0) continue;//the effect is disabled
-
- //Clean up the samples used by the system effects
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- tmpmixl[i]=0.0;
- tmpmixr[i]=0.0;
- };
-
- //Mix the channels according to the part settings about System Effect
- for (npart=0;npart<NUM_MIDI_PARTS;npart++){
- //skip if the part has no output to effect
- if (Psysefxvol[nefx][npart]==0) continue;
-
- //skip if the part is disabled
- if (part[npart]->Penabled==0) continue;
-
- //the output volume of each part to system effect
- REALTYPE vol=sysefxvol[nefx][npart];
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- tmpmixl[i]+=part[npart]->partoutl[i]*vol;
- tmpmixr[i]+=part[npart]->partoutr[i]*vol;
- };
- };
-
- // system effect send to next ones
- for (int nefxfrom=0;nefxfrom<nefx;nefxfrom++){
- if (Psysefxsend[nefxfrom][nefx]!=0){
- REALTYPE v=sysefxsend[nefxfrom][nefx];
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- tmpmixl[i]+=sysefx[nefxfrom]->efxoutl[i]*v;
- tmpmixr[i]+=sysefx[nefxfrom]->efxoutr[i]*v;
- };
- };
- };
-
- sysefx[nefx]->out(tmpmixl,tmpmixr);
-
- //Add the System Effect to sound output
- REALTYPE outvol=sysefx[nefx]->sysefxgetvolume();
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- outl[i]+=tmpmixl[i]*outvol;
- outr[i]+=tmpmixr[i]*outvol;
- };
+ for (nefx=0;nefx<NUM_SYS_EFX;nefx++) {
+ if (sysefx[nefx]->geteffect()==0) continue;//the effect is disabled
+
+ //Clean up the samples used by the system effects
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpmixl[i]=0.0;
+ tmpmixr[i]=0.0;
+ };
+
+ //Mix the channels according to the part settings about System Effect
+ for (npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ //skip if the part has no output to effect
+ if (Psysefxvol[nefx][npart]==0) continue;
+
+ //skip if the part is disabled
+ if (part[npart]->Penabled==0) continue;
+
+ //the output volume of each part to system effect
+ REALTYPE vol=sysefxvol[nefx][npart];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpmixl[i]+=part[npart]->partoutl[i]*vol;
+ tmpmixr[i]+=part[npart]->partoutr[i]*vol;
+ };
+ };
+
+ // system effect send to next ones
+ for (int nefxfrom=0;nefxfrom<nefx;nefxfrom++) {
+ if (Psysefxsend[nefxfrom][nefx]!=0) {
+ REALTYPE v=sysefxsend[nefxfrom][nefx];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpmixl[i]+=sysefx[nefxfrom]->efxoutl[i]*v;
+ tmpmixr[i]+=sysefx[nefxfrom]->efxoutr[i]*v;
+ };
+ };
+ };
+
+ sysefx[nefx]->out(tmpmixl,tmpmixr);
+
+ //Add the System Effect to sound output
+ REALTYPE outvol=sysefx[nefx]->sysefxgetvolume();
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]+=tmpmixl[i]*outvol;
+ outr[i]+=tmpmixr[i]*outvol;
+ };
};
//Mix all parts
- for (npart=0;npart<NUM_MIDI_PARTS;npart++){
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {//the volume did not changed
- outl[i]+=part[npart]->partoutl[i];
- outr[i]+=part[npart]->partoutr[i];
- };
+ for (npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ for (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 (nefx=0;nefx<NUM_INS_EFX;nefx++){
- if (Pinsparts[nefx] == -2)
- insefx[nefx]->out(outl,outr);
+ for (nefx=0;nefx<NUM_INS_EFX;nefx++) {
+ if (Pinsparts[nefx] == -2)
+ insefx[nefx]->out(outl,outr);
};
//Master Volume
@@ -382,7 +393,8 @@ void Master::AudioOut(REALTYPE *outl,REALTYPE *outr){
};
//Peak computation (for vumeters)
- vuoutpeakl=1e-12;vuoutpeakr=1e-12;
+ vuoutpeakl=1e-12;
+ vuoutpeakr=1e-12;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
if (fabs(outl[i])>vuoutpeakl) vuoutpeakl=fabs(outl[i]);
if (fabs(outr[i])>vuoutpeakr) vuoutpeakr=fabs(outr[i]);
@@ -392,39 +404,40 @@ void Master::AudioOut(REALTYPE *outl,REALTYPE *outr){
if (vumaxoutpeakr<vuoutpeakr) vumaxoutpeakr=vuoutpeakr;
//RMS Peak computation (for vumeters)
- vurmspeakl=1e-12;vurmspeakr=1e-12;
+ vurmspeakl=1e-12;
+ vurmspeakr=1e-12;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
vurmspeakl+=outl[i]*outl[i];
vurmspeakr+=outr[i]*outr[i];
};
vurmspeakl=sqrt(vurmspeakl/SOUND_BUFFER_SIZE);
vurmspeakr=sqrt(vurmspeakr/SOUND_BUFFER_SIZE);
-
+
//Part Peak computation (for Part vumeters or fake part vumeters)
- for (npart=0;npart<NUM_MIDI_PARTS;npart++){
- vuoutpeakpart[npart]=1.0e-12;
- if (part[npart]->Penabled!=0) {
- REALTYPE *outl=part[npart]->partoutl,
- *outr=part[npart]->partoutr;
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- REALTYPE tmp=fabs(outl[i]+outr[i]);
- if (tmp>vuoutpeakpart[npart]) vuoutpeakpart[npart]=tmp;
- };
- vuoutpeakpart[npart]*=volume;
- } else {
- if (fakepeakpart[npart]>1) fakepeakpart[npart]--;
- };
+ for (npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ vuoutpeakpart[npart]=1.0e-12;
+ if (part[npart]->Penabled!=0) {
+ REALTYPE *outl=part[npart]->partoutl,
+ *outr=part[npart]->partoutr;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE tmp=fabs(outl[i]+outr[i]);
+ if (tmp>vuoutpeakpart[npart]) vuoutpeakpart[npart]=tmp;
+ };
+ vuoutpeakpart[npart]*=volume;
+ } else {
+ if (fakepeakpart[npart]>1) fakepeakpart[npart]--;
+ };
};
//Shutup if it is asked (with fade-out)
- if (shutup!=0){
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- REALTYPE tmp=(SOUND_BUFFER_SIZE-i)/(REALTYPE) SOUND_BUFFER_SIZE;
- outl[i]*=tmp;
- outr[i]*=tmp;
- };
- ShutUp();
+ if (shutup!=0) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE tmp=(SOUND_BUFFER_SIZE-i)/(REALTYPE) SOUND_BUFFER_SIZE;
+ outl[i]*=tmp;
+ outr[i]*=tmp;
+ };
+ ShutUp();
};
//update the LFO's time
@@ -434,67 +447,69 @@ void Master::AudioOut(REALTYPE *outl,REALTYPE *outr){
dump.inctick();
};
-void Master::GetAudioOutSamples(int nsamples,int samplerate,REALTYPE *outl,REALTYPE *outr){
- if (ksoundbuffersample==-1){//first time
- AudioOut(&audiooutl[0],&audiooutr[0]);
- ksoundbuffersample=0;
+void Master::GetAudioOutSamples(int nsamples,int samplerate,REALTYPE *outl,REALTYPE *outr)
+{
+ if (ksoundbuffersample==-1) {//first time
+ AudioOut(&audiooutl[0],&audiooutr[0]);
+ ksoundbuffersample=0;
};
- if (samplerate==SAMPLE_RATE){//no resample
- int ksample=0;
- while (ksample<nsamples){
- outl[ksample]=audiooutl[ksoundbuffersample];
- outr[ksample]=audiooutr[ksoundbuffersample];
+ if (samplerate==SAMPLE_RATE) {//no resample
+ int ksample=0;
+ while (ksample<nsamples) {
+ outl[ksample]=audiooutl[ksoundbuffersample];
+ outr[ksample]=audiooutr[ksoundbuffersample];
- ksample++;
- ksoundbuffersample++;
- if (ksoundbuffersample>=SOUND_BUFFER_SIZE){
- AudioOut(&audiooutl[0],&audiooutr[0]);
- ksoundbuffersample=0;
- };
- };
+ ksample++;
+ ksoundbuffersample++;
+ if (ksoundbuffersample>=SOUND_BUFFER_SIZE) {
+ AudioOut(&audiooutl[0],&audiooutr[0]);
+ ksoundbuffersample=0;
+ };
+ };
} else {//Resample
- int ksample=0;
- REALTYPE srinc=SAMPLE_RATE/(REALTYPE)samplerate;
-
- while (ksample<nsamples){
- if (ksoundbuffersample!=0){
- outl[ksample]=audiooutl[ksoundbuffersample]*ksoundbuffersamplelow
- +audiooutl[ksoundbuffersample-1]*(1.0-ksoundbuffersamplelow);
- outr[ksample]=audiooutr[ksoundbuffersample]*ksoundbuffersamplelow
- +audiooutr[ksoundbuffersample-1]*(1.0-ksoundbuffersamplelow);
- } else {
- outl[ksample]=audiooutl[ksoundbuffersample]*ksoundbuffersamplelow
- +oldsamplel*(1.0-ksoundbuffersamplelow);
- outr[ksample]=audiooutr[ksoundbuffersample]*ksoundbuffersamplelow
- +oldsampler*(1.0-ksoundbuffersamplelow);
- };
-
- ksample++;
-
- ksoundbuffersamplelow+=srinc;
- if (ksoundbuffersamplelow>=1.0){
- ksoundbuffersample+=(int) floor(ksoundbuffersamplelow);
- ksoundbuffersamplelow=ksoundbuffersamplelow-floor(ksoundbuffersamplelow);
- };
-
- if (ksoundbuffersample>=SOUND_BUFFER_SIZE){
- oldsamplel=audiooutl[SOUND_BUFFER_SIZE-1];
- oldsampler=audiooutr[SOUND_BUFFER_SIZE-1];
- AudioOut(&audiooutl[0],&audiooutr[0]);
- ksoundbuffersample=0;
- };
- };
+ int ksample=0;
+ REALTYPE srinc=SAMPLE_RATE/(REALTYPE)samplerate;
+
+ while (ksample<nsamples) {
+ if (ksoundbuffersample!=0) {
+ outl[ksample]=audiooutl[ksoundbuffersample]*ksoundbuffersamplelow
+ +audiooutl[ksoundbuffersample-1]*(1.0-ksoundbuffersamplelow);
+ outr[ksample]=audiooutr[ksoundbuffersample]*ksoundbuffersamplelow
+ +audiooutr[ksoundbuffersample-1]*(1.0-ksoundbuffersamplelow);
+ } else {
+ outl[ksample]=audiooutl[ksoundbuffersample]*ksoundbuffersamplelow
+ +oldsamplel*(1.0-ksoundbuffersamplelow);
+ outr[ksample]=audiooutr[ksoundbuffersample]*ksoundbuffersamplelow
+ +oldsampler*(1.0-ksoundbuffersamplelow);
+ };
+
+ ksample++;
+
+ ksoundbuffersamplelow+=srinc;
+ if (ksoundbuffersamplelow>=1.0) {
+ ksoundbuffersample+=(int) floor(ksoundbuffersamplelow);
+ ksoundbuffersamplelow=ksoundbuffersamplelow-floor(ksoundbuffersamplelow);
+ };
+
+ if (ksoundbuffersample>=SOUND_BUFFER_SIZE) {
+ oldsamplel=audiooutl[SOUND_BUFFER_SIZE-1];
+ oldsampler=audiooutr[SOUND_BUFFER_SIZE-1];
+ AudioOut(&audiooutl[0],&audiooutr[0]);
+ ksoundbuffersample=0;
+ };
+ };
};
};
-Master::~Master(){
+Master::~Master()
+{
for (int npart=0;npart<NUM_MIDI_PARTS;npart++) delete part[npart];
for (int nefx=0;nefx<NUM_INS_EFX;nefx++) delete insefx[nefx];
for (int nefx=0;nefx<NUM_SYS_EFX;nefx++) delete sysefx[nefx];
-
+
delete [] audiooutl;
delete [] audiooutr;
delete [] tmpmixl;
@@ -508,23 +523,27 @@ Master::~Master(){
/*
* Parameter control
*/
-void Master::setPvolume(char Pvolume_){
+void Master::setPvolume(char Pvolume_)
+{
Pvolume=Pvolume_;
volume=dB2rap((Pvolume-96.0)/96.0*40.0);
};
-void Master::setPkeyshift(char Pkeyshift_){
+void Master::setPkeyshift(char Pkeyshift_)
+{
Pkeyshift=Pkeyshift_;
keyshift=(int)Pkeyshift-64;
};
-void Master::setPsysefxvol(int Ppart,int Pefx,char Pvol){
+void Master::setPsysefxvol(int Ppart,int Pefx,char Pvol)
+{
Psysefxvol[Pefx][Ppart]=Pvol;
sysefxvol[Pefx][Ppart]=pow(0.1,(1.0-Pvol/96.0)*2.0);
};
-void Master::setPsysefxsend(int Pefxfrom,int Pefxto,char Pvol){
+void Master::setPsysefxsend(int Pefxfrom,int Pefxto,char Pvol)
+{
Psysefxsend[Pefxfrom][Pefxto]=Pvol;
sysefxsend[Pefxfrom][Pefxto]=pow(0.1,(1.0-Pvol/96.0)*2.0);
};
@@ -533,10 +552,11 @@ void Master::setPsysefxsend(int Pefxfrom,int Pefxto,char Pvol){
/*
* Panic! (Clean up all parts and effects)
*/
-void Master::ShutUp(){
+void Master::ShutUp()
+{
for (int npart=0;npart<NUM_MIDI_PARTS;npart++) {
- part[npart]->cleanup();
- fakepeakpart[npart]=0;
+ part[npart]->cleanup();
+ fakepeakpart[npart]=0;
};
for (int nefx=0;nefx<NUM_INS_EFX;nefx++) insefx[nefx]->cleanup();
for (int nefx=0;nefx<NUM_SYS_EFX;nefx++) sysefx[nefx]->cleanup();
@@ -548,75 +568,82 @@ void Master::ShutUp(){
/*
* Reset peaks and clear the "cliped" flag (for VU-meter)
*/
-void Master::vuresetpeaks(){
- vuoutpeakl=1e-9;vuoutpeakr=1e-9;vumaxoutpeakl=1e-9;vumaxoutpeakr=1e-9;
+void Master::vuresetpeaks()
+{
+ vuoutpeakl=1e-9;
+ vuoutpeakr=1e-9;
+ vumaxoutpeakl=1e-9;
+ vumaxoutpeakr=1e-9;
vuclipped=0;
};
-void Master::applyparameters(){
- for (int npart=0;npart<NUM_MIDI_PARTS;npart++){
- part[npart]->applyparameters();
+void Master::applyparameters()
+{
+ for (int npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ part[npart]->applyparameters();
};
};
-void Master::add2XML(XMLwrapper *xml){
+void Master::add2XML(XMLwrapper *xml)
+{
xml->addpar("volume",Pvolume);
xml->addpar("key_shift",Pkeyshift);
xml->addparbool("nrpn_receive",ctl.NRPN.receive);
xml->beginbranch("MICROTONAL");
- microtonal.add2XML(xml);
+ microtonal.add2XML(xml);
xml->endbranch();
- for (int npart=0;npart<NUM_MIDI_PARTS;npart++){
- xml->beginbranch("PART",npart);
- part[npart]->add2XML(xml);
- xml->endbranch();
+ for (int npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ xml->beginbranch("PART",npart);
+ part[npart]->add2XML(xml);
+ xml->endbranch();
};
-
+
xml->beginbranch("SYSTEM_EFFECTS");
- for (int nefx=0;nefx<NUM_SYS_EFX;nefx++){
- xml->beginbranch("SYSTEM_EFFECT",nefx);
- xml->beginbranch("EFFECT");
- sysefx[nefx]->add2XML(xml);
- xml->endbranch();
-
- for (int pefx=0;pefx<NUM_MIDI_PARTS;pefx++){
- xml->beginbranch("VOLUME",pefx);
- xml->addpar("vol",Psysefxvol[nefx][pefx]);
- xml->endbranch();
- };
-
- for (int tonefx=nefx+1;tonefx<NUM_SYS_EFX;tonefx++){
- xml->beginbranch("SENDTO",tonefx);
- xml->addpar("send_vol",Psysefxsend[nefx][tonefx]);
- xml->endbranch();
- };
-
-
- xml->endbranch();
- };
+ for (int nefx=0;nefx<NUM_SYS_EFX;nefx++) {
+ xml->beginbranch("SYSTEM_EFFECT",nefx);
+ xml->beginbranch("EFFECT");
+ sysefx[nefx]->add2XML(xml);
+ xml->endbranch();
+
+ for (int pefx=0;pefx<NUM_MIDI_PARTS;pefx++) {
+ xml->beginbranch("VOLUME",pefx);
+ xml->addpar("vol",Psysefxvol[nefx][pefx]);
+ xml->endbranch();
+ };
+
+ for (int tonefx=nefx+1;tonefx<NUM_SYS_EFX;tonefx++) {
+ xml->beginbranch("SENDTO",tonefx);
+ xml->addpar("send_vol",Psysefxsend[nefx][tonefx]);
+ xml->endbranch();
+ };
+
+
+ xml->endbranch();
+ };
xml->endbranch();
xml->beginbranch("INSERTION_EFFECTS");
- for (int nefx=0;nefx<NUM_INS_EFX;nefx++){
- xml->beginbranch("INSERTION_EFFECT",nefx);
- xml->addpar("part",Pinsparts[nefx]);
-
- xml->beginbranch("EFFECT");
- insefx[nefx]->add2XML(xml);
- xml->endbranch();
- xml->endbranch();
- };
-
+ for (int nefx=0;nefx<NUM_INS_EFX;nefx++) {
+ xml->beginbranch("INSERTION_EFFECT",nefx);
+ xml->addpar("part",Pinsparts[nefx]);
+
+ xml->beginbranch("EFFECT");
+ insefx[nefx]->add2XML(xml);
+ xml->endbranch();
+ xml->endbranch();
+ };
+
xml->endbranch();
-
+
};
-int Master::getalldata(char **data){
+int Master::getalldata(char **data)
+{
XMLwrapper *xml=new XMLwrapper();
xml->beginbranch("MASTER");
@@ -632,25 +659,27 @@ int Master::getalldata(char **data){
return(strlen(*data)+1);
};
-void Master::putalldata(char *data,int size){
+void Master::putalldata(char *data,int size)
+{
XMLwrapper *xml=new XMLwrapper();
if (!xml->putXMLdata(data)) {
- delete(xml);
- return;
+ delete(xml);
+ return;
};
-
+
if (xml->enterbranch("MASTER")==0) return;
pthread_mutex_lock(&mutex);
- getfromXML(xml);
+ getfromXML(xml);
pthread_mutex_unlock(&mutex);
xml->exitbranch();
-
+
delete(xml);
};
-int Master::saveXML(const char *filename){
+int Master::saveXML(const char *filename)
+{
XMLwrapper *xml=new XMLwrapper();
xml->beginbranch("MASTER");
@@ -664,82 +693,84 @@ int Master::saveXML(const char *filename){
-int Master::loadXML(const char *filename){
+int Master::loadXML(const char *filename)
+{
XMLwrapper *xml=new XMLwrapper();
if (xml->loadXMLfile(filename)<0) {
- delete(xml);
- return(-1);
+ delete(xml);
+ return(-1);
};
-
+
if (xml->enterbranch("MASTER")==0) return(-10);
- getfromXML(xml);
+ getfromXML(xml);
xml->exitbranch();
-
+
delete(xml);
return(0);
};
-void Master::getfromXML(XMLwrapper *xml){
+void Master::getfromXML(XMLwrapper *xml)
+{
setPvolume(xml->getpar127("volume",Pvolume));
setPkeyshift(xml->getpar127("key_shift",Pkeyshift));
ctl.NRPN.receive=xml->getparbool("nrpn_receive",ctl.NRPN.receive);
-
-
+
+
part[0]->Penabled=0;
- for (int npart=0;npart<NUM_MIDI_PARTS;npart++){
- if (xml->enterbranch("PART",npart)==0) continue;
- part[npart]->getfromXML(xml);
- xml->exitbranch();
+ for (int npart=0;npart<NUM_MIDI_PARTS;npart++) {
+ if (xml->enterbranch("PART",npart)==0) continue;
+ part[npart]->getfromXML(xml);
+ xml->exitbranch();
};
- if (xml->enterbranch("MICROTONAL")){
+ if (xml->enterbranch("MICROTONAL")) {
microtonal.getfromXML(xml);
xml->exitbranch();
};
-
+
sysefx[0]->changeeffect(0);
- if (xml->enterbranch("SYSTEM_EFFECTS")){
- for (int nefx=0;nefx<NUM_SYS_EFX;nefx++){
- if (xml->enterbranch("SYSTEM_EFFECT",nefx)==0) continue;
- if (xml->enterbranch("EFFECT")){
- sysefx[nefx]->getfromXML(xml);
- xml->exitbranch();
- };
-
- for (int partefx=0;partefx<NUM_MIDI_PARTS;partefx++){
- if (xml->enterbranch("VOLUME",partefx)==0) continue;
- setPsysefxvol(partefx,nefx,xml->getpar127("vol",Psysefxvol[partefx][nefx]));
- xml->exitbranch();
- };
-
- for (int tonefx=nefx+1;tonefx<NUM_SYS_EFX;tonefx++){
- if (xml->enterbranch("SENDTO",tonefx)==0) continue;
- setPsysefxsend(nefx,tonefx,xml->getpar127("send_vol",Psysefxsend[nefx][tonefx]));
- xml->exitbranch();
- };
- xml->exitbranch();
- };
- xml->exitbranch();
- };
-
-
- if (xml->enterbranch("INSERTION_EFFECTS")){
- for (int nefx=0;nefx<NUM_INS_EFX;nefx++){
-
- if (xml->enterbranch("INSERTION_EFFECT",nefx)==0) continue;
- Pinsparts[nefx]=xml->getpar("part",Pinsparts[nefx],-2,NUM_MIDI_PARTS);
- if (xml->enterbranch("EFFECT")){
- insefx[nefx]->getfromXML(xml);
- xml->exitbranch();
- };
- xml->exitbranch();
-
- };
-
- xml->exitbranch();
- };
-
-
+ if (xml->enterbranch("SYSTEM_EFFECTS")) {
+ for (int nefx=0;nefx<NUM_SYS_EFX;nefx++) {
+ if (xml->enterbranch("SYSTEM_EFFECT",nefx)==0) continue;
+ if (xml->enterbranch("EFFECT")) {
+ sysefx[nefx]->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ for (int partefx=0;partefx<NUM_MIDI_PARTS;partefx++) {
+ if (xml->enterbranch("VOLUME",partefx)==0) continue;
+ setPsysefxvol(partefx,nefx,xml->getpar127("vol",Psysefxvol[partefx][nefx]));
+ xml->exitbranch();
+ };
+
+ for (int tonefx=nefx+1;tonefx<NUM_SYS_EFX;tonefx++) {
+ if (xml->enterbranch("SENDTO",tonefx)==0) continue;
+ setPsysefxsend(nefx,tonefx,xml->getpar127("send_vol",Psysefxsend[nefx][tonefx]));
+ xml->exitbranch();
+ };
+ xml->exitbranch();
+ };
+ xml->exitbranch();
+ };
+
+
+ if (xml->enterbranch("INSERTION_EFFECTS")) {
+ for (int nefx=0;nefx<NUM_INS_EFX;nefx++) {
+
+ if (xml->enterbranch("INSERTION_EFFECT",nefx)==0) continue;
+ Pinsparts[nefx]=xml->getpar("part",Pinsparts[nefx],-2,NUM_MIDI_PARTS);
+ if (xml->enterbranch("EFFECT")) {
+ insefx[nefx]->getfromXML(xml);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
+
+ };
+
+ xml->exitbranch();
+ };
+
+
};
diff --git a/src/Misc/Master.h b/src/Misc/Master.h
@@ -1,13 +1,13 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Master.h - It sends Midi Messages to Parts, receives samples from parts,
process them with system/insertion effects and mix them
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -38,129 +38,130 @@
extern Dump dump;
/** It sends Midi Messages to Parts, receives samples from parts,
* process them with system/insertion effects and mix them */
-class Master{
- public:
- /** Constructor*/
- Master();
- /** Destructor*/
- ~Master();
-
- /**Saves all settings to a XML file
- * @return 0 for ok or <0 if there is an error*/
- int saveXML(const char *filename);
-
- /**This adds the parameters to the XML data*/
- void add2XML(XMLwrapper *xml);
-
- void defaults();
-
-
- /**loads all settings from a XML file
- * @return 0 for ok or -1 if there is an error*/
- int loadXML(const char *filename);
- void applyparameters();
-
- void getfromXML(XMLwrapper *xml);
-
- /**get all data to a newly allocated array (used for VST)
- * @return the datasize*/
- int getalldata(char **data);
- /**put all data from the *data array to zynaddsubfx parameters (used for VST)*/
- void putalldata(char *data,int size);
-
-
-
- //Midi IN
- void NoteOn(unsigned char chan,unsigned char note,unsigned char velocity);
- void NoteOff(unsigned char chan,unsigned char note);
- void SetController(unsigned char chan,unsigned int type,int par);
- //void NRPN...
-
-
- void ShutUp();
- int shutup;
-
- /**Audio Output*/
- void AudioOut(REALTYPE *outl,REALTYPE *outr);
- /**Audio Output (for callback mode). This allows the program to be controled by an external program*/
- void GetAudioOutSamples(int nsamples,int samplerate,REALTYPE *outl,REALTYPE *outr);
-
-
- void partonoff(int npart,int what);
-
- /**parts \todo see if this can be made to be dynamic*/
- Part *part[NUM_MIDI_PARTS];
-
- //parameters
- unsigned char Pvolume;
- unsigned char Pkeyshift;
- unsigned char Psysefxvol[NUM_SYS_EFX][NUM_MIDI_PARTS];
- unsigned char Psysefxsend[NUM_SYS_EFX][NUM_SYS_EFX];
-
- //parameters control
- void setPvolume(char Pvolume_);
- void setPkeyshift(char Pkeyshift_);
- void setPsysefxvol(int Ppart,int Pefx,char Pvol);
- void setPsysefxsend(int Pefxfrom,int Pefxto,char Pvol);
-
- //effects
- EffectMgr *sysefx[NUM_SYS_EFX];//system
- EffectMgr *insefx[NUM_INS_EFX];//insertion
+class Master
+{
+public:
+ /** Constructor*/
+ Master();
+ /** Destructor*/
+ ~Master();
+
+ /**Saves all settings to a XML file
+ * @return 0 for ok or <0 if there is an error*/
+ int saveXML(const char *filename);
+
+ /**This adds the parameters to the XML data*/
+ void add2XML(XMLwrapper *xml);
+
+ void defaults();
+
+
+ /**loads all settings from a XML file
+ * @return 0 for ok or -1 if there is an error*/
+ int loadXML(const char *filename);
+ void applyparameters();
+
+ void getfromXML(XMLwrapper *xml);
+
+ /**get all data to a newly allocated array (used for VST)
+ * @return the datasize*/
+ int getalldata(char **data);
+ /**put all data from the *data array to zynaddsubfx parameters (used for VST)*/
+ void putalldata(char *data,int size);
+
+
+
+ //Midi IN
+ void NoteOn(unsigned char chan,unsigned char note,unsigned char velocity);
+ void NoteOff(unsigned char chan,unsigned char note);
+ void SetController(unsigned char chan,unsigned int type,int par);
+ //void NRPN...
+
+
+ void ShutUp();
+ int shutup;
+
+ /**Audio Output*/
+ void AudioOut(REALTYPE *outl,REALTYPE *outr);
+ /**Audio Output (for callback mode). This allows the program to be controled by an external program*/
+ void GetAudioOutSamples(int nsamples,int samplerate,REALTYPE *outl,REALTYPE *outr);
+
+
+ void partonoff(int npart,int what);
+
+ /**parts \todo see if this can be made to be dynamic*/
+ Part *part[NUM_MIDI_PARTS];
+
+ //parameters
+ unsigned char Pvolume;
+ unsigned char Pkeyshift;
+ unsigned char Psysefxvol[NUM_SYS_EFX][NUM_MIDI_PARTS];
+ unsigned char Psysefxsend[NUM_SYS_EFX][NUM_SYS_EFX];
+
+ //parameters control
+ void setPvolume(char Pvolume_);
+ void setPkeyshift(char Pkeyshift_);
+ void setPsysefxvol(int Ppart,int Pefx,char Pvol);
+ void setPsysefxsend(int Pefxfrom,int Pefxto,char Pvol);
+
+ //effects
+ EffectMgr *sysefx[NUM_SYS_EFX];//system
+ EffectMgr *insefx[NUM_INS_EFX];//insertion
// void swapcopyeffects(int what,int type,int neff1,int neff2);
- //HDD recorder
- Recorder HDDRecorder;
-
- //part that's apply the insertion effect; -1 to disable
- short int Pinsparts[NUM_INS_EFX];
-
- //peaks for VU-meter
- void vuresetpeaks();
- REALTYPE vuoutpeakl,vuoutpeakr,vumaxoutpeakl,vumaxoutpeakr,vurmspeakl,vurmspeakr;
- int vuclipped;
-
- //peaks for part VU-meters
- REALTYPE vuoutpeakpart[NUM_MIDI_PARTS];
- unsigned char fakepeakpart[NUM_MIDI_PARTS];//this is used to compute the "peak" when the part is disabled
-
- Controller ctl;
- int swaplr;//1 if L and R are swapped
-
- //Sequencer
- Sequencer seq;
-
- //other objects
- Microtonal microtonal;
- Bank bank;
-
- FFTwrapper *fft;
- pthread_mutex_t mutex;
-
- private:
- REALTYPE volume;
- REALTYPE sysefxvol[NUM_SYS_EFX][NUM_MIDI_PARTS];
- REALTYPE sysefxsend[NUM_SYS_EFX][NUM_SYS_EFX];
-
- //Temporary mixing samples for part samples which is sent to system effect
- REALTYPE *tmpmixl;
- REALTYPE *tmpmixr;
-
-
- int keyshift;
-
- //Audio Output samples (if it used GetAudioOutSamples - eg. for Jack output; elsewhere is unused)
- REALTYPE *audiooutl;
- REALTYPE *audiooutr;
-
- int ksoundbuffersample;//this is used to know if there is need to call AudioOut by GetAudioOutSamples method
- REALTYPE ksoundbuffersamplelow;//this is used for resampling (eg. if Jack samplerate!= SAMPLE_RATE)
- REALTYPE oldsamplel,oldsampler;//this is used for resampling
-
- //These are called by the NoteOn, NoteOff,SetController (which are from external sources like MIDI, Virtual Keyboard)
- //and are called by internal parts of the program (like sequencer)
- void noteon(unsigned char chan,unsigned char note,unsigned char velocity);
- void noteoff(unsigned char chan,unsigned char note);
- void setcontroller(unsigned char chan,unsigned int type,int par);
+ //HDD recorder
+ Recorder HDDRecorder;
+
+ //part that's apply the insertion effect; -1 to disable
+ short int Pinsparts[NUM_INS_EFX];
+
+ //peaks for VU-meter
+ void vuresetpeaks();
+ REALTYPE vuoutpeakl,vuoutpeakr,vumaxoutpeakl,vumaxoutpeakr,vurmspeakl,vurmspeakr;
+ int vuclipped;
+
+ //peaks for part VU-meters
+ REALTYPE vuoutpeakpart[NUM_MIDI_PARTS];
+ unsigned char fakepeakpart[NUM_MIDI_PARTS];//this is used to compute the "peak" when the part is disabled
+
+ Controller ctl;
+ int swaplr;//1 if L and R are swapped
+
+ //Sequencer
+ Sequencer seq;
+
+ //other objects
+ Microtonal microtonal;
+ Bank bank;
+
+ FFTwrapper *fft;
+ pthread_mutex_t mutex;
+
+private:
+ REALTYPE volume;
+ REALTYPE sysefxvol[NUM_SYS_EFX][NUM_MIDI_PARTS];
+ REALTYPE sysefxsend[NUM_SYS_EFX][NUM_SYS_EFX];
+
+ //Temporary mixing samples for part samples which is sent to system effect
+ REALTYPE *tmpmixl;
+ REALTYPE *tmpmixr;
+
+
+ int keyshift;
+
+ //Audio Output samples (if it used GetAudioOutSamples - eg. for Jack output; elsewhere is unused)
+ REALTYPE *audiooutl;
+ REALTYPE *audiooutr;
+
+ int ksoundbuffersample;//this is used to know if there is need to call AudioOut by GetAudioOutSamples method
+ REALTYPE ksoundbuffersamplelow;//this is used for resampling (eg. if Jack samplerate!= SAMPLE_RATE)
+ REALTYPE oldsamplel,oldsampler;//this is used for resampling
+
+ //These are called by the NoteOn, NoteOff,SetController (which are from external sources like MIDI, Virtual Keyboard)
+ //and are called by internal parts of the program (like sequencer)
+ void noteon(unsigned char chan,unsigned char note,unsigned char velocity);
+ void noteoff(unsigned char chan,unsigned char note);
+ void setcontroller(unsigned char chan,unsigned int type,int par);
};
diff --git a/src/Misc/Microtonal.C b/src/Misc/Microtonal.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Microtonal.C - Tuning settings and microtonal capabilities
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,13 +26,15 @@
#define MAX_LINE_SIZE 80
-Microtonal::Microtonal(){
+Microtonal::Microtonal()
+{
Pname=new unsigned char[MICROTONAL_MAX_NAME_LEN];
Pcomment=new unsigned char[MICROTONAL_MAX_NAME_LEN];
defaults();
};
-void Microtonal::defaults(){
+void Microtonal::defaults()
+{
Pinvertupdown=0;
Pinvertupdowncenter=60;
octavesize=12;
@@ -40,30 +42,35 @@ void Microtonal::defaults(){
PAnote=69;
PAfreq=440.0;
Pscaleshift=64;
-
- Pfirstkey=0;Plastkey=127;
- Pmiddlenote=60;Pmapsize=12;
+
+ Pfirstkey=0;
+ Plastkey=127;
+ Pmiddlenote=60;
+ Pmapsize=12;
Pmappingenabled=0;
-
+
for (int i=0;i<128;i++) Pmapping[i]=i;
-
- for (int i=0;i<MAX_OCTAVE_SIZE;i++){
- octave[i].tuning=tmpoctave[i].tuning=pow(2,(i%octavesize+1)/12.0);
- octave[i].type=tmpoctave[i].type=1;
- octave[i].x1=tmpoctave[i].x1=(i%octavesize+1)*100;
- octave[i].x2=tmpoctave[i].x2=0;
+
+ for (int i=0;i<MAX_OCTAVE_SIZE;i++) {
+ octave[i].tuning=tmpoctave[i].tuning=pow(2,(i%octavesize+1)/12.0);
+ 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;octave[11].x2=1;
- for (int i=0;i<MICROTONAL_MAX_NAME_LEN;i++){
- Pname[i]='\0';
- Pcomment[i]='\0';
+ octave[11].type=2;
+ octave[11].x1=2;
+ octave[11].x2=1;
+ for (int i=0;i<MICROTONAL_MAX_NAME_LEN;i++) {
+ Pname[i]='\0';
+ Pcomment[i]='\0';
};
snprintf((char *) Pname,MICROTONAL_MAX_NAME_LEN,"12tET");
- snprintf((char *) Pcomment,MICROTONAL_MAX_NAME_LEN,"Equal Temperament 12 notes per octave");
+ snprintf((char *) Pcomment,MICROTONAL_MAX_NAME_LEN,"Equal Temperament 12 notes per octave");
Pglobalfinedetune=64;
};
-Microtonal::~Microtonal(){
+Microtonal::~Microtonal()
+{
delete [] Pname;
delete [] Pcomment;
};
@@ -71,86 +78,91 @@ Microtonal::~Microtonal(){
/*
* Get the size of the octave
*/
-unsigned char Microtonal::getoctavesize(){
+unsigned char Microtonal::getoctavesize()
+{
if (Penabled!=0) return(octavesize);
- else return(12);
+ else return(12);
};
/*
* Get the frequency according the note number
*/
-REALTYPE Microtonal::getnotefreq(int note,int keyshift){
+REALTYPE Microtonal::getnotefreq(int note,int keyshift)
+{
// in this function will appears many times things like this:
- // var=(a+b*100)%b
+ // var=(a+b*100)%b
// I had written this way because if I use var=a%b gives unwanted results when a<0
// This is the same with divisions.
-
+
if ((Pinvertupdown!=0)&&((Pmappingenabled==0)||(Penabled==0))) note=(int) Pinvertupdowncenter*2-note;
//compute global fine detune
REALTYPE globalfinedetunerap=pow(2.0,(Pglobalfinedetune-64.0)/1200.0);//-64.0 .. 63.0 cents
-
+
if (Penabled==0) return(pow(2.0,(note-PAnote+keyshift)/12.0)*PAfreq*globalfinedetunerap);//12tET
-
+
int scaleshift=((int)Pscaleshift-64+(int) octavesize*100)%octavesize;
//compute the keyshift
REALTYPE rap_keyshift=1.0;
- if (keyshift!=0){
- int kskey=(keyshift+(int)octavesize*100)%octavesize;
- int ksoct=(keyshift+(int)octavesize*100)/octavesize-100;
- rap_keyshift=(kskey==0) ? (1.0):(octave[kskey-1].tuning);
- rap_keyshift*=pow(octave[octavesize-1].tuning,ksoct);
+ if (keyshift!=0) {
+ int kskey=(keyshift+(int)octavesize*100)%octavesize;
+ int ksoct=(keyshift+(int)octavesize*100)/octavesize-100;
+ rap_keyshift=(kskey==0) ? (1.0):(octave[kskey-1].tuning);
+ rap_keyshift*=pow(octave[octavesize-1].tuning,ksoct);
};
//if the mapping is enabled
- if (Pmappingenabled!=0){
- if ((note<Pfirstkey)||(note>Plastkey)) return (-1.0);
- //Compute how many mapped keys are from middle note to reference note
- //and find out the proportion between the freq. of middle note and "A" note
- int tmp=PAnote-Pmiddlenote,minus=0;
- if (tmp<0) { tmp=-tmp; minus=1; };
- int deltanote=0;
- for (int i=0;i<tmp;i++) if (Pmapping[i%Pmapsize]>=0) deltanote++;
- REALTYPE rap_anote_middlenote=(deltanote==0) ? (1.0) : (octave[(deltanote-1)%octavesize].tuning);
- if (deltanote!=0) rap_anote_middlenote*=pow(octave[octavesize-1].tuning,(deltanote-1)/octavesize);
- if (minus!=0) rap_anote_middlenote=1.0/rap_anote_middlenote;
-
- //Convert from note (midi) to degree (note from the tunning)
- int degoct=(note-(int)Pmiddlenote+(int) Pmapsize*200)/(int)Pmapsize-200;
- int degkey=(note-Pmiddlenote+(int)Pmapsize*100)%Pmapsize;
- degkey=Pmapping[degkey];
- if (degkey<0) return(-1.0);//this key is not mapped
-
- //invert the keyboard upside-down if it is asked for
- //TODO: do the right way by using Pinvertupdowncenter
- if (Pinvertupdown!=0){
- degkey=octavesize-degkey-1;
- degoct=-degoct;
- };
- //compute the frequency of the note
- degkey=degkey+scaleshift;
- degoct+=degkey/octavesize;
- degkey%=octavesize;
-
- REALTYPE freq=(degkey==0) ? (1.0):octave[degkey-1].tuning;
- freq*=pow(octave[octavesize-1].tuning,degoct);
- freq*=PAfreq/rap_anote_middlenote;
- freq*=globalfinedetunerap;
- if (scaleshift!=0) freq/=octave[scaleshift-1].tuning;
- return(freq*rap_keyshift);
+ if (Pmappingenabled!=0) {
+ if ((note<Pfirstkey)||(note>Plastkey)) return (-1.0);
+ //Compute how many mapped keys are from middle note to reference note
+ //and find out the proportion between the freq. of middle note and "A" note
+ int tmp=PAnote-Pmiddlenote,minus=0;
+ if (tmp<0) {
+ tmp=-tmp;
+ minus=1;
+ };
+ int deltanote=0;
+ for (int i=0;i<tmp;i++) if (Pmapping[i%Pmapsize]>=0) deltanote++;
+ REALTYPE rap_anote_middlenote=(deltanote==0) ? (1.0) : (octave[(deltanote-1)%octavesize].tuning);
+ if (deltanote!=0) rap_anote_middlenote*=pow(octave[octavesize-1].tuning,(deltanote-1)/octavesize);
+ if (minus!=0) rap_anote_middlenote=1.0/rap_anote_middlenote;
+
+ //Convert from note (midi) to degree (note from the tunning)
+ int degoct=(note-(int)Pmiddlenote+(int) Pmapsize*200)/(int)Pmapsize-200;
+ int degkey=(note-Pmiddlenote+(int)Pmapsize*100)%Pmapsize;
+ degkey=Pmapping[degkey];
+ if (degkey<0) return(-1.0);//this key is not mapped
+
+ //invert the keyboard upside-down if it is asked for
+ //TODO: do the right way by using Pinvertupdowncenter
+ if (Pinvertupdown!=0) {
+ degkey=octavesize-degkey-1;
+ degoct=-degoct;
+ };
+ //compute the frequency of the note
+ degkey=degkey+scaleshift;
+ degoct+=degkey/octavesize;
+ degkey%=octavesize;
+
+ REALTYPE freq=(degkey==0) ? (1.0):octave[degkey-1].tuning;
+ freq*=pow(octave[octavesize-1].tuning,degoct);
+ freq*=PAfreq/rap_anote_middlenote;
+ freq*=globalfinedetunerap;
+ if (scaleshift!=0) freq/=octave[scaleshift-1].tuning;
+ 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;
-
- REALTYPE oct=octave[octavesize-1].tuning;
- REALTYPE freq=octave[(ntkey+octavesize-1)%octavesize].tuning*pow(oct,ntoct)*PAfreq;
- if (ntkey==0) freq/=oct;
+ int nt=note-PAnote+scaleshift;
+ int ntkey=(nt+(int)octavesize*100)%octavesize;
+ int ntoct=(nt-ntkey)/octavesize;
+
+ REALTYPE oct=octave[octavesize-1].tuning;
+ REALTYPE freq=octave[(ntkey+octavesize-1)%octavesize].tuning*pow(oct,ntoct)*PAfreq;
+ if (ntkey==0) freq/=oct;
if (scaleshift!=0) freq/=octave[scaleshift-1].tuning;
// fprintf(stderr,"note=%d freq=%.3f cents=%d\n",note,freq,(int)floor(log(freq/PAfreq)/log(2.0)*1200.0+0.5));
- freq*=globalfinedetunerap;
- return(freq*rap_keyshift);
+ freq*=globalfinedetunerap;
+ return(freq*rap_keyshift);
};
};
@@ -158,82 +170,86 @@ REALTYPE Microtonal::getnotefreq(int note,int keyshift){
/*
* Convert a line to tunings; returns -1 if it ok
*/
-int Microtonal::linetotunings(unsigned int nline,const char *line){
+int Microtonal::linetotunings(unsigned int nline,const char *line)
+{
int x1=-1,x2=-1,type=-1;
REALTYPE x=-1.0,tmp,tuning=1.0;
- if (strstr(line,"/")==NULL){
- if (strstr(line,".")==NULL){// M case (M=M/1)
- sscanf(line,"%d",&x1);
- x2=1;
- type=2;//division
- } else {// float number case
- sscanf(line,"%f",&x);
- if (x<0.000001) return(1);
- type=1;//float type(cents)
- };
+ if (strstr(line,"/")==NULL) {
+ if (strstr(line,".")==NULL) {// M case (M=M/1)
+ sscanf(line,"%d",&x1);
+ x2=1;
+ type=2;//division
+ } else {// float number case
+ sscanf(line,"%f",&x);
+ if (x<0.000001) return(1);
+ type=1;//float type(cents)
+ };
} else {// M/N case
- sscanf(line,"%d/%d",&x1,&x2);
- if ((x1<0)||(x2<0)) return(1);
- if (x2==0) x2=1;
- type=2;//division
+ sscanf(line,"%d/%d",&x1,&x2);
+ if ((x1<0)||(x2<0)) return(1);
+ if (x2==0) x2=1;
+ type=2;//division
};
-
+
if (x1<=0) x1=1;//not allow zero frequency sounds (consider 0 as 1)
-
+
//convert to float if the number are too big
- if ((type==2)&&((x1>(128*128*128-1))||(x2>(128*128*128-1)))){
- type=1;
- x=((REALTYPE) x1)/x2;
+ if ((type==2)&&((x1>(128*128*128-1))||(x2>(128*128*128-1)))) {
+ type=1;
+ x=((REALTYPE) x1)/x2;
};
- switch (type){
- case 1: x1=(int) floor(x);
- tmp=fmod(x,1.0);
- x2=(int) (floor (tmp*1e6));
- tuning=pow(2.0,x/1200.0);
- break;
- case 2: x=((REALTYPE)x1)/x2;
- tuning=x;
- break;
+ switch (type) {
+ case 1:
+ x1=(int) floor(x);
+ tmp=fmod(x,1.0);
+ x2=(int) (floor (tmp*1e6));
+ tuning=pow(2.0,x/1200.0);
+ break;
+ case 2:
+ x=((REALTYPE)x1)/x2;
+ tuning=x;
+ break;
};
-
+
tmpoctave[nline].tuning=tuning;
tmpoctave[nline].type=type;
tmpoctave[nline].x1=x1;
tmpoctave[nline].x2=x2;
-
+
return(-1);//ok
};
/*
* Convert the text to tunnings
*/
-int Microtonal::texttotunings(const char *text){
+int Microtonal::texttotunings(const char *text)
+{
unsigned int i,k=0,nl=0;
char *lin;
lin=new char[MAX_LINE_SIZE+1];
- while (k<strlen(text)){
- for (i=0;i<MAX_LINE_SIZE;i++){
- lin[i]=text[k++];
- if (lin[i]<0x20) break;
- };
- lin[i]='\0';
- if (strlen(lin)==0) continue;
- int err=linetotunings(nl,lin);
- if (err!=-1) {
- delete [] lin;
- return(nl);//Parse error
- };
- nl++;
+ while (k<strlen(text)) {
+ for (i=0;i<MAX_LINE_SIZE;i++) {
+ lin[i]=text[k++];
+ if (lin[i]<0x20) break;
+ };
+ lin[i]='\0';
+ if (strlen(lin)==0) continue;
+ int err=linetotunings(nl,lin);
+ if (err!=-1) {
+ delete [] lin;
+ return(nl);//Parse error
+ };
+ nl++;
};
delete [] lin;
if (nl>MAX_OCTAVE_SIZE) nl=MAX_OCTAVE_SIZE;
if (nl==0) return(-2);//the input is empty
octavesize=nl;
- for (i=0;i<octavesize;i++){
- octave[i].tuning=tmpoctave[i].tuning;
- octave[i].type=tmpoctave[i].type;
- octave[i].x1=tmpoctave[i].x1;
- octave[i].x2=tmpoctave[i].x2;
+ for (i=0;i<octavesize;i++) {
+ octave[i].tuning=tmpoctave[i].tuning;
+ octave[i].type=tmpoctave[i].type;
+ octave[i].x1=tmpoctave[i].x1;
+ octave[i].x2=tmpoctave[i].x2;
};
return(-1);//ok
};
@@ -241,29 +257,30 @@ int Microtonal::texttotunings(const char *text){
/*
* Convert the text to mapping
*/
-void Microtonal::texttomapping(const char *text){
+void Microtonal::texttomapping(const char *text)
+{
unsigned int i,k=0;
char *lin;
lin=new char[MAX_LINE_SIZE+1];
for (i=0;i<128;i++) Pmapping[i]=-1;
int tx=0;
- while (k<strlen(text)){
- for (i=0;i<MAX_LINE_SIZE;i++){
- lin[i]=text[k++];
- if (lin[i]<0x20) break;
- };
- lin[i]='\0';
- if (strlen(lin)==0) continue;
-
- int tmp=0;
- if (sscanf(lin,"%d",&tmp)==0) tmp=-1;
- if (tmp<-1) tmp=-1;
- Pmapping[tx]=tmp;
-
- if ((tx++)>127) break;
+ while (k<strlen(text)) {
+ for (i=0;i<MAX_LINE_SIZE;i++) {
+ lin[i]=text[k++];
+ if (lin[i]<0x20) break;
+ };
+ lin[i]='\0';
+ if (strlen(lin)==0) continue;
+
+ int tmp=0;
+ if (sscanf(lin,"%d",&tmp)==0) tmp=-1;
+ if (tmp<-1) tmp=-1;
+ Pmapping[tx]=tmp;
+
+ if ((tx++)>127) break;
};
delete [] lin;
-
+
if (tx==0) tx=1;
Pmapsize=tx;
};
@@ -271,26 +288,29 @@ void Microtonal::texttomapping(const char *text){
/*
* Convert tunning to text line
*/
-void Microtonal::tuningtoline(int n,char *line,int maxn){
+void Microtonal::tuningtoline(int n,char *line,int maxn)
+{
if ((n>octavesize) || (n>MAX_OCTAVE_SIZE)) {
- line[0]='\0';
- return;
+ line[0]='\0';
+ return;
};
if (octave[n].type==1) snprintf(line,maxn,"%d.%d",octave[n].x1,octave[n].x2);
if (octave[n].type==2) snprintf(line,maxn,"%d/%d",octave[n].x1,octave[n].x2);
};
-
-int Microtonal::loadline(FILE *file,char *line){
+
+int Microtonal::loadline(FILE *file,char *line)
+{
do {
- if (fgets(line,500,file)==0) return(1);
+ if (fgets(line,500,file)==0) return(1);
} while (line[0]=='!');
return(0);
};
/*
* Loads the tunnings from a scl file
*/
-int Microtonal::loadscl(const char *filename){
+int Microtonal::loadscl(const char *filename)
+{
FILE *file=fopen(filename, "r");
char tmp[500];
fseek(file,0,SEEK_SET);
@@ -305,18 +325,18 @@ int Microtonal::loadscl(const char *filename){
sscanf(&tmp[0],"%d",&nnotes);
if (nnotes>MAX_OCTAVE_SIZE) return (2);
//load the tunnings
- for (int nline=0;nline<nnotes;nline++){
- if (loadline(file,&tmp[0])!=0) return(2);
- linetotunings(nline,&tmp[0]);
+ for (int nline=0;nline<nnotes;nline++) {
+ if (loadline(file,&tmp[0])!=0) return(2);
+ linetotunings(nline,&tmp[0]);
};
fclose(file);
octavesize=nnotes;
- for (int i=0;i<octavesize;i++){
- octave[i].tuning=tmpoctave[i].tuning;
- octave[i].type=tmpoctave[i].type;
- octave[i].x1=tmpoctave[i].x1;
- octave[i].x2=tmpoctave[i].x2;
+ for (int i=0;i<octavesize;i++) {
+ octave[i].tuning=tmpoctave[i].tuning;
+ octave[i].type=tmpoctave[i].type;
+ octave[i].x1=tmpoctave[i].x1;
+ octave[i].x2=tmpoctave[i].x2;
};
return(0);
@@ -325,7 +345,8 @@ int Microtonal::loadscl(const char *filename){
/*
* Loads the mapping from a kbm file
*/
-int Microtonal::loadkbm(const char *filename){
+int Microtonal::loadkbm(const char *filename)
+{
FILE *file=fopen(filename, "r");
int x;
char tmp[500];
@@ -334,27 +355,32 @@ int Microtonal::loadkbm(const char *filename){
//loads the mapsize
if (loadline(file,&tmp[0])!=0) return(2);
if (sscanf(&tmp[0],"%d",&x)==0) return(2);
- if (x<1) x=0;if (x>127) x=127;//just in case...
+ if (x<1) x=0;
+ if (x>127) x=127;//just in case...
Pmapsize=x;
//loads first MIDI note to retune
if (loadline(file,&tmp[0])!=0) return(2);
if (sscanf(&tmp[0],"%d",&x)==0) return(2);
- if (x<1) x=0;if (x>127) x=127;//just in case...
+ if (x<1) x=0;
+ if (x>127) x=127;//just in case...
Pfirstkey=x;
//loads last MIDI note to retune
if (loadline(file,&tmp[0])!=0) return(2);
if (sscanf(&tmp[0],"%d",&x)==0) return(2);
- if (x<1) x=0;if (x>127) x=127;//just in case...
+ if (x<1) x=0;
+ if (x>127) x=127;//just in case...
Plastkey=x;
//loads last the middle note where scale fro scale degree=0
if (loadline(file,&tmp[0])!=0) return(2);
if (sscanf(&tmp[0],"%d",&x)==0) return(2);
- if (x<1) x=0;if (x>127) x=127;//just in case...
+ if (x<1) x=0;
+ if (x>127) x=127;//just in case...
Pmiddlenote=x;
//loads the reference note
if (loadline(file,&tmp[0])!=0) return(2);
if (sscanf(&tmp[0],"%d",&x)==0) return(2);
- if (x<1) x=0;if (x>127) x=127;//just in case...
+ if (x<1) x=0;
+ if (x>127) x=127;//just in case...
PAnote=x;
//loads the reference freq.
if (loadline(file,&tmp[0])!=0) return(2);
@@ -366,17 +392,17 @@ int Microtonal::loadkbm(const char *filename){
if (loadline(file,&tmp[0])!=0) return(2);
//load the mappings
- if (Pmapsize!=0){
- for (int nline=0;nline<Pmapsize;nline++){
- if (loadline(file,&tmp[0])!=0) return(2);
- if (sscanf(&tmp[0],"%d",&x)==0) x=-1;
- Pmapping[nline]=x;
- };
- Pmappingenabled=1;
+ if (Pmapsize!=0) {
+ for (int nline=0;nline<Pmapsize;nline++) {
+ if (loadline(file,&tmp[0])!=0) return(2);
+ if (sscanf(&tmp[0],"%d",&x)==0) x=-1;
+ Pmapping[nline]=x;
+ };
+ Pmappingenabled=1;
} else {
- Pmappingenabled=0;
- Pmapping[0]=0;
- Pmapsize=1;
+ Pmappingenabled=0;
+ Pmapping[0]=0;
+ Pmapsize=1;
};
fclose(file);
@@ -385,7 +411,8 @@ int Microtonal::loadkbm(const char *filename){
-void Microtonal::add2XML(XMLwrapper *xml){
+void Microtonal::add2XML(XMLwrapper *xml)
+{
xml->addparstr("name",(char *) Pname);
xml->addparstr("comment",(char *) Pcomment);
@@ -401,39 +428,40 @@ void Microtonal::add2XML(XMLwrapper *xml){
if ((Penabled==0)&&(xml->minimal)) return;
xml->beginbranch("SCALE");
- xml->addpar("scale_shift",Pscaleshift);
- xml->addpar("first_key",Pfirstkey);
- xml->addpar("last_key",Plastkey);
- xml->addpar("middle_note",Pmiddlenote);
-
- xml->beginbranch("OCTAVE");
- xml->addpar("octave_size",octavesize);
- for (int i=0;i<octavesize;i++){
- xml->beginbranch("DEGREE",i);
- if (octave[i].type==1){
- xml->addparreal("cents",octave[i].tuning);
- };
- if (octave[i].type==2){
- xml->addpar("numerator",octave[i].x1);
- xml->addpar("denominator",octave[i].x2);
- };
- xml->endbranch();
- };
- xml->endbranch();
-
- xml->beginbranch("KEYBOARD_MAPPING");
- xml->addpar("map_size",Pmapsize);
- xml->addpar("mapping_enabled",Pmappingenabled);
- for (int i=0;i<Pmapsize;i++){
- xml->beginbranch("KEYMAP",i);
- xml->addpar("degree",Pmapping[i]);
- xml->endbranch();
- };
- xml->endbranch();
+ xml->addpar("scale_shift",Pscaleshift);
+ xml->addpar("first_key",Pfirstkey);
+ xml->addpar("last_key",Plastkey);
+ xml->addpar("middle_note",Pmiddlenote);
+
+ xml->beginbranch("OCTAVE");
+ xml->addpar("octave_size",octavesize);
+ for (int i=0;i<octavesize;i++) {
+ xml->beginbranch("DEGREE",i);
+ if (octave[i].type==1) {
+ xml->addparreal("cents",octave[i].tuning);
+ };
+ if (octave[i].type==2) {
+ xml->addpar("numerator",octave[i].x1);
+ xml->addpar("denominator",octave[i].x2);
+ };
+ xml->endbranch();
+ };
+ xml->endbranch();
+
+ xml->beginbranch("KEYBOARD_MAPPING");
+ xml->addpar("map_size",Pmapsize);
+ xml->addpar("mapping_enabled",Pmappingenabled);
+ for (int i=0;i<Pmapsize;i++) {
+ xml->beginbranch("KEYMAP",i);
+ xml->addpar("degree",Pmapping[i]);
+ xml->endbranch();
+ };
+ xml->endbranch();
xml->endbranch();
};
-void Microtonal::getfromXML(XMLwrapper *xml){
+void Microtonal::getfromXML(XMLwrapper *xml)
+{
xml->getparstr("name",(char *) Pname,MICROTONAL_MAX_NAME_LEN);
xml->getparstr("comment",(char *) Pcomment,MICROTONAL_MAX_NAME_LEN);
@@ -446,45 +474,46 @@ void Microtonal::getfromXML(XMLwrapper *xml){
PAnote=xml->getpar127("a_note",PAnote);
PAfreq=xml->getparreal("a_freq",PAfreq,1.0,10000.0);
- if (xml->enterbranch("SCALE")){
- Pscaleshift=xml->getpar127("scale_shift",Pscaleshift);
- Pfirstkey=xml->getpar127("first_key",Pfirstkey);
- Plastkey=xml->getpar127("last_key",Plastkey);
- Pmiddlenote=xml->getpar127("middle_note",Pmiddlenote);
-
- if (xml->enterbranch("OCTAVE")){
- octavesize=xml->getpar127("octave_size",octavesize);
- for (int i=0;i<octavesize;i++){
- if (xml->enterbranch("DEGREE",i)==0) continue;
- octave[i].x2=0;
- octave[i].tuning=xml->getparreal("cents",octave[i].tuning);
- octave[i].x1=xml->getpar127("numerator",octave[i].x1);
- octave[i].x2=xml->getpar127("denominator",octave[i].x2);
-
- if (octave[i].x2!=0) octave[i].type=2;
- else octave[i].type=1;
-
- xml->exitbranch();
- };
- xml->exitbranch();
- };
-
- if (xml->enterbranch("KEYBOARD_MAPPING")){
- Pmapsize=xml->getpar127("map_size",Pmapsize);
- Pmappingenabled=xml->getpar127("mapping_enabled",Pmappingenabled);
- for (int i=0;i<Pmapsize;i++){
- if (xml->enterbranch("KEYMAP",i)==0) continue;
- Pmapping[i]=xml->getpar127("degree",Pmapping[i]);
- xml->exitbranch();
- };
- xml->exitbranch();
- };
- xml->exitbranch();
+ if (xml->enterbranch("SCALE")) {
+ Pscaleshift=xml->getpar127("scale_shift",Pscaleshift);
+ Pfirstkey=xml->getpar127("first_key",Pfirstkey);
+ Plastkey=xml->getpar127("last_key",Plastkey);
+ Pmiddlenote=xml->getpar127("middle_note",Pmiddlenote);
+
+ if (xml->enterbranch("OCTAVE")) {
+ octavesize=xml->getpar127("octave_size",octavesize);
+ for (int i=0;i<octavesize;i++) {
+ if (xml->enterbranch("DEGREE",i)==0) continue;
+ octave[i].x2=0;
+ octave[i].tuning=xml->getparreal("cents",octave[i].tuning);
+ octave[i].x1=xml->getpar127("numerator",octave[i].x1);
+ octave[i].x2=xml->getpar127("denominator",octave[i].x2);
+
+ if (octave[i].x2!=0) octave[i].type=2;
+ else octave[i].type=1;
+
+ xml->exitbranch();
+ };
+ xml->exitbranch();
+ };
+
+ if (xml->enterbranch("KEYBOARD_MAPPING")) {
+ Pmapsize=xml->getpar127("map_size",Pmapsize);
+ Pmappingenabled=xml->getpar127("mapping_enabled",Pmappingenabled);
+ for (int i=0;i<Pmapsize;i++) {
+ if (xml->enterbranch("KEYMAP",i)==0) continue;
+ Pmapping[i]=xml->getpar127("degree",Pmapping[i]);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
};
-int Microtonal::saveXML(char *filename){
+int Microtonal::saveXML(char *filename)
+{
XMLwrapper *xml=new XMLwrapper();
xml->beginbranch("MICROTONAL");
@@ -496,17 +525,18 @@ int Microtonal::saveXML(char *filename){
return(result);
};
-int Microtonal::loadXML(char *filename){
+int Microtonal::loadXML(char *filename)
+{
XMLwrapper *xml=new XMLwrapper();
if (xml->loadXMLfile(filename)<0) {
- delete(xml);
- return(-1);
+ delete(xml);
+ return(-1);
};
-
+
if (xml->enterbranch("MICROTONAL")==0) return(-10);
- getfromXML(xml);
+ getfromXML(xml);
xml->exitbranch();
-
+
delete(xml);
return(0);
};
diff --git a/src/Misc/Microtonal.h b/src/Misc/Microtonal.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Microtonal.h - Tuning settings and microtonal capabilities
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -33,89 +33,90 @@
/**Tuning settings and microtonal capabilities*/
-class Microtonal{
- public:
- /**Constructor*/
- Microtonal();
- /**Destructor*/
- ~Microtonal();
- void defaults();
- /**Calculates the frequency for a given note
- */
- REALTYPE getnotefreq(int note,int keyshift);
-
-
- //Parameters
- /**if the keys are inversed (the pitch is lower to keys from the right direction) \todo figure out why this is not a bool*/
- unsigned char Pinvertupdown;
-
- /**the central key of the inversion*/
- unsigned char Pinvertupdowncenter;
-
- /**0 for 12 key temperate scale, 1 for microtonal
- * \todo find one good reason why this is not a bool*/
- unsigned char Penabled;
-
- /**the note of "A" key*/
- unsigned char PAnote;
-
- /**the frequency of the "A" note*/
- REALTYPE PAfreq;
-
- /**if the scale is "tuned" to a note, you can tune to other note*/
- unsigned char Pscaleshift;
-
- //first and last key (to retune)
- unsigned char Pfirstkey;
- unsigned char Plastkey;
-
- //The middle note where scale degree 0 is mapped to
- unsigned char Pmiddlenote;
-
- //Map size
- unsigned char Pmapsize;
-
- //Mapping ON/OFF
- unsigned char Pmappingenabled;
- //Mapping (keys)
- short int Pmapping[128];
-
- unsigned char Pglobalfinedetune;
-
- // Functions
- /** Return the current octave size
- * \todo why is this not an int?*/
- unsigned char getoctavesize();
- void tuningtoline(int n,char *line,int maxn);
- int loadscl(const char *filename);//load the tunnings from a .scl file
- int loadkbm(const char *filename);//load the mapping from .kbm file
- int texttotunings(const char *text);
- void texttomapping(const char *text);
- unsigned char *Pname;
- unsigned char *Pcomment;
-
- void add2XML(XMLwrapper *xml);
- void getfromXML(XMLwrapper *xml);
- int saveXML(char *filename);
- int loadXML(char *filename);
-
- private:
- int linetotunings(unsigned int nline,const char *line);
- int loadline(FILE *file,char *line);//loads a line from the text file, while ignoring the lines beggining with "!"
- unsigned char octavesize;
- struct {
- unsigned char type;//1 for cents or 2 for division
-
- // the real tuning (eg. +1.05946 for one halftone)
- // or 2.0 for one octave
- REALTYPE tuning;
-
- //the real tunning is x1/x2
- unsigned int x1,x2;
-
- } octave[MAX_OCTAVE_SIZE],tmpoctave[MAX_OCTAVE_SIZE];
-
-};
-
+class Microtonal
+{
+public:
+ /**Constructor*/
+ Microtonal();
+ /**Destructor*/
+ ~Microtonal();
+ void defaults();
+ /**Calculates the frequency for a given note
+ */
+ REALTYPE getnotefreq(int note,int keyshift);
+
+
+ //Parameters
+ /**if the keys are inversed (the pitch is lower to keys from the right direction) \todo figure out why this is not a bool*/
+ unsigned char Pinvertupdown;
+
+ /**the central key of the inversion*/
+ unsigned char Pinvertupdowncenter;
+
+ /**0 for 12 key temperate scale, 1 for microtonal
+ * \todo find one good reason why this is not a bool*/
+ unsigned char Penabled;
+
+ /**the note of "A" key*/
+ unsigned char PAnote;
+
+ /**the frequency of the "A" note*/
+ REALTYPE PAfreq;
+
+ /**if the scale is "tuned" to a note, you can tune to other note*/
+ unsigned char Pscaleshift;
+
+ //first and last key (to retune)
+ unsigned char Pfirstkey;
+ unsigned char Plastkey;
+
+ //The middle note where scale degree 0 is mapped to
+ unsigned char Pmiddlenote;
+
+ //Map size
+ unsigned char Pmapsize;
+
+ //Mapping ON/OFF
+ unsigned char Pmappingenabled;
+ //Mapping (keys)
+ short int Pmapping[128];
+
+ unsigned char Pglobalfinedetune;
+
+ // Functions
+ /** Return the current octave size
+ * \todo why is this not an int?*/
+ unsigned char getoctavesize();
+ void tuningtoline(int n,char *line,int maxn);
+ int loadscl(const char *filename);//load the tunnings from a .scl file
+ int loadkbm(const char *filename);//load the mapping from .kbm file
+ int texttotunings(const char *text);
+ void texttomapping(const char *text);
+ unsigned char *Pname;
+ unsigned char *Pcomment;
+
+ void add2XML(XMLwrapper *xml);
+ void getfromXML(XMLwrapper *xml);
+ int saveXML(char *filename);
+ int loadXML(char *filename);
+
+private:
+ int linetotunings(unsigned int nline,const char *line);
+ int loadline(FILE *file,char *line);//loads a line from the text file, while ignoring the lines beggining with "!"
+ unsigned char octavesize;
+ struct {
+ unsigned char type;//1 for cents or 2 for division
+
+ // the real tuning (eg. +1.05946 for one halftone)
+ // or 2.0 for one octave
+ REALTYPE tuning;
+
+ //the real tunning is x1/x2
+ unsigned int x1,x2;
+
+ } octave[MAX_OCTAVE_SIZE],tmpoctave[MAX_OCTAVE_SIZE];
+
+};
+
#endif
diff --git a/src/Misc/Part.C b/src/Misc/Part.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Part.C - Part implementation
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,18 +26,21 @@
#include <stdio.h>
#include <string.h>
-Part::Part(Microtonal *microtonal_,FFTwrapper *fft_, pthread_mutex_t *mutex_){
- microtonal=microtonal_;
+Part::Part(Microtonal *microtonal_,FFTwrapper *fft_, pthread_mutex_t *mutex_)
+{
+ microtonal=microtonal_;
fft=fft_;
mutex=mutex_;
partoutl=new REALTYPE [SOUND_BUFFER_SIZE];
partoutr=new REALTYPE [SOUND_BUFFER_SIZE];
tmpoutl=new REALTYPE [SOUND_BUFFER_SIZE];
tmpoutr=new REALTYPE [SOUND_BUFFER_SIZE];
-
- for (int n=0;n<NUM_KIT_ITEMS;n++){
- kit[n].Pname=new unsigned char [PART_MAX_NAME_LEN];
- kit[n].adpars=NULL;kit[n].subpars=NULL;kit[n].padpars=NULL;
+
+ for (int n=0;n<NUM_KIT_ITEMS;n++) {
+ kit[n].Pname=new unsigned char [PART_MAX_NAME_LEN];
+ kit[n].adpars=NULL;
+ kit[n].subpars=NULL;
+ kit[n].padpars=NULL;
};
kit[0].adpars=new ADnoteParameters(fft);
@@ -47,44 +50,45 @@ Part::Part(Microtonal *microtonal_,FFTwrapper *fft_, pthread_mutex_t *mutex_){
// SUBPartParameters=kit[0].subpars;
//Part's Insertion Effects init
- for (int nefx=0;nefx<NUM_PART_EFX;nefx++)
- partefx[nefx]=new EffectMgr(1,mutex);
+ for (int nefx=0;nefx<NUM_PART_EFX;nefx++)
+ partefx[nefx]=new EffectMgr(1,mutex);
for (int n=0;n<NUM_PART_EFX+1;n++) {
- partfxinputl[n]=new REALTYPE [SOUND_BUFFER_SIZE];
- partfxinputr[n]=new REALTYPE [SOUND_BUFFER_SIZE];
- Pefxbypass[n]=false;
+ partfxinputl[n]=new REALTYPE [SOUND_BUFFER_SIZE];
+ partfxinputr[n]=new REALTYPE [SOUND_BUFFER_SIZE];
+ Pefxbypass[n]=false;
};
killallnotes=0;
oldfreq=-1.0;
int i,j;
- for (i=0;i<POLIPHONY;i++){
- partnote[i].status=KEY_OFF;
- partnote[i].note=-1;
- partnote[i].itemsplaying=0;
- for (j=0;j<NUM_KIT_ITEMS;j++){
- partnote[i].kititem[j].adnote=NULL;
- partnote[i].kititem[j].subnote=NULL;
- partnote[i].kititem[j].padnote=NULL;
- };
- partnote[i].time=0;
+ for (i=0;i<POLIPHONY;i++) {
+ partnote[i].status=KEY_OFF;
+ partnote[i].note=-1;
+ partnote[i].itemsplaying=0;
+ for (j=0;j<NUM_KIT_ITEMS;j++) {
+ partnote[i].kititem[j].adnote=NULL;
+ partnote[i].kititem[j].subnote=NULL;
+ partnote[i].kititem[j].padnote=NULL;
+ };
+ partnote[i].time=0;
};
- cleanup();
+ cleanup();
Pname=new unsigned char [PART_MAX_NAME_LEN];
-
+
oldvolumel=oldvolumer=0.5;
lastnote=-1;
lastpos=0; // lastpos will store previously used NoteOn(...)'s pos.
lastlegatomodevalid=false; // To store previous legatomodevalid value.
-
-
+
+
defaults();
};
-void Part::defaults(){
+void Part::defaults()
+{
Penabled=0;
Pminkey=0;
Pmaxkey=127;
@@ -102,7 +106,8 @@ void Part::defaults(){
ctl.defaults();
};
-void Part::defaultsinstrument(){
+void Part::defaultsinstrument()
+{
ZERO(Pname,PART_MAX_NAME_LEN);
info.Ptype=0;
@@ -112,23 +117,27 @@ void Part::defaultsinstrument(){
Pkitmode=0;
Pdrummode=0;
- for (int n=0;n<NUM_KIT_ITEMS;n++){
- kit[n].Penabled=0;kit[n].Pmuted=0;
- kit[n].Pminkey=0;kit[n].Pmaxkey=127;
- kit[n].Padenabled=0;kit[n].Psubenabled=0;kit[n].Ppadenabled=0;
- ZERO(kit[n].Pname,PART_MAX_NAME_LEN);
- kit[n].Psendtoparteffect=0;
- if (n!=0) setkititemstatus(n,0);
+ for (int n=0;n<NUM_KIT_ITEMS;n++) {
+ kit[n].Penabled=0;
+ kit[n].Pmuted=0;
+ kit[n].Pminkey=0;
+ kit[n].Pmaxkey=127;
+ kit[n].Padenabled=0;
+ kit[n].Psubenabled=0;
+ kit[n].Ppadenabled=0;
+ ZERO(kit[n].Pname,PART_MAX_NAME_LEN);
+ kit[n].Psendtoparteffect=0;
+ if (n!=0) setkititemstatus(n,0);
};
kit[0].Penabled=1;
kit[0].Padenabled=1;
kit[0].adpars->defaults();
kit[0].subpars->defaults();
kit[0].padpars->defaults();
-
+
for (int nefx=0;nefx<NUM_PART_EFX;nefx++) {
- partefx[nefx]->defaults();
- Pefxroute[nefx]=0;//route to next effect
+ partefx[nefx]->defaults();
+ Pefxroute[nefx]=0;//route to next effect
};
};
@@ -138,32 +147,36 @@ void Part::defaultsinstrument(){
/*
* Cleanup the part
*/
-void Part::cleanup(){
+void Part::cleanup()
+{
for (int k=0;k<POLIPHONY;k++) KillNotePos(k);
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- partoutl[i]=denormalkillbuf[i];
- partoutr[i]=denormalkillbuf[i];
- tmpoutl[i]=0.0;
- tmpoutr[i]=0.0;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ partoutl[i]=denormalkillbuf[i];
+ partoutr[i]=denormalkillbuf[i];
+ tmpoutl[i]=0.0;
+ tmpoutr[i]=0.0;
};
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 i=0;i<SOUND_BUFFER_SIZE;i++){
- partfxinputl[n][i]=denormalkillbuf[i];
- partfxinputr[n][i]=denormalkillbuf[i];
- };
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ partfxinputl[n][i]=denormalkillbuf[i];
+ partfxinputr[n][i]=denormalkillbuf[i];
+ };
};
};
-Part::~Part(){
+Part::~Part()
+{
cleanup();
- for (int n=0;n<NUM_KIT_ITEMS;n++){
- if (kit[n].adpars!=NULL) delete (kit[n].adpars);
- if (kit[n].subpars!=NULL) delete (kit[n].subpars);
- if (kit[n].padpars!=NULL) delete (kit[n].padpars);
- kit[n].adpars=NULL;kit[n].subpars=NULL;kit[n].padpars=NULL;
- delete [] kit[n].Pname;
+ for (int n=0;n<NUM_KIT_ITEMS;n++) {
+ if (kit[n].adpars!=NULL) delete (kit[n].adpars);
+ if (kit[n].subpars!=NULL) delete (kit[n].subpars);
+ if (kit[n].padpars!=NULL) delete (kit[n].padpars);
+ kit[n].adpars=NULL;
+ kit[n].subpars=NULL;
+ kit[n].padpars=NULL;
+ delete [] kit[n].Pname;
};
delete [] Pname;
@@ -171,18 +184,19 @@ Part::~Part(){
delete [] partoutr;
delete [] tmpoutl;
delete [] tmpoutr;
- for (int nefx=0;nefx<NUM_PART_EFX;nefx++)
- delete (partefx[nefx]);
+ for (int nefx=0;nefx<NUM_PART_EFX;nefx++)
+ delete (partefx[nefx]);
for (int n=0;n<NUM_PART_EFX+1;n++) {
- delete [] partfxinputl[n];
- delete [] partfxinputr[n];
+ delete [] partfxinputl[n];
+ delete [] partfxinputr[n];
}
};
/*
* Note On Messages
*/
-void Part::NoteOn(unsigned char note,unsigned char velocity,int masterkeyshift){
+void Part::NoteOn(unsigned char note,unsigned char velocity,int masterkeyshift)
+{
int i,pos;
// Legato and MonoMem used vars:
@@ -197,255 +211,256 @@ void Part::NoteOn(unsigned char note,unsigned char velocity,int masterkeyshift){
if ((note<Pminkey)||(note>Pmaxkey)) return;
// MonoMem stuff:
- if (Ppolymode==0){ // If Poly is off
- monomemnotes.push_back(note); // Add note to the list.
- monomem[note].velocity=velocity; // Store this note's velocity.
- monomem[note].mkeyshift=masterkeyshift; /* Store masterkeyshift too,
+ if (Ppolymode==0) { // If Poly is off
+ monomemnotes.push_back(note); // Add note to the list.
+ monomem[note].velocity=velocity; // Store this note's velocity.
+ monomem[note].mkeyshift=masterkeyshift; /* Store masterkeyshift too,
I'm not sure why though... */
- if ((partnote[lastpos].status!=KEY_PLAYING)
- && (partnote[lastpos].status!=KEY_RELASED_AND_SUSTAINED)){
- ismonofirstnote=true; // No other keys are held or sustained.
- }
+ if ((partnote[lastpos].status!=KEY_PLAYING)
+ && (partnote[lastpos].status!=KEY_RELASED_AND_SUSTAINED)) {
+ ismonofirstnote=true; // No other keys are held or sustained.
+ }
} else {
- // Poly mode is On so just make sure the list is empty.
- if (not monomemnotes.empty()) monomemnotes.clear();
+ // Poly mode is On so just make sure the list is empty.
+ if (not monomemnotes.empty()) monomemnotes.clear();
}
lastnote=note;
pos=-1;
- for (i=0;i<POLIPHONY;i++){
- if (partnote[i].status==KEY_OFF){
- pos=i;
- break;
- };
+ 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, "ZynAddSubFX WARNING: Poly and Legato modes are both On, that should not happen ! ... Disabling Legato mode ! - (Part.C::NoteOn(..))\n");
- Plegatomode=0;
- } else {
- // Legato mode is on and applicable.
- legatomodevalid=true;
- if ((not ismonofirstnote)&&(lastlegatomodevalid)){
- // At least one other key is held or sustained, and the
- // previous note was played while in valid legato mode.
- doinglegato=true; // So we'll do a legato note.
- pos=lastpos; // A legato note uses same pos as previous..
- posb=lastposb; // .. same goes for posb.
- } else {
- // Legato mode is valid, but this is only a first note.
- for (i=0;i<POLIPHONY;i++)
- if ((partnote[i].status==KEY_PLAYING) ||
- (partnote[i].status==KEY_RELASED_AND_SUSTAINED))
- RelaseNotePos(i);
-
- // Set posb
- posb=(pos+1)%POLIPHONY;//We really want it (if the following fails)
- for (i=0;i<POLIPHONY;i++){
- if ((partnote[i].status==KEY_OFF) && (pos!=i)){
- posb=i;
- break;
- }
- }
- }
- lastposb=posb;// Keep a trace of used posb
- }
+ if ((Plegatomode!=0) && (Pdrummode==0)) {
+ if (Ppolymode!=0) {
+ fprintf(stderr, "ZynAddSubFX WARNING: Poly and Legato modes are both On, that should not happen ! ... Disabling Legato mode ! - (Part.C::NoteOn(..))\n");
+ Plegatomode=0;
+ } else {
+ // Legato mode is on and applicable.
+ legatomodevalid=true;
+ if ((not ismonofirstnote)&&(lastlegatomodevalid)) {
+ // At least one other key is held or sustained, and the
+ // previous note was played while in valid legato mode.
+ doinglegato=true; // So we'll do a legato note.
+ pos=lastpos; // A legato note uses same pos as previous..
+ posb=lastposb; // .. same goes for posb.
+ } else {
+ // Legato mode is valid, but this is only a first note.
+ for (i=0;i<POLIPHONY;i++)
+ if ((partnote[i].status==KEY_PLAYING) ||
+ (partnote[i].status==KEY_RELASED_AND_SUSTAINED))
+ RelaseNotePos(i);
+
+ // Set posb
+ posb=(pos+1)%POLIPHONY;//We really want it (if the following fails)
+ for (i=0;i<POLIPHONY;i++) {
+ if ((partnote[i].status==KEY_OFF) && (pos!=i)) {
+ posb=i;
+ break;
+ }
+ }
+ }
+ lastposb=posb;// Keep a trace of used posb
+ }
} else { // Legato mode is either off or non-applicable.
- if (Ppolymode==0){//if the mode is 'mono' turn off all other notes
- for (i=0;i<POLIPHONY;i++)
- if (partnote[i].status==KEY_PLAYING) RelaseNotePos(i);
- RelaseSustainedKeys();
- }
+ if (Ppolymode==0) {//if the mode is 'mono' turn off all other notes
+ for (i=0;i<POLIPHONY;i++)
+ if (partnote[i].status==KEY_PLAYING) RelaseNotePos(i);
+ RelaseSustainedKeys();
+ }
}
lastlegatomodevalid=legatomodevalid;
- if (pos==-1){
+ if (pos==-1) {
//test
- fprintf(stderr,"%s","NOTES TOO MANY (> POLIPHONY) - (Part.C::NoteOn(..))\n");
+ fprintf(stderr,"%s","NOTES TOO MANY (> POLIPHONY) - (Part.C::NoteOn(..))\n");
} else {
- //start the note
- partnote[pos].status=KEY_PLAYING;
- partnote[pos].note=note;
- if (legatomodevalid){
- partnote[posb].status=KEY_PLAYING;
- partnote[posb].note=note;
- }
-
- //this computes the velocity sensing of the part
- REALTYPE vel=VelF(velocity/127.0,Pvelsns);
-
- //compute the velocity offset
- vel+=(Pveloffs-64.0)/64.0;
- if (vel<0.0) vel=0.0; else if (vel>1.0) vel=1.0;
-
- //compute the keyshift
- int partkeyshift=(int)Pkeyshift-64;
- int keyshift=masterkeyshift+partkeyshift;
-
- //initialise note frequency
- REALTYPE notebasefreq;
- if (Pdrummode==0){
- notebasefreq=microtonal->getnotefreq(note,keyshift);
- if (notebasefreq<0.0) return;//the key is no mapped
- } else {
- notebasefreq=440.0*pow(2.0,(note-69.0)/12.0);
- };
-
- //Portamento
- if (oldfreq<1.0) oldfreq=notebasefreq;//this is only the first note is played
-
- // For Mono/Legato: Force Portamento Off on first
- // notes. That means it is required that the previous note is
- // still held down or sustained for the Portamento to activate
- // (that's like Legato).
- int portamento=0;
- if ((Ppolymode!=0) || (not ismonofirstnote)){
- // I added a third argument to the
- // ctl.initportamento(...) function to be able
- // to tell it if we're doing a legato note.
- portamento=ctl.initportamento(oldfreq,notebasefreq,doinglegato);
- }
-
- if (portamento!=0) ctl.portamento.noteusing=pos;
- oldfreq=notebasefreq;
-
- lastpos=pos; // Keep a trace of used pos.
-
- if (doinglegato){
- // Do Legato note
- if (Pkitmode==0){ // "normal mode" legato note
- if ((kit[0].Padenabled!=0)
- && (partnote[pos].kititem[0].adnote!=NULL)
- && (partnote[posb].kititem[0].adnote!=NULL)){
- partnote[pos].kititem[0].adnote->ADlegatonote(notebasefreq, vel, portamento, note, true);//'true' is to tell it it's being called from here.
- partnote[posb].kititem[0].adnote->ADlegatonote(notebasefreq, vel, portamento, note, true);
- }
-
- if ((kit[0].Psubenabled!=0)
- && (partnote[pos].kititem[0].subnote!=NULL)
- && (partnote[posb].kititem[0].subnote!=NULL)){
- partnote[pos].kititem[0].subnote->SUBlegatonote(notebasefreq, vel, portamento, note, true);
- partnote[posb].kititem[0].subnote->SUBlegatonote(notebasefreq, vel, portamento, note, true);
- }
-
- if ((kit[0].Ppadenabled!=0)
- && (partnote[pos].kititem[0].padnote!=NULL)
- && (partnote[posb].kititem[0].padnote!=NULL)){
- partnote[pos].kititem[0].padnote->PADlegatonote(notebasefreq, vel, portamento, note, true);
- partnote[posb].kititem[0].padnote->PADlegatonote(notebasefreq, vel, portamento, note, true);
- }
-
- } else { // "kit mode" legato note
- int ci=0;
- for (int item=0;item<NUM_KIT_ITEMS;item++){
- if (kit[item].Pmuted!=0) continue;
- if ((note<kit[item].Pminkey)||(note>kit[item].Pmaxkey)) continue;
-
- if ((lastnotecopy<kit[item].Pminkey)
- ||(lastnotecopy>kit[item].Pmaxkey))
- continue; // We will not perform legato across 2 key regions.
-
- partnote[pos].kititem[ci].sendtoparteffect=( kit[item].Psendtoparteffect<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);
-
- if ((kit[item].Padenabled!=0) && (kit[item].adpars!=NULL)
- && (partnote[pos].kititem[ci].adnote!=NULL)
- && (partnote[posb].kititem[ci].adnote!=NULL)){
- partnote[pos].kititem[ci].adnote->ADlegatonote(notebasefreq,vel,portamento,note,true);
- partnote[posb].kititem[ci].adnote->ADlegatonote(notebasefreq,vel,portamento,note,true);
- }
- if ((kit[item].Psubenabled!=0) && (kit[item].subpars!=NULL)
- && (partnote[pos].kititem[ci].subnote!=NULL)
- && (partnote[posb].kititem[ci].subnote!=NULL)){
- partnote[pos].kititem[ci].subnote->SUBlegatonote(notebasefreq,vel,portamento,note,true);
- partnote[posb].kititem[ci].subnote->SUBlegatonote(notebasefreq,vel,portamento,note,true);
- }
- if ((kit[item].Ppadenabled!=0) && (kit[item].padpars!=NULL)
- && (partnote[pos].kititem[ci].padnote!=NULL)
- && (partnote[posb].kititem[ci].padnote!=NULL)){
- partnote[pos].kititem[ci].padnote->PADlegatonote(notebasefreq,vel,portamento,note,true);
- partnote[posb].kititem[ci].padnote->PADlegatonote(notebasefreq,vel,portamento,note,true);
- }
-
- if ((kit[item].adpars!=NULL)||(kit[item].subpars!=NULL)||(kit[item].padpars!=NULL)) {
- ci++;
- if ( ((kit[item].Padenabled!=0)||(kit[item].Psubenabled!=0)||(kit[item].Ppadenabled!=0)) && (Pkitmode==2) ) break;
- }
- }
- if (ci==0){
- // No legato were performed at all, so pretend nothing happened:
- monomemnotes.pop_back(); // Remove last note from the list.
- lastnote=lastnotecopy; // Set lastnote back to previous value.
- }
- }
- return; // Ok, Legato note done, return.
- }
-
- partnote[pos].itemsplaying=0;
- if (legatomodevalid) partnote[posb].itemsplaying=0;
-
- if (Pkitmode==0){//init the notes for the "normal mode"
- 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);
- if (kit[0].Psubenabled!=0) partnote[pos].kititem[0].subnote=new SUBnote(kit[0].subpars,&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);
- if ((kit[0].Padenabled!=0)||(kit[0].Psubenabled!=0)||(kit[0].Ppadenabled!=0)) partnote[pos].itemsplaying++;
-
- // Spawn another note (but silent) if legatomodevalid==true
- if (legatomodevalid){
- 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.
- if (kit[0].Psubenabled!=0) 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);
- if ((kit[0].Padenabled!=0)||(kit[0].Psubenabled!=0)||(kit[0].Ppadenabled!=0)) partnote[posb].itemsplaying++;
- }
-
- } else {//init the notes for the "kit mode"
- for (int item=0;item<NUM_KIT_ITEMS;item++){
- if (kit[item].Pmuted!=0) continue;
- if ((note<kit[item].Pminkey)||(note>kit[item].Pmaxkey)) continue;
-
- int ci=partnote[pos].itemsplaying;//ci=current item
-
- partnote[pos].kititem[ci].sendtoparteffect=( kit[item].Psendtoparteffect<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[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);
-
- 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);
-
- // 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"
-
- 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.
- 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);
- 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);
-
- if ((kit[item].adpars!=NULL)|| (kit[item].subpars!=NULL))
- partnote[posb].itemsplaying++;
- }
-
- if ((kit[item].adpars!=NULL)|| (kit[item].subpars!=NULL)) {
- partnote[pos].itemsplaying++;
- if ( ((kit[item].Padenabled!=0)||(kit[item].Psubenabled!=0)||(kit[item].Ppadenabled!=0))
- && (Pkitmode==2) ) break;
- };
- };
- };
+ //start the note
+ partnote[pos].status=KEY_PLAYING;
+ partnote[pos].note=note;
+ if (legatomodevalid) {
+ partnote[posb].status=KEY_PLAYING;
+ partnote[posb].note=note;
+ }
+
+ //this computes the velocity sensing of the part
+ REALTYPE vel=VelF(velocity/127.0,Pvelsns);
+
+ //compute the velocity offset
+ vel+=(Pveloffs-64.0)/64.0;
+ if (vel<0.0) vel=0.0;
+ else if (vel>1.0) vel=1.0;
+
+ //compute the keyshift
+ int partkeyshift=(int)Pkeyshift-64;
+ int keyshift=masterkeyshift+partkeyshift;
+
+ //initialise note frequency
+ REALTYPE notebasefreq;
+ if (Pdrummode==0) {
+ notebasefreq=microtonal->getnotefreq(note,keyshift);
+ if (notebasefreq<0.0) return;//the key is no mapped
+ } else {
+ notebasefreq=440.0*pow(2.0,(note-69.0)/12.0);
+ };
+
+ //Portamento
+ if (oldfreq<1.0) oldfreq=notebasefreq;//this is only the first note is played
+
+ // For Mono/Legato: Force Portamento Off on first
+ // notes. That means it is required that the previous note is
+ // still held down or sustained for the Portamento to activate
+ // (that's like Legato).
+ int portamento=0;
+ if ((Ppolymode!=0) || (not ismonofirstnote)) {
+ // I added a third argument to the
+ // ctl.initportamento(...) function to be able
+ // to tell it if we're doing a legato note.
+ portamento=ctl.initportamento(oldfreq,notebasefreq,doinglegato);
+ }
+
+ if (portamento!=0) ctl.portamento.noteusing=pos;
+ oldfreq=notebasefreq;
+
+ lastpos=pos; // Keep a trace of used pos.
+
+ if (doinglegato) {
+ // Do Legato note
+ if (Pkitmode==0) { // "normal mode" legato note
+ if ((kit[0].Padenabled!=0)
+ && (partnote[pos].kititem[0].adnote!=NULL)
+ && (partnote[posb].kititem[0].adnote!=NULL)) {
+ partnote[pos].kititem[0].adnote->ADlegatonote(notebasefreq, vel, portamento, note, true);//'true' is to tell it it's being called from here.
+ partnote[posb].kititem[0].adnote->ADlegatonote(notebasefreq, vel, portamento, note, true);
+ }
+
+ if ((kit[0].Psubenabled!=0)
+ && (partnote[pos].kititem[0].subnote!=NULL)
+ && (partnote[posb].kititem[0].subnote!=NULL)) {
+ partnote[pos].kititem[0].subnote->SUBlegatonote(notebasefreq, vel, portamento, note, true);
+ partnote[posb].kititem[0].subnote->SUBlegatonote(notebasefreq, vel, portamento, note, true);
+ }
+
+ if ((kit[0].Ppadenabled!=0)
+ && (partnote[pos].kititem[0].padnote!=NULL)
+ && (partnote[posb].kititem[0].padnote!=NULL)) {
+ partnote[pos].kititem[0].padnote->PADlegatonote(notebasefreq, vel, portamento, note, true);
+ partnote[posb].kititem[0].padnote->PADlegatonote(notebasefreq, vel, portamento, note, true);
+ }
+
+ } else { // "kit mode" legato note
+ int ci=0;
+ for (int item=0;item<NUM_KIT_ITEMS;item++) {
+ if (kit[item].Pmuted!=0) continue;
+ if ((note<kit[item].Pminkey)||(note>kit[item].Pmaxkey)) continue;
+
+ if ((lastnotecopy<kit[item].Pminkey)
+ ||(lastnotecopy>kit[item].Pmaxkey))
+ continue; // We will not perform legato across 2 key regions.
+
+ partnote[pos].kititem[ci].sendtoparteffect=( kit[item].Psendtoparteffect<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);
+
+ if ((kit[item].Padenabled!=0) && (kit[item].adpars!=NULL)
+ && (partnote[pos].kititem[ci].adnote!=NULL)
+ && (partnote[posb].kititem[ci].adnote!=NULL)) {
+ partnote[pos].kititem[ci].adnote->ADlegatonote(notebasefreq,vel,portamento,note,true);
+ partnote[posb].kititem[ci].adnote->ADlegatonote(notebasefreq,vel,portamento,note,true);
+ }
+ if ((kit[item].Psubenabled!=0) && (kit[item].subpars!=NULL)
+ && (partnote[pos].kititem[ci].subnote!=NULL)
+ && (partnote[posb].kititem[ci].subnote!=NULL)) {
+ partnote[pos].kititem[ci].subnote->SUBlegatonote(notebasefreq,vel,portamento,note,true);
+ partnote[posb].kititem[ci].subnote->SUBlegatonote(notebasefreq,vel,portamento,note,true);
+ }
+ if ((kit[item].Ppadenabled!=0) && (kit[item].padpars!=NULL)
+ && (partnote[pos].kititem[ci].padnote!=NULL)
+ && (partnote[posb].kititem[ci].padnote!=NULL)) {
+ partnote[pos].kititem[ci].padnote->PADlegatonote(notebasefreq,vel,portamento,note,true);
+ partnote[posb].kititem[ci].padnote->PADlegatonote(notebasefreq,vel,portamento,note,true);
+ }
+
+ if ((kit[item].adpars!=NULL)||(kit[item].subpars!=NULL)||(kit[item].padpars!=NULL)) {
+ ci++;
+ if ( ((kit[item].Padenabled!=0)||(kit[item].Psubenabled!=0)||(kit[item].Ppadenabled!=0)) && (Pkitmode==2) ) break;
+ }
+ }
+ if (ci==0) {
+ // No legato were performed at all, so pretend nothing happened:
+ monomemnotes.pop_back(); // Remove last note from the list.
+ lastnote=lastnotecopy; // Set lastnote back to previous value.
+ }
+ }
+ return; // Ok, Legato note done, return.
+ }
+
+ partnote[pos].itemsplaying=0;
+ if (legatomodevalid) partnote[posb].itemsplaying=0;
+
+ if (Pkitmode==0) {//init the notes for the "normal mode"
+ 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);
+ if (kit[0].Psubenabled!=0) partnote[pos].kititem[0].subnote=new SUBnote(kit[0].subpars,&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);
+ if ((kit[0].Padenabled!=0)||(kit[0].Psubenabled!=0)||(kit[0].Ppadenabled!=0)) partnote[pos].itemsplaying++;
+
+ // Spawn another note (but silent) if legatomodevalid==true
+ if (legatomodevalid) {
+ 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.
+ if (kit[0].Psubenabled!=0) 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);
+ if ((kit[0].Padenabled!=0)||(kit[0].Psubenabled!=0)||(kit[0].Ppadenabled!=0)) partnote[posb].itemsplaying++;
+ }
+
+ } else {//init the notes for the "kit mode"
+ for (int item=0;item<NUM_KIT_ITEMS;item++) {
+ if (kit[item].Pmuted!=0) continue;
+ if ((note<kit[item].Pminkey)||(note>kit[item].Pmaxkey)) continue;
+
+ int ci=partnote[pos].itemsplaying;//ci=current item
+
+ partnote[pos].kititem[ci].sendtoparteffect=( kit[item].Psendtoparteffect<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[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);
+
+ 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);
+
+ // 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"
+
+ 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.
+ 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);
+ 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);
+
+ if ((kit[item].adpars!=NULL)|| (kit[item].subpars!=NULL))
+ partnote[posb].itemsplaying++;
+ }
+
+ if ((kit[item].adpars!=NULL)|| (kit[item].subpars!=NULL)) {
+ partnote[pos].itemsplaying++;
+ if ( ((kit[item].Padenabled!=0)||(kit[item].Psubenabled!=0)||(kit[item].Ppadenabled!=0))
+ && (Pkitmode==2) ) break;
+ };
+ };
+ };
};
-
+
//this only relase the keys if there is maximum number of keys allowed
setkeylimit(Pkeylimit);
};
@@ -453,185 +468,205 @@ void Part::NoteOn(unsigned char note,unsigned char velocity,int masterkeyshift){
/*
* Note Off Messages
*/
-void Part::NoteOff(unsigned char note){//relase the key
- int i;
+void Part::NoteOff(unsigned char note) //relase the key
+{
+ int i;
// This note is released, so we remove it from the list.
if (not monomemnotes.empty()) monomemnotes.remove(note);
- for (i=POLIPHONY-1;i>=0;i--){ //first note in, is first out if there are same note multiple times
- if ((partnote[i].status==KEY_PLAYING)&&(partnote[i].note==note)) {
- if (ctl.sustain.sustain==0){ //the sustain pedal is not pushed
- if ((Ppolymode==0) && (not monomemnotes.empty())){
- MonoMemRenote(); // To play most recent still held note.
- } else {
- RelaseNotePos(i);
- /// break;
- }
- } else {//the sustain pedal is pushed
- partnote[i].status=KEY_RELASED_AND_SUSTAINED;
- }
- }
+ for (i=POLIPHONY-1;i>=0;i--) { //first note in, is first out if there are same note multiple times
+ if ((partnote[i].status==KEY_PLAYING)&&(partnote[i].note==note)) {
+ if (ctl.sustain.sustain==0) { //the sustain pedal is not pushed
+ if ((Ppolymode==0) && (not monomemnotes.empty())) {
+ MonoMemRenote(); // To play most recent still held note.
+ } else {
+ RelaseNotePos(i);
+ /// break;
+ }
+ } else {//the sustain pedal is pushed
+ partnote[i].status=KEY_RELASED_AND_SUSTAINED;
+ }
+ }
}
};
/*
* Controllers
*/
-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) 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
-
- for (int item=0;item<NUM_KIT_ITEMS;item++){
- if (kit[item].adpars==NULL) continue;
- kit[item].adpars->GlobalPar.Reson->
- sendcontroller(C_resonance_center,1.0);
-
- kit[item].adpars->GlobalPar.Reson->
- sendcontroller(C_resonance_bandwidth,1.0);
- };
- //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;
+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) 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
+
+ for (int item=0;item<NUM_KIT_ITEMS;item++) {
+ if (kit[item].adpars==NULL) continue;
+ kit[item].adpars->GlobalPar.Reson->
+ sendcontroller(C_resonance_center,1.0);
+
+ kit[item].adpars->GlobalPar.Reson->
+ sendcontroller(C_resonance_bandwidth,1.0);
+ };
+ //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;
};
};
/*
* Relase the sustained keys
*/
-void Part::RelaseSustainedKeys(){
+void Part::RelaseSustainedKeys()
+{
// Let's call MonoMemRenote() on some conditions:
if ((Ppolymode==0) && (not monomemnotes.empty()))
- if (monomemnotes.back()!=lastnote) // Sustain controller manipulation would cause repeated same note respawn without this check.
- MonoMemRenote(); // To play most recent still held note.
+ if (monomemnotes.back()!=lastnote) // Sustain controller manipulation would cause repeated same note respawn without this check.
+ MonoMemRenote(); // To play most recent still held note.
for (int i=0;i<POLIPHONY;i++)
- if (partnote[i].status==KEY_RELASED_AND_SUSTAINED) RelaseNotePos(i);
+ if (partnote[i].status==KEY_RELASED_AND_SUSTAINED) RelaseNotePos(i);
};
/*
* Relase all keys
*/
-void Part::RelaseAllKeys(){
- for (int i=0;i<POLIPHONY;i++){
- if ((partnote[i].status!=KEY_RELASED)&&
- (partnote[i].status!=KEY_OFF)) //thanks to Frank Neumann
- RelaseNotePos(i);
+void Part::RelaseAllKeys()
+{
+ for (int i=0;i<POLIPHONY;i++) {
+ if ((partnote[i].status!=KEY_RELASED)&&
+ (partnote[i].status!=KEY_OFF)) //thanks to Frank Neumann
+ RelaseNotePos(i);
};
};
// Call NoteOn(...) with the most recent still held key as new note
// (Made for Mono/Legato).
-void Part::MonoMemRenote(){
- unsigned char mmrtempnote=monomemnotes.back(); // Last list element.
- monomemnotes.pop_back();// We remove it, will be added again in NoteOn(...).
- if (Pnoteon==0){
- RelaseNotePos(lastpos);
- } else {
- NoteOn(mmrtempnote, monomem[mmrtempnote].velocity,
- monomem[mmrtempnote].mkeyshift);
- }
+void Part::MonoMemRenote()
+{
+ unsigned char mmrtempnote=monomemnotes.back(); // Last list element.
+ monomemnotes.pop_back();// We remove it, will be added again in NoteOn(...).
+ if (Pnoteon==0) {
+ RelaseNotePos(lastpos);
+ } else {
+ NoteOn(mmrtempnote, monomem[mmrtempnote].velocity,
+ monomem[mmrtempnote].mkeyshift);
+ }
}
/*
* Release note at position
*/
-void Part::RelaseNotePos(int pos){
+void Part::RelaseNotePos(int pos)
+{
- for (int j=0;j<NUM_KIT_ITEMS;j++){
+ for (int j=0;j<NUM_KIT_ITEMS;j++) {
- if (partnote[pos].kititem[j].adnote!=NULL)
- if (partnote[pos].kititem[j].adnote)
- partnote[pos].kititem[j].adnote->relasekey();
+ if (partnote[pos].kititem[j].adnote!=NULL)
+ if (partnote[pos].kititem[j].adnote)
+ partnote[pos].kititem[j].adnote->relasekey();
- if (partnote[pos].kititem[j].subnote!=NULL)
- if (partnote[pos].kititem[j].subnote!=NULL)
- partnote[pos].kititem[j].subnote->relasekey();
+ if (partnote[pos].kititem[j].subnote!=NULL)
+ if (partnote[pos].kititem[j].subnote!=NULL)
+ partnote[pos].kititem[j].subnote->relasekey();
- if (partnote[pos].kititem[j].padnote!=NULL)
- if (partnote[pos].kititem[j].padnote)
- partnote[pos].kititem[j].padnote->relasekey();
+ if (partnote[pos].kititem[j].padnote!=NULL)
+ if (partnote[pos].kititem[j].padnote)
+ partnote[pos].kititem[j].padnote->relasekey();
};
partnote[pos].status=KEY_RELASED;
-};
-
-
+};
+
+
/*
* Kill note at position
*/
-void Part::KillNotePos(int pos){
+void Part::KillNotePos(int pos)
+{
partnote[pos].status=KEY_OFF;
partnote[pos].note=-1;
partnote[pos].time=0;
partnote[pos].itemsplaying=0;
- for (int j=0;j<NUM_KIT_ITEMS;j++){
- if (partnote[pos].kititem[j].adnote!=NULL) {
- delete(partnote[pos].kititem[j].adnote);
- partnote[pos].kititem[j].adnote=NULL;
- };
- if (partnote[pos].kititem[j].subnote!=NULL) {
- delete(partnote[pos].kititem[j].subnote);
- partnote[pos].kititem[j].subnote=NULL;
- };
- if (partnote[pos].kititem[j].padnote!=NULL) {
- delete(partnote[pos].kititem[j].padnote);
- partnote[pos].kititem[j].padnote=NULL;
- };
+ for (int j=0;j<NUM_KIT_ITEMS;j++) {
+ if (partnote[pos].kititem[j].adnote!=NULL) {
+ delete(partnote[pos].kititem[j].adnote);
+ partnote[pos].kititem[j].adnote=NULL;
+ };
+ if (partnote[pos].kititem[j].subnote!=NULL) {
+ delete(partnote[pos].kititem[j].subnote);
+ partnote[pos].kititem[j].subnote=NULL;
+ };
+ if (partnote[pos].kititem[j].padnote!=NULL) {
+ delete(partnote[pos].kititem[j].padnote);
+ partnote[pos].kititem[j].padnote=NULL;
+ };
};
if (pos==ctl.portamento.noteusing) {
- ctl.portamento.noteusing=-1;
- ctl.portamento.used=0;
+ ctl.portamento.noteusing=-1;
+ ctl.portamento.used=0;
};
};
@@ -639,29 +674,30 @@ void Part::KillNotePos(int pos){
/*
* Set Part's key limit
*/
-void Part::setkeylimit(unsigned char Pkeylimit){
+void Part::setkeylimit(unsigned char Pkeylimit)
+{
this->Pkeylimit=Pkeylimit;
int keylimit=Pkeylimit;
if (keylimit==0) keylimit=POLIPHONY-5;
//release old keys if the number of notes>keylimit
- if (Ppolymode!=0){
- int notecount=0;
- for (int i=0;i<POLIPHONY;i++){
- if ((partnote[i].status==KEY_PLAYING)||(partnote[i].status==KEY_RELASED_AND_SUSTAINED))
- notecount++;
- };
- int oldestnotepos=-1,maxtime=0;
- if (notecount>keylimit){//find out the oldest note
- for (int i=0;i<POLIPHONY;i++){
- if ( ((partnote[i].status==KEY_PLAYING)||(partnote[i].status==KEY_RELASED_AND_SUSTAINED))
- && (partnote[i].time>maxtime)){
- maxtime=partnote[i].time;
- oldestnotepos=i;
- };
- };
- };
- if (oldestnotepos!=-1) RelaseNotePos(oldestnotepos);
+ if (Ppolymode!=0) {
+ int notecount=0;
+ for (int i=0;i<POLIPHONY;i++) {
+ if ((partnote[i].status==KEY_PLAYING)||(partnote[i].status==KEY_RELASED_AND_SUSTAINED))
+ notecount++;
+ };
+ int oldestnotepos=-1,maxtime=0;
+ if (notecount>keylimit) {//find out the oldest note
+ for (int i=0;i<POLIPHONY;i++) {
+ if ( ((partnote[i].status==KEY_PLAYING)||(partnote[i].status==KEY_RELASED_AND_SUSTAINED))
+ && (partnote[i].time>maxtime)) {
+ maxtime=partnote[i].time;
+ oldestnotepos=i;
+ };
+ };
+ };
+ if (oldestnotepos!=-1) RelaseNotePos(oldestnotepos);
};
};
@@ -669,124 +705,135 @@ void Part::setkeylimit(unsigned char Pkeylimit){
/*
* Prepare all notes to be turned off
*/
-void Part::AllNotesOff(){
- killallnotes=1;
+void Part::AllNotesOff()
+{
+ killallnotes=1;
};
/*
* Compute Part samples and store them in the partoutl[] and partoutr[]
*/
-void Part::ComputePartSmps(){
+void Part::ComputePartSmps()
+{
int i,k;
int noteplay;//0 if there is nothing activated
- for (int nefx=0;nefx<NUM_PART_EFX+1;nefx++){
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- partfxinputl[nefx][i]=0.0;
- partfxinputr[nefx][i]=0.0;
- };
+ for (int nefx=0;nefx<NUM_PART_EFX+1;nefx++) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ partfxinputl[nefx][i]=0.0;
+ partfxinputr[nefx][i]=0.0;
+ };
};
-
- for (k=0;k<POLIPHONY;k++){
- if (partnote[k].status==KEY_OFF) continue;
- noteplay=0;
- partnote[k].time++;
- //get the sampledata of the note and kill it if it's finished
-
- for (int item=0;item<partnote[k].itemsplaying;item++){
-
- int sendcurrenttofx=partnote[k].kititem[item].sendtoparteffect;
-
- ADnote *adnote=partnote[k].kititem[item].adnote;
- SUBnote *subnote=partnote[k].kititem[item].subnote;
- PADnote *padnote=partnote[k].kititem[item].padnote;
- //get from the ADnote
+
+ for (k=0;k<POLIPHONY;k++) {
+ if (partnote[k].status==KEY_OFF) continue;
+ noteplay=0;
+ partnote[k].time++;
+ //get the sampledata of the note and kill it if it's finished
+
+ for (int item=0;item<partnote[k].itemsplaying;item++) {
+
+ int sendcurrenttofx=partnote[k].kititem[item].sendtoparteffect;
+
+ ADnote *adnote=partnote[k].kititem[item].adnote;
+ SUBnote *subnote=partnote[k].kititem[item].subnote;
+ PADnote *padnote=partnote[k].kititem[item].padnote;
+ //get from the ADnote
if (adnote!=NULL) {
- noteplay++;
- if (adnote->ready!=0) adnote->noteout(&tmpoutl[0],&tmpoutr[0]);
- else for (i=0;i<SOUND_BUFFER_SIZE;i++){tmpoutl[i]=0.0;tmpoutr[i]=0.0;};
- if (adnote->finished()!=0){
- delete (adnote);
- partnote[k].kititem[item].adnote=NULL;
- };
- for (i=0;i<SOUND_BUFFER_SIZE;i++){//add the ADnote to part(mix)
- partfxinputl[sendcurrenttofx][i]+=tmpoutl[i];
- partfxinputr[sendcurrenttofx][i]+=tmpoutr[i];
- };
- };
- //get from the SUBnote
- if (subnote!=NULL) {
- noteplay++;
- if (subnote->ready!=0) subnote->noteout(&tmpoutl[0],&tmpoutr[0]);
- else for (i=0;i<SOUND_BUFFER_SIZE;i++){tmpoutl[i]=0.0;tmpoutr[i]=0.0;};
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++){//add the SUBnote to part(mix)
- partfxinputl[sendcurrenttofx][i]+=tmpoutl[i];
- partfxinputr[sendcurrenttofx][i]+=tmpoutr[i];
- };
- if (subnote->finished()!=0){
- delete (subnote);
- partnote[k].kititem[item].subnote=NULL;
- };
- };
- //get from the PADnote
+ noteplay++;
+ if (adnote->ready!=0) adnote->noteout(&tmpoutl[0],&tmpoutr[0]);
+ else for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpoutl[i]=0.0;
+ tmpoutr[i]=0.0;
+ };
+ if (adnote->finished()!=0) {
+ delete (adnote);
+ partnote[k].kititem[item].adnote=NULL;
+ };
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {//add the ADnote to part(mix)
+ partfxinputl[sendcurrenttofx][i]+=tmpoutl[i];
+ partfxinputr[sendcurrenttofx][i]+=tmpoutr[i];
+ };
+ };
+ //get from the SUBnote
+ if (subnote!=NULL) {
+ noteplay++;
+ if (subnote->ready!=0) subnote->noteout(&tmpoutl[0],&tmpoutr[0]);
+ else for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpoutl[i]=0.0;
+ tmpoutr[i]=0.0;
+ };
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {//add the SUBnote to part(mix)
+ partfxinputl[sendcurrenttofx][i]+=tmpoutl[i];
+ partfxinputr[sendcurrenttofx][i]+=tmpoutr[i];
+ };
+ if (subnote->finished()!=0) {
+ delete (subnote);
+ partnote[k].kititem[item].subnote=NULL;
+ };
+ };
+ //get from the PADnote
if (padnote!=NULL) {
- noteplay++;
- if (padnote->ready!=0) padnote->noteout(&tmpoutl[0],&tmpoutr[0]);
- else for (i=0;i<SOUND_BUFFER_SIZE;i++){tmpoutl[i]=0.0;tmpoutr[i]=0.0;};
- if (padnote->finished()!=0){
- delete (padnote);
- partnote[k].kititem[item].padnote=NULL;
- };
- for (i=0;i<SOUND_BUFFER_SIZE;i++){//add the PADnote to part(mix)
- partfxinputl[sendcurrenttofx][i]+=tmpoutl[i];
- partfxinputr[sendcurrenttofx][i]+=tmpoutr[i];
- };
- };
-
- };
- //Kill note if there is no synth on that note
- if (noteplay==0) KillNotePos(k);
+ noteplay++;
+ if (padnote->ready!=0) padnote->noteout(&tmpoutl[0],&tmpoutr[0]);
+ else for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpoutl[i]=0.0;
+ tmpoutr[i]=0.0;
+ };
+ if (padnote->finished()!=0) {
+ delete (padnote);
+ partnote[k].kititem[item].padnote=NULL;
+ };
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {//add the PADnote to part(mix)
+ partfxinputl[sendcurrenttofx][i]+=tmpoutl[i];
+ partfxinputr[sendcurrenttofx][i]+=tmpoutr[i];
+ };
+ };
+
+ };
+ //Kill note if there is no synth on that note
+ if (noteplay==0) KillNotePos(k);
};
//Apply part's effects and mix them
for (int nefx=0;nefx<NUM_PART_EFX;nefx++) {
- if (!Pefxbypass[nefx]) {
- partefx[nefx]->out(partfxinputl[nefx],partfxinputr[nefx]);
- if (Pefxroute[nefx]==2){
- for (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 (i=0;i<SOUND_BUFFER_SIZE;i++){
- partfxinputl[routeto][i]+=partfxinputl[nefx][i];
- partfxinputr[routeto][i]+=partfxinputr[nefx][i];
- };
-
+ if (!Pefxbypass[nefx]) {
+ partefx[nefx]->out(partfxinputl[nefx],partfxinputr[nefx]);
+ if (Pefxroute[nefx]==2) {
+ for (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 (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ partfxinputl[routeto][i]+=partfxinputl[nefx][i];
+ partfxinputr[routeto][i]+=partfxinputr[nefx][i];
+ };
+
};
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- partoutl[i]=partfxinputl[NUM_PART_EFX][i];
- partoutr[i]=partfxinputr[NUM_PART_EFX][i];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ partoutl[i]=partfxinputl[NUM_PART_EFX][i];
+ partoutr[i]=partfxinputr[NUM_PART_EFX][i];
};
//Kill All Notes if killallnotes!=0
if (killallnotes!=0) {
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- REALTYPE tmp=(SOUND_BUFFER_SIZE-i)/(REALTYPE) SOUND_BUFFER_SIZE;
- partoutl[i]*=tmp;
- partoutr[i]*=tmp;
- tmpoutl[i]=0.0;
- tmpoutr[i]=0.0;
- };
- for (int k=0;k<POLIPHONY;k++) KillNotePos(k);
- killallnotes=0;
- for (int nefx=0;nefx<NUM_PART_EFX;nefx++) {
- partefx[nefx]->cleanup();
- };
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE tmp=(SOUND_BUFFER_SIZE-i)/(REALTYPE) SOUND_BUFFER_SIZE;
+ partoutl[i]*=tmp;
+ partoutr[i]*=tmp;
+ tmpoutl[i]=0.0;
+ tmpoutr[i]=0.0;
+ };
+ for (int k=0;k<POLIPHONY;k++) KillNotePos(k);
+ killallnotes=0;
+ for (int nefx=0;nefx<NUM_PART_EFX;nefx++) {
+ partefx[nefx]->cleanup();
+ };
};
ctl.updateportamento();
};
@@ -794,114 +841,122 @@ void Part::ComputePartSmps(){
/*
* Parameter control
*/
-void Part::setPvolume(char Pvolume_){
+void Part::setPvolume(char Pvolume_)
+{
Pvolume=Pvolume_;
volume=dB2rap((Pvolume-96.0)/96.0*40.0)*ctl.expression.relvolume;
};
-void Part::setPpanning(char Ppanning_){
+void Part::setPpanning(char Ppanning_)
+{
Ppanning=Ppanning_;
panning=Ppanning/127.0+ctl.panning.pan;
- if (panning<0.0) panning=0.0;else if (panning>1.0) panning=1.0;
+ if (panning<0.0) panning=0.0;
+ else if (panning>1.0) panning=1.0;
};
/*
* Enable or disable a kit item
*/
-void Part::setkititemstatus(int kititem,int Penabled_){
+void Part::setkititemstatus(int kititem,int Penabled_)
+{
if ((kititem==0)&&(kititem>=NUM_KIT_ITEMS)) return;//nonexistent kit item and the first kit item is always enabled
kit[kititem].Penabled=Penabled_;
-
+
bool resetallnotes=false;
- if (Penabled_==0){
- if (kit[kititem].adpars!=NULL) delete (kit[kititem].adpars);
- if (kit[kititem].subpars!=NULL) delete (kit[kititem].subpars);
- if (kit[kititem].padpars!=NULL) {
- delete (kit[kititem].padpars);
- resetallnotes=true;
- };
- kit[kititem].adpars=NULL;kit[kititem].subpars=NULL;kit[kititem].padpars=NULL;
- kit[kititem].Pname[0]='\0';
+ if (Penabled_==0) {
+ if (kit[kititem].adpars!=NULL) delete (kit[kititem].adpars);
+ if (kit[kititem].subpars!=NULL) delete (kit[kititem].subpars);
+ if (kit[kititem].padpars!=NULL) {
+ delete (kit[kititem].padpars);
+ resetallnotes=true;
+ };
+ kit[kititem].adpars=NULL;
+ kit[kititem].subpars=NULL;
+ kit[kititem].padpars=NULL;
+ kit[kititem].Pname[0]='\0';
} else {
- if (kit[kititem].adpars==NULL) kit[kititem].adpars=new ADnoteParameters(fft);
- if (kit[kititem].subpars==NULL) kit[kititem].subpars=new SUBnoteParameters();
- if (kit[kititem].padpars==NULL) kit[kititem].padpars=new PADnoteParameters(fft,mutex);
+ if (kit[kititem].adpars==NULL) kit[kititem].adpars=new ADnoteParameters(fft);
+ if (kit[kititem].subpars==NULL) kit[kititem].subpars=new SUBnoteParameters();
+ if (kit[kititem].padpars==NULL) kit[kititem].padpars=new PADnoteParameters(fft,mutex);
};
-
+
if (resetallnotes) for (int k=0;k<POLIPHONY;k++) KillNotePos(k);
};
-void Part::add2XMLinstrument(XMLwrapper *xml){
+void Part::add2XMLinstrument(XMLwrapper *xml)
+{
xml->beginbranch("INFO");
- xml->addparstr("name",(char *)Pname);
- xml->addparstr("author",(char *)info.Pauthor);
- xml->addparstr("comments",(char *)info.Pcomments);
- xml->addpar("type",info.Ptype);
+ xml->addparstr("name",(char *)Pname);
+ xml->addparstr("author",(char *)info.Pauthor);
+ xml->addparstr("comments",(char *)info.Pcomments);
+ xml->addpar("type",info.Ptype);
xml->endbranch();
-
-
+
+
xml->beginbranch("INSTRUMENT_KIT");
- xml->addpar("kit_mode",Pkitmode);
- xml->addparbool("drum_mode",Pdrummode);
-
- for (int i=0;i<NUM_KIT_ITEMS;i++){
- xml->beginbranch("INSTRUMENT_KIT_ITEM",i);
- xml->addparbool("enabled",kit[i].Penabled);
- if (kit[i].Penabled!=0) {
- xml->addparstr("name",(char *)kit[i].Pname);
-
- xml->addparbool("muted",kit[i].Pmuted);
- xml->addpar("min_key",kit[i].Pminkey);
- xml->addpar("max_key",kit[i].Pmaxkey);
-
- xml->addpar("send_to_instrument_effect",kit[i].Psendtoparteffect);
-
- xml->addparbool("add_enabled",kit[i].Padenabled);
- if ((kit[i].Padenabled!=0)&&(kit[i].adpars!=NULL)){
- xml->beginbranch("ADD_SYNTH_PARAMETERS");
- kit[i].adpars->add2XML(xml);
- xml->endbranch();
- };
-
- xml->addparbool("sub_enabled",kit[i].Psubenabled);
- if ((kit[i].Psubenabled!=0)&&(kit[i].subpars!=NULL)){
- xml->beginbranch("SUB_SYNTH_PARAMETERS");
- kit[i].subpars->add2XML(xml);
- xml->endbranch();
- };
-
- xml->addparbool("pad_enabled",kit[i].Ppadenabled);
- if ((kit[i].Ppadenabled!=0)&&(kit[i].padpars!=NULL)){
- xml->beginbranch("PAD_SYNTH_PARAMETERS");
- kit[i].padpars->add2XML(xml);
- xml->endbranch();
- };
-
- };
- xml->endbranch();
- };
+ xml->addpar("kit_mode",Pkitmode);
+ xml->addparbool("drum_mode",Pdrummode);
+
+ for (int i=0;i<NUM_KIT_ITEMS;i++) {
+ xml->beginbranch("INSTRUMENT_KIT_ITEM",i);
+ xml->addparbool("enabled",kit[i].Penabled);
+ if (kit[i].Penabled!=0) {
+ xml->addparstr("name",(char *)kit[i].Pname);
+
+ xml->addparbool("muted",kit[i].Pmuted);
+ xml->addpar("min_key",kit[i].Pminkey);
+ xml->addpar("max_key",kit[i].Pmaxkey);
+
+ xml->addpar("send_to_instrument_effect",kit[i].Psendtoparteffect);
+
+ xml->addparbool("add_enabled",kit[i].Padenabled);
+ if ((kit[i].Padenabled!=0)&&(kit[i].adpars!=NULL)) {
+ xml->beginbranch("ADD_SYNTH_PARAMETERS");
+ kit[i].adpars->add2XML(xml);
+ xml->endbranch();
+ };
+
+ xml->addparbool("sub_enabled",kit[i].Psubenabled);
+ if ((kit[i].Psubenabled!=0)&&(kit[i].subpars!=NULL)) {
+ xml->beginbranch("SUB_SYNTH_PARAMETERS");
+ kit[i].subpars->add2XML(xml);
+ xml->endbranch();
+ };
+
+ xml->addparbool("pad_enabled",kit[i].Ppadenabled);
+ if ((kit[i].Ppadenabled!=0)&&(kit[i].padpars!=NULL)) {
+ xml->beginbranch("PAD_SYNTH_PARAMETERS");
+ kit[i].padpars->add2XML(xml);
+ xml->endbranch();
+ };
+
+ };
+ xml->endbranch();
+ };
xml->endbranch();
-
- xml->beginbranch("INSTRUMENT_EFFECTS");
- for (int nefx=0;nefx<NUM_PART_EFX;nefx++){
- xml->beginbranch("INSTRUMENT_EFFECT",nefx);
- xml->beginbranch("EFFECT");
- partefx[nefx]->add2XML(xml);
- xml->endbranch();
-
- xml->addpar("route",Pefxroute[nefx]);
- partefx[nefx]->setdryonly(Pefxroute[nefx]==2);
- xml->addparbool("bypass",Pefxbypass[nefx]);
- xml->endbranch();
+
+ xml->beginbranch("INSTRUMENT_EFFECTS");
+ for (int nefx=0;nefx<NUM_PART_EFX;nefx++) {
+ xml->beginbranch("INSTRUMENT_EFFECT",nefx);
+ xml->beginbranch("EFFECT");
+ partefx[nefx]->add2XML(xml);
+ xml->endbranch();
+
+ xml->addpar("route",Pefxroute[nefx]);
+ partefx[nefx]->setdryonly(Pefxroute[nefx]==2);
+ xml->addparbool("bypass",Pefxbypass[nefx]);
+ xml->endbranch();
};
xml->endbranch();
};
-void Part::add2XML(XMLwrapper *xml){
+void Part::add2XML(XMLwrapper *xml)
+{
//parameters
xml->addparbool("enabled",Penabled);
if ((Penabled==0)&&(xml->minimal)) return;
@@ -923,15 +978,16 @@ void Part::add2XML(XMLwrapper *xml){
xml->addpar("key_limit",Pkeylimit);
xml->beginbranch("INSTRUMENT");
- add2XMLinstrument(xml);
+ add2XMLinstrument(xml);
xml->endbranch();
-
+
xml->beginbranch("CONTROLLER");
- ctl.add2XML(xml);
+ ctl.add2XML(xml);
xml->endbranch();
};
-int Part::saveXML(char *filename){
+int Part::saveXML(char *filename)
+{
XMLwrapper *xml;
xml=new XMLwrapper();
@@ -944,103 +1000,107 @@ int Part::saveXML(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) {
- delete(xml);
- return(-1);
+ delete(xml);
+ return(-1);
};
-
+
if (xml->enterbranch("INSTRUMENT")==0) return(-10);
- getfromXMLinstrument(xml);
+ getfromXMLinstrument(xml);
xml->exitbranch();
-
+
delete(xml);
return(0);
};
-void Part::applyparameters(){
- for (int n=0;n<NUM_KIT_ITEMS;n++){
- if ((kit[n].padpars!=NULL)&&(kit[n].Ppadenabled!=0)) kit[n].padpars->applyparameters(true);
+void Part::applyparameters()
+{
+ for (int n=0;n<NUM_KIT_ITEMS;n++) {
+ if ((kit[n].padpars!=NULL)&&(kit[n].Ppadenabled!=0)) kit[n].padpars->applyparameters(true);
};
};
-void Part::getfromXMLinstrument(XMLwrapper *xml){
- if (xml->enterbranch("INFO")){
- xml->getparstr("name",(char *)Pname,PART_MAX_NAME_LEN);
- xml->getparstr("author",(char *)info.Pauthor,MAX_INFO_TEXT_SIZE);
- xml->getparstr("comments",(char *)info.Pcomments,MAX_INFO_TEXT_SIZE);
- info.Ptype=xml->getpar("type",info.Ptype,0,16);
-
- xml->exitbranch();
+void Part::getfromXMLinstrument(XMLwrapper *xml)
+{
+ if (xml->enterbranch("INFO")) {
+ xml->getparstr("name",(char *)Pname,PART_MAX_NAME_LEN);
+ xml->getparstr("author",(char *)info.Pauthor,MAX_INFO_TEXT_SIZE);
+ xml->getparstr("comments",(char *)info.Pcomments,MAX_INFO_TEXT_SIZE);
+ info.Ptype=xml->getpar("type",info.Ptype,0,16);
+
+ xml->exitbranch();
};
- if (xml->enterbranch("INSTRUMENT_KIT")){
- Pkitmode=xml->getpar127("kit_mode",Pkitmode);
- Pdrummode=xml->getparbool("drum_mode",Pdrummode);
-
- setkititemstatus(0,0);
- for (int i=0;i<NUM_KIT_ITEMS;i++){
- if (xml->enterbranch("INSTRUMENT_KIT_ITEM",i)==0) continue;
- setkititemstatus(i,xml->getparbool("enabled",kit[i].Penabled));
- if (kit[i].Penabled==0) {
- xml->exitbranch();
- continue;
- };
-
- xml->getparstr("name",(char *)kit[i].Pname,PART_MAX_NAME_LEN);
-
- kit[i].Pmuted=xml->getparbool("muted",kit[i].Pmuted);
- kit[i].Pminkey=xml->getpar127("min_key",kit[i].Pminkey);
- kit[i].Pmaxkey=xml->getpar127("max_key",kit[i].Pmaxkey);
-
- kit[i].Psendtoparteffect=xml->getpar127("send_to_instrument_effect",kit[i].Psendtoparteffect);
-
- kit[i].Padenabled=xml->getparbool("add_enabled",kit[i].Padenabled);
- if (xml->enterbranch("ADD_SYNTH_PARAMETERS")){
- kit[i].adpars->getfromXML(xml);
- xml->exitbranch();
- };
-
- kit[i].Psubenabled=xml->getparbool("sub_enabled",kit[i].Psubenabled);
- if (xml->enterbranch("SUB_SYNTH_PARAMETERS")){
- kit[i].subpars->getfromXML(xml);
- xml->exitbranch();
- };
-
- kit[i].Ppadenabled=xml->getparbool("pad_enabled",kit[i].Ppadenabled);
- if (xml->enterbranch("PAD_SYNTH_PARAMETERS")){
- kit[i].padpars->getfromXML(xml);
- xml->exitbranch();
- };
-
- xml->exitbranch();
- };
-
- xml->exitbranch();
+ if (xml->enterbranch("INSTRUMENT_KIT")) {
+ Pkitmode=xml->getpar127("kit_mode",Pkitmode);
+ Pdrummode=xml->getparbool("drum_mode",Pdrummode);
+
+ setkititemstatus(0,0);
+ for (int i=0;i<NUM_KIT_ITEMS;i++) {
+ if (xml->enterbranch("INSTRUMENT_KIT_ITEM",i)==0) continue;
+ setkititemstatus(i,xml->getparbool("enabled",kit[i].Penabled));
+ if (kit[i].Penabled==0) {
+ xml->exitbranch();
+ continue;
+ };
+
+ xml->getparstr("name",(char *)kit[i].Pname,PART_MAX_NAME_LEN);
+
+ kit[i].Pmuted=xml->getparbool("muted",kit[i].Pmuted);
+ kit[i].Pminkey=xml->getpar127("min_key",kit[i].Pminkey);
+ kit[i].Pmaxkey=xml->getpar127("max_key",kit[i].Pmaxkey);
+
+ kit[i].Psendtoparteffect=xml->getpar127("send_to_instrument_effect",kit[i].Psendtoparteffect);
+
+ kit[i].Padenabled=xml->getparbool("add_enabled",kit[i].Padenabled);
+ if (xml->enterbranch("ADD_SYNTH_PARAMETERS")) {
+ kit[i].adpars->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ kit[i].Psubenabled=xml->getparbool("sub_enabled",kit[i].Psubenabled);
+ if (xml->enterbranch("SUB_SYNTH_PARAMETERS")) {
+ kit[i].subpars->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ kit[i].Ppadenabled=xml->getparbool("pad_enabled",kit[i].Ppadenabled);
+ if (xml->enterbranch("PAD_SYNTH_PARAMETERS")) {
+ kit[i].padpars->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ xml->exitbranch();
+ };
+
+ xml->exitbranch();
};
-
- if (xml->enterbranch("INSTRUMENT_EFFECTS")){
- for (int nefx=0;nefx<NUM_PART_EFX;nefx++){
- if (xml->enterbranch("INSTRUMENT_EFFECT",nefx)==0) continue;
- if (xml->enterbranch("EFFECT")){
- partefx[nefx]->getfromXML(xml);
- xml->exitbranch();
- };
-
- 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();
- };
- xml->exitbranch();
- };
+
+ if (xml->enterbranch("INSTRUMENT_EFFECTS")) {
+ for (int nefx=0;nefx<NUM_PART_EFX;nefx++) {
+ if (xml->enterbranch("INSTRUMENT_EFFECT",nefx)==0) continue;
+ if (xml->enterbranch("EFFECT")) {
+ partefx[nefx]->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ 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();
+ };
+ xml->exitbranch();
+ };
};
-void Part::getfromXML(XMLwrapper *xml){
+void Part::getfromXML(XMLwrapper *xml)
+{
Penabled=xml->getparbool("enabled",Penabled);
setPvolume(xml->getpar127("volume",Pvolume));
@@ -1061,14 +1121,14 @@ void Part::getfromXML(XMLwrapper *xml){
Pkeylimit=xml->getpar127("key_limit",Pkeylimit);
- if (xml->enterbranch("INSTRUMENT")){
- getfromXMLinstrument(xml);
+ if (xml->enterbranch("INSTRUMENT")) {
+ getfromXMLinstrument(xml);
xml->exitbranch();
};
-
- if (xml->enterbranch("CONTROLLER")){
- ctl.getfromXML(xml);
- xml->exitbranch();
+
+ if (xml->enterbranch("CONTROLLER")) {
+ ctl.getfromXML(xml);
+ xml->exitbranch();
};
};
diff --git a/src/Misc/Part.h b/src/Misc/Part.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Part.h - Part implementation
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -41,145 +41,146 @@
#include <list> // For the monomemnotes list.
/** Part implementation*/
-class Part{
-
- public:
- /**Constructor
- * @param microtonal_ Pointer to the microtonal object
- * @param fft_ Pointer to the FFTwrapper
- * @param mutex_ Pointer to the master pthread_mutex_t*/
- Part(Microtonal *microtonal_,FFTwrapper *fft_,pthread_mutex_t *mutex_);
- /**Destructor*/
- ~Part();
-
- // Midi commands implemented
- void NoteOn(unsigned char note,unsigned char velocity,int masterkeyshift);
- void NoteOff(unsigned char note);
- void AllNotesOff();//panic
- void SetController(unsigned int type,int par);
- void RelaseSustainedKeys();//this is called when the sustain pedal is relased
- void RelaseAllKeys();//this is called on AllNotesOff controller
-
- /* The synthesizer part output */
- void ComputePartSmps();//Part output
-
- //instrumentonly: 0 - save all, 1 - save only instrumnet, 2 - save only instrument without the name(used in bank)
-
-
- //saves the instrument settings to a XML file
- //returns 0 for ok or <0 if there is an error
- int saveXML(char *filename);
- int loadXMLinstrument(const char *filename);
-
- void add2XML(XMLwrapper *xml);
- void add2XMLinstrument(XMLwrapper *xml);
-
- void defaults();
- void defaultsinstrument();
-
- void applyparameters();
-
- void getfromXML(XMLwrapper *xml);
- void getfromXMLinstrument(XMLwrapper *xml);
-
- void cleanup();
+class Part
+{
+
+public:
+ /**Constructor
+ * @param microtonal_ Pointer to the microtonal object
+ * @param fft_ Pointer to the FFTwrapper
+ * @param mutex_ Pointer to the master pthread_mutex_t*/
+ Part(Microtonal *microtonal_,FFTwrapper *fft_,pthread_mutex_t *mutex_);
+ /**Destructor*/
+ ~Part();
+
+ // Midi commands implemented
+ void NoteOn(unsigned char note,unsigned char velocity,int masterkeyshift);
+ void NoteOff(unsigned char note);
+ void AllNotesOff();//panic
+ void SetController(unsigned int type,int par);
+ void RelaseSustainedKeys();//this is called when the sustain pedal is relased
+ void RelaseAllKeys();//this is called on AllNotesOff controller
+
+ /* The synthesizer part output */
+ void ComputePartSmps();//Part output
+
+ //instrumentonly: 0 - save all, 1 - save only instrumnet, 2 - save only instrument without the name(used in bank)
+
+
+ //saves the instrument settings to a XML file
+ //returns 0 for ok or <0 if there is an error
+ int saveXML(char *filename);
+ int loadXMLinstrument(const char *filename);
+
+ void add2XML(XMLwrapper *xml);
+ void add2XMLinstrument(XMLwrapper *xml);
+
+ void defaults();
+ void defaultsinstrument();
+
+ void applyparameters();
+
+ void getfromXML(XMLwrapper *xml);
+ void getfromXMLinstrument(XMLwrapper *xml);
+
+ void cleanup();
// ADnoteParameters *ADPartParameters;
// SUBnoteParameters *SUBPartParameters;
- //the part's kit
- struct {
+ //the part's kit
+ struct {
unsigned char Penabled,Pmuted,Pminkey,Pmaxkey;
- unsigned char *Pname;
- unsigned char Padenabled,Psubenabled,Ppadenabled;
- unsigned char Psendtoparteffect;
+ unsigned char *Pname;
+ unsigned char Padenabled,Psubenabled,Ppadenabled;
+ unsigned char Psendtoparteffect;
ADnoteParameters *adpars;
SUBnoteParameters *subpars;
PADnoteParameters *padpars;
- } kit[NUM_KIT_ITEMS];
-
-
- //Part parameters
- void setkeylimit(unsigned char Pkeylimit);
- void setkititemstatus(int kititem,int Penabled_);
-
- unsigned char Penabled;/**<if the part is enabled*/
- unsigned char Pvolume;/**<part volume*/
- unsigned char Pminkey;/**<the minimum key that the part receives noteon messages*/
- unsigned char Pmaxkey;//the maximum key that the part receives noteon messages
- void setPvolume(char Pvolume);
- unsigned char Pkeyshift;//Part keyshift
- unsigned char Prcvchn;//from what midi channel it receive commnads
- unsigned char Ppanning;//part panning
- void setPpanning(char Ppanning);
- unsigned char Pvelsns;//velocity sensing (amplitude velocity scale)
- unsigned char Pveloffs;//velocity offset
- unsigned char Pnoteon;//if the part receives NoteOn messages
- unsigned char Pkitmode;//if the kitmode is enabled
- unsigned char Pdrummode;//if all keys are mapped and the system is 12tET (used for drums)
-
- unsigned char Ppolymode;//Part mode - 0=monophonic , 1=polyphonic
- unsigned char Plegatomode;// 0=normal, 1=legato
- unsigned char Pkeylimit;//how many keys are alowed to be played same time (0=off), the older will be relased
-
- unsigned char *Pname; //name of the instrument
- struct{//instrument additional information
- unsigned char Ptype;
- unsigned char Pauthor[MAX_INFO_TEXT_SIZE+1];
- unsigned char Pcomments[MAX_INFO_TEXT_SIZE+1];
- } info;
-
-
- REALTYPE *partoutl;//Left channel output of the part
- REALTYPE *partoutr;//Right channel output of the part
-
- REALTYPE *partfxinputl[NUM_PART_EFX+1],*partfxinputr[NUM_PART_EFX+1];//Left and right signal that pass thru part effects; partfxinput l/r [NUM_PART_EFX] is for "no effect" buffer
-
- enum NoteStatus{KEY_OFF,KEY_PLAYING,KEY_RELASED_AND_SUSTAINED,KEY_RELASED};
-
- REALTYPE volume,oldvolumel,oldvolumer;//this is applied by Master
- REALTYPE panning;//this is applied by Master, too
-
- Controller ctl;//Part controllers
-
- EffectMgr *partefx[NUM_PART_EFX];//insertion part effects (they are part of the instrument)
- unsigned char Pefxroute[NUM_PART_EFX];//how the effect's output is routed(to next effect/to out)
- bool Pefxbypass[NUM_PART_EFX];//if the effects are bypassed
-
-
- pthread_mutex_t *mutex;
-
- int lastnote;
-
- private:
- void KillNotePos(int pos);
- void RelaseNotePos(int pos);
- void MonoMemRenote(); // MonoMem stuff.
-
- int killallnotes;//is set to 1 if I want to kill all notes
-
- struct PartNotes{
+ } kit[NUM_KIT_ITEMS];
+
+
+ //Part parameters
+ void setkeylimit(unsigned char Pkeylimit);
+ void setkititemstatus(int kititem,int Penabled_);
+
+ unsigned char Penabled;/**<if the part is enabled*/
+ unsigned char Pvolume;/**<part volume*/
+ unsigned char Pminkey;/**<the minimum key that the part receives noteon messages*/
+ unsigned char Pmaxkey;//the maximum key that the part receives noteon messages
+ void setPvolume(char Pvolume);
+ unsigned char Pkeyshift;//Part keyshift
+ unsigned char Prcvchn;//from what midi channel it receive commnads
+ unsigned char Ppanning;//part panning
+ void setPpanning(char Ppanning);
+ unsigned char Pvelsns;//velocity sensing (amplitude velocity scale)
+ unsigned char Pveloffs;//velocity offset
+ unsigned char Pnoteon;//if the part receives NoteOn messages
+ unsigned char Pkitmode;//if the kitmode is enabled
+ unsigned char Pdrummode;//if all keys are mapped and the system is 12tET (used for drums)
+
+ unsigned char Ppolymode;//Part mode - 0=monophonic , 1=polyphonic
+ unsigned char Plegatomode;// 0=normal, 1=legato
+ unsigned char Pkeylimit;//how many keys are alowed to be played same time (0=off), the older will be relased
+
+ unsigned char *Pname; //name of the instrument
+ struct {//instrument additional information
+ unsigned char Ptype;
+ unsigned char Pauthor[MAX_INFO_TEXT_SIZE+1];
+ unsigned char Pcomments[MAX_INFO_TEXT_SIZE+1];
+ } info;
+
+
+ REALTYPE *partoutl;//Left channel output of the part
+ REALTYPE *partoutr;//Right channel output of the part
+
+ REALTYPE *partfxinputl[NUM_PART_EFX+1],*partfxinputr[NUM_PART_EFX+1];//Left and right signal that pass thru part effects; partfxinput l/r [NUM_PART_EFX] is for "no effect" buffer
+
+ enum NoteStatus {KEY_OFF,KEY_PLAYING,KEY_RELASED_AND_SUSTAINED,KEY_RELASED};
+
+ REALTYPE volume,oldvolumel,oldvolumer;//this is applied by Master
+ REALTYPE panning;//this is applied by Master, too
+
+ Controller ctl;//Part controllers
+
+ EffectMgr *partefx[NUM_PART_EFX];//insertion part effects (they are part of the instrument)
+ unsigned char Pefxroute[NUM_PART_EFX];//how the effect's output is routed(to next effect/to out)
+ bool Pefxbypass[NUM_PART_EFX];//if the effects are bypassed
+
+
+ pthread_mutex_t *mutex;
+
+ int lastnote;
+
+private:
+ void KillNotePos(int pos);
+ void RelaseNotePos(int pos);
+ void MonoMemRenote(); // MonoMem stuff.
+
+ int killallnotes;//is set to 1 if I want to kill all notes
+
+ struct PartNotes {
NoteStatus status;
int note;//if there is no note playing, the "note"=-1
- int itemsplaying;
- struct {
- ADnote *adnote;
- SUBnote *subnote;
- PADnote *padnote;
- int sendtoparteffect;
- } kititem[NUM_KIT_ITEMS];
- int time;
- };
-
- int lastpos, lastposb; // To keep track of previously used pos and posb.
- bool lastlegatomodevalid; // To keep track of previous legatomodevalid.
-
- // MonoMem stuff
- std::list<unsigned char> monomemnotes; // A list to remember held notes.
- struct {
- unsigned char velocity;
- int mkeyshift;// I'm not sure masterkeyshift should be remembered.
- } monomem[256]; /* 256 is to cover all possible note values.
+ int itemsplaying;
+ struct {
+ ADnote *adnote;
+ SUBnote *subnote;
+ PADnote *padnote;
+ int sendtoparteffect;
+ } kititem[NUM_KIT_ITEMS];
+ int time;
+ };
+
+ int lastpos, lastposb; // To keep track of previously used pos and posb.
+ bool lastlegatomodevalid; // To keep track of previous legatomodevalid.
+
+ // MonoMem stuff
+ std::list<unsigned char> monomemnotes; // A list to remember held notes.
+ struct {
+ unsigned char velocity;
+ int mkeyshift;// I'm not sure masterkeyshift should be remembered.
+ } monomem[256]; /* 256 is to cover all possible note values.
monomem[] is used in conjunction with the list to
store the velocity and masterkeyshift values of a
given note (the list only store note values).
@@ -187,14 +188,14 @@ class Part{
velocity value of the note 'note'.
*/
- PartNotes partnote[POLIPHONY];
-
- REALTYPE *tmpoutl;//used to get the note
- REALTYPE *tmpoutr;
-
- REALTYPE oldfreq;//this is used for portamento
- Microtonal *microtonal;
- FFTwrapper *fft;
+ PartNotes partnote[POLIPHONY];
+
+ REALTYPE *tmpoutl;//used to get the note
+ REALTYPE *tmpoutr;
+
+ REALTYPE oldfreq;//this is used for portamento
+ Microtonal *microtonal;
+ FFTwrapper *fft;
};
#endif
diff --git a/src/Misc/Stereo.C b/src/Misc/Stereo.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Stereo.C - Object for storing a pair of objects
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -21,12 +21,12 @@
template <class T>
Stereo<T>::Stereo(const T &left, const T &right)
- :leftChannel(left),rightChannel(right)
+ :leftChannel(left),rightChannel(right)
{}
template <class T>
Stereo<T>::Stereo(const T &val)
- :leftChannel(val),rightChannel(val)
+ :leftChannel(val),rightChannel(val)
{}
template <class T>
diff --git a/src/Misc/Stereo.h b/src/Misc/Stereo.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Stereo.h - Object for storing a pair of objects
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -22,27 +22,44 @@
#define STEREO_H
template <class T>
-class Stereo{
- public:
- Stereo(const T &left,const T &right);
-
- /**Initializes Stereo with left and right set to val
- * @param val the value for both channels*/
- Stereo(const T &val);
- ~Stereo(){};
-
- void operator=(const Stereo<T> &smp);
- T &left(){return leftChannel;};
- T &right(){return rightChannel;};
- T &l(){return leftChannel;};
- T &r(){return rightChannel;};
- const T &left()const{return leftChannel;};
- const T &right()const{return rightChannel;};
- const T &l()const{return leftChannel;};
- const T &r()const{return rightChannel;};
- private:
- T leftChannel;
- T rightChannel;
+class Stereo
+{
+public:
+ Stereo(const T &left,const T &right);
+
+ /**Initializes Stereo with left and right set to val
+ * @param val the value for both channels*/
+ Stereo(const T &val);
+ ~Stereo() {};
+
+ void operator=(const Stereo<T> &smp);
+ T &left() {
+ return leftChannel;
+ };
+ T &right() {
+ return rightChannel;
+ };
+ T &l() {
+ return leftChannel;
+ };
+ T &r() {
+ return rightChannel;
+ };
+ const T &left()const {
+ return leftChannel;
+ };
+ const T &right()const {
+ return rightChannel;
+ };
+ const T &l()const {
+ return leftChannel;
+ };
+ const T &r()const {
+ return rightChannel;
+ };
+private:
+ T leftChannel;
+ T rightChannel;
};
#include "Stereo.C"
diff --git a/src/Misc/Util.C b/src/Misc/Util.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Util.C - Miscellaneous functions
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -42,17 +42,19 @@ REALTYPE *denormalkillbuf;
/*
* Transform the velocity according the scaling parameter (velocity sensing)
*/
-REALTYPE VelF(REALTYPE velocity,unsigned char scaling){
+REALTYPE VelF(REALTYPE velocity,unsigned char scaling)
+{
REALTYPE x;
x=pow(VELOCITY_MAX_SCALE,(64.0-scaling)/64.0);
if ((scaling==127)||(velocity>0.99)) return(1.0);
- else return(pow(velocity,x));
+ else return(pow(velocity,x));
};
/*
- * Get the detune in cents
+ * Get the detune in cents
*/
-REALTYPE getdetune(unsigned char type,unsigned short int coarsedetune,unsigned short int finedetune){
+REALTYPE getdetune(unsigned char type,unsigned short int coarsedetune,unsigned short int finedetune)
+{
REALTYPE det=0.0,octdet=0.0,cdet=0.0,findet=0.0;
//Get Octave
int octave=coarsedetune/1024;
@@ -62,49 +64,57 @@ REALTYPE getdetune(unsigned char type,unsigned short int coarsedetune,unsigned s
//Coarse and fine detune
int cdetune=coarsedetune%1024;
if (cdetune>512) cdetune-=1024;
-
+
int fdetune=finedetune-8192;
- switch (type){
+ switch (type) {
// case 1: is used for the default (see below)
- case 2: cdet=fabs(cdetune*10.0);
- findet=fabs(fdetune/8192.0)*10.0;
- break;
- case 3: cdet=fabs(cdetune*100);
- findet=pow(10,fabs(fdetune/8192.0)*3.0)/10.0-0.1;
- break;
- case 4: cdet=fabs(cdetune*701.95500087);//perfect fifth
- findet=(pow(2,fabs(fdetune/8192.0)*12.0)-1.0)/4095*1200;
- break;
- //case ...: need to update N_DETUNE_TYPES, if you'll add more
- default:cdet=fabs(cdetune*50.0);
- findet=fabs(fdetune/8192.0)*35.0;//almost like "Paul's Sound Designer 2"
- break;
+ case 2:
+ cdet=fabs(cdetune*10.0);
+ findet=fabs(fdetune/8192.0)*10.0;
+ break;
+ case 3:
+ cdet=fabs(cdetune*100);
+ findet=pow(10,fabs(fdetune/8192.0)*3.0)/10.0-0.1;
+ break;
+ case 4:
+ cdet=fabs(cdetune*701.95500087);//perfect fifth
+ findet=(pow(2,fabs(fdetune/8192.0)*12.0)-1.0)/4095*1200;
+ break;
+ //case ...: need to update N_DETUNE_TYPES, if you'll add more
+ default:
+ cdet=fabs(cdetune*50.0);
+ findet=fabs(fdetune/8192.0)*35.0;//almost like "Paul's Sound Designer 2"
+ break;
};
if (finedetune<8192) findet=-findet;
if (cdetune<0) cdet=-cdet;
-
+
det=octdet+cdet+findet;
return(det);
};
-bool fileexists(const char *filename){
+bool fileexists(const char *filename)
+{
struct stat tmp;
int result=stat(filename,&tmp);
if (result>=0) return(true);
-
+
return(false);
};
-void newFFTFREQS(FFTFREQS *f,int size){
+void newFFTFREQS(FFTFREQS *f,int size)
+{
f->c=new REALTYPE[size];
f->s=new REALTYPE[size];
- for (int i=0;i<size;i++){
- f->c[i]=0.0;f->s[i]=0.0;
+ for (int i=0;i<size;i++) {
+ f->c[i]=0.0;
+ f->s[i]=0.0;
};
};
-void deleteFFTFREQS(FFTFREQS *f){
+void deleteFFTFREQS(FFTFREQS *f)
+{
delete[] f->c;
delete[] f->s;
f->c=f->s=NULL;
diff --git a/src/Misc/Util.h b/src/Misc/Util.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Util.h - Miscellaneous functions
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
diff --git a/src/Misc/XMLwrapper.C b/src/Misc/XMLwrapper.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
XMLwrapper.C - XML wrapper
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -33,40 +33,42 @@ using namespace std;
int xml_k=0;
char tabs[STACKSIZE+2];
-const char *XMLwrapper_whitespace_callback(mxml_node_t *node,int where){
+const char *XMLwrapper_whitespace_callback(mxml_node_t *node,int where)
+{
const char *name=node->value.element.name;
if ((where==MXML_WS_BEFORE_OPEN)&&(!strcmp(name,"?xml"))) return(NULL);
if ((where==MXML_WS_BEFORE_CLOSE)&&(!strcmp(name,"string"))) return(NULL);
if ((where==MXML_WS_BEFORE_OPEN)||(where==MXML_WS_BEFORE_CLOSE)) {
-/* const char *tmp=node->value.element.name;
- if (tmp!=NULL) {
- if ((strstr(tmp,"par")!=tmp)&&(strstr(tmp,"string")!=tmp)) {
- printf("%s ",tmp);
- if (where==MXML_WS_BEFORE_OPEN) xml_k++;
- if (where==MXML_WS_BEFORE_CLOSE) xml_k--;
- if (xml_k>=STACKSIZE) xml_k=STACKSIZE-1;
- if (xml_k<0) xml_k=0;
- printf("%d\n",xml_k);
- printf("\n");
- };
-
- };
- int i=0;
- for (i=1;i<xml_k;i++) tabs[i]='\t';
- tabs[0]='\n';tabs[i+1]='\0';
- if (where==MXML_WS_BEFORE_OPEN) return(tabs);
- else return("\n");
-*/
- return("\n");
+ /* const char *tmp=node->value.element.name;
+ if (tmp!=NULL) {
+ if ((strstr(tmp,"par")!=tmp)&&(strstr(tmp,"string")!=tmp)) {
+ printf("%s ",tmp);
+ if (where==MXML_WS_BEFORE_OPEN) xml_k++;
+ if (where==MXML_WS_BEFORE_CLOSE) xml_k--;
+ if (xml_k>=STACKSIZE) xml_k=STACKSIZE-1;
+ if (xml_k<0) xml_k=0;
+ printf("%d\n",xml_k);
+ printf("\n");
+ };
+
+ };
+ int i=0;
+ for (i=1;i<xml_k;i++) tabs[i]='\t';
+ tabs[0]='\n';tabs[i+1]='\0';
+ if (where==MXML_WS_BEFORE_OPEN) return(tabs);
+ else return("\n");
+ */
+ return("\n");
};
-
+
return(0);
};
-XMLwrapper::XMLwrapper(){
+XMLwrapper::XMLwrapper()
+{
ZERO(&parentstack,(int)sizeof(parentstack));
ZERO(&values,(int)sizeof(values));
@@ -74,50 +76,53 @@ XMLwrapper::XMLwrapper(){
stackpos=0;
information.PADsynth_used=false;
-
+
tree=mxmlNewElement(MXML_NO_PARENT,"?xml version=\"1.0\" encoding=\"UTF-8\"?");
-/* for mxml 2.1 (and older)
- tree=mxmlNewElement(MXML_NO_PARENT,"?xml");
- mxmlElementSetAttr(tree,"version","1.0");
- mxmlElementSetAttr(tree,"encoding","UTF-8");
-*/
-
+ /* for mxml 2.1 (and older)
+ tree=mxmlNewElement(MXML_NO_PARENT,"?xml");
+ mxmlElementSetAttr(tree,"version","1.0");
+ mxmlElementSetAttr(tree,"encoding","UTF-8");
+ */
+
mxml_node_t *doctype=mxmlNewElement(tree,"!DOCTYPE");
mxmlElementSetAttr(doctype,"ZynAddSubFX-data",NULL);
node=root=mxmlNewElement(tree,"ZynAddSubFX-data");
-
+
mxmlElementSetAttr(root,"version-major","1");
mxmlElementSetAttr(root,"version-minor","1");
mxmlElementSetAttr(root,"ZynAddSubFX-author","Nasca Octavian Paul");
//make the empty branch that will contain the information parameters
info=addparams0("INFORMATION");
-
+
//save zynaddsubfx specifications
beginbranch("BASE_PARAMETERS");
- addpar("max_midi_parts",NUM_MIDI_PARTS);
- addpar("max_kit_items_per_instrument",NUM_KIT_ITEMS);
+ addpar("max_midi_parts",NUM_MIDI_PARTS);
+ addpar("max_kit_items_per_instrument",NUM_KIT_ITEMS);
- addpar("max_system_effects",NUM_SYS_EFX);
- addpar("max_insertion_effects",NUM_INS_EFX);
- addpar("max_instrument_effects",NUM_PART_EFX);
+ addpar("max_system_effects",NUM_SYS_EFX);
+ addpar("max_insertion_effects",NUM_INS_EFX);
+ addpar("max_instrument_effects",NUM_PART_EFX);
- addpar("max_addsynth_voices",NUM_VOICES);
+ addpar("max_addsynth_voices",NUM_VOICES);
endbranch();
};
-XMLwrapper::~XMLwrapper(){
+XMLwrapper::~XMLwrapper()
+{
if (tree!=NULL) mxmlDelete(tree);
};
-bool XMLwrapper::checkfileinformation(const char *filename){
+bool XMLwrapper::checkfileinformation(const char *filename)
+{
stackpos=0;
ZERO(&parentstack,(int)sizeof(parentstack));
information.PADsynth_used=false;
- if (tree!=NULL) mxmlDelete(tree);tree=NULL;
+ if (tree!=NULL) mxmlDelete(tree);
+ tree=NULL;
char *xmldata=doloadfile(filename);
if (xmldata==NULL) return(-1);//the file could not be loaded or uncompressed
@@ -126,31 +131,31 @@ bool XMLwrapper::checkfileinformation(const char *filename){
char *end=strstr(xmldata,"</INFORMATION>");
if ((start==NULL)||(end==NULL)||(start>end)) {
- delete []xmldata;
- return(false);
+ delete []xmldata;
+ return(false);
};
end+=strlen("</INFORMATION>");
- end[0]='\0';
-
+ end[0]='\0';
+
tree=mxmlNewElement(MXML_NO_PARENT,"?xml");
node=root=mxmlLoadString(tree,xmldata,MXML_OPAQUE_CALLBACK);
if (root==NULL) {
- delete []xmldata;
- mxmlDelete(tree);
- node=root=tree=NULL;
- return(false);
+ delete []xmldata;
+ mxmlDelete(tree);
+ node=root=tree=NULL;
+ return(false);
};
root=mxmlFindElement(tree,tree,"INFORMATION",NULL,NULL,MXML_DESCEND);
push(root);
- if (root==NULL){
- delete []xmldata;
- mxmlDelete(tree);
- node=root=tree=NULL;
- return(false);
+ if (root==NULL) {
+ delete []xmldata;
+ mxmlDelete(tree);
+ node=root=tree=NULL;
+ return(false);
};
-
+
information.PADsynth_used=getparbool("PADsynth_used",false);
exitbranch();
@@ -164,27 +169,29 @@ bool XMLwrapper::checkfileinformation(const char *filename){
/* SAVE XML members */
-int XMLwrapper::saveXMLfile(const string &filename){
+int XMLwrapper::saveXMLfile(const string &filename)
+{
char *xmldata=getXMLdata();
if (xmldata==NULL) return(-2);
int compression=config.cfg.GzipCompression;
int result=dosavefile(filename.c_str(),compression,xmldata);
-
- free(xmldata);
+
+ free(xmldata);
return(result);
};
-char *XMLwrapper::getXMLdata(){
+char *XMLwrapper::getXMLdata()
+{
xml_k=0;
ZERO(tabs,STACKSIZE+2);
-
+
mxml_node_t *oldnode=node;
-
+
node=info;
//Info storing
addparbool("PADsynth_used",information.PADsynth_used);
-
+
node=oldnode;
char *xmldata=mxmlSaveAllocString(tree,XMLwrapper_whitespace_callback);
@@ -192,62 +199,70 @@ char *XMLwrapper::getXMLdata(){
};
-int XMLwrapper::dosavefile(const char *filename,int compression,const char *xmldata){
- if (compression==0){
- FILE *file;
- file=fopen(filename,"w");
- if (file==NULL) return(-1);
- fputs(xmldata,file);
- fclose(file);
+int XMLwrapper::dosavefile(const char *filename,int compression,const char *xmldata)
+{
+ if (compression==0) {
+ FILE *file;
+ file=fopen(filename,"w");
+ if (file==NULL) return(-1);
+ fputs(xmldata,file);
+ fclose(file);
} else {
- if (compression>9) compression=9;
- if (compression<1) compression=1;
- char options[10];
- snprintf(options,10,"wb%d",compression);
-
- gzFile gzfile;
- gzfile=gzopen(filename,options);
- if (gzfile==NULL) return(-1);
- gzputs(gzfile,xmldata);
- gzclose(gzfile);
+ if (compression>9) compression=9;
+ if (compression<1) compression=1;
+ char options[10];
+ snprintf(options,10,"wb%d",compression);
+
+ gzFile gzfile;
+ gzfile=gzopen(filename,options);
+ if (gzfile==NULL) return(-1);
+ gzputs(gzfile,xmldata);
+ gzclose(gzfile);
};
-
+
return(0);
};
-void XMLwrapper::addpar(const string &name,int val){
+void XMLwrapper::addpar(const string &name,int val)
+{
addparams2("par","name",name.c_str(),"value",int2str(val));
};
-void XMLwrapper::addparreal(const string &name,REALTYPE val){
+void XMLwrapper::addparreal(const string &name,REALTYPE val)
+{
addparams2("par_real","name",name.c_str(),"value",real2str(val));
};
-void XMLwrapper::addparbool(const string &name,int val){
+void XMLwrapper::addparbool(const string &name,int val)
+{
if (val!=0) addparams2("par_bool","name",name.c_str(),"value","yes");
else addparams2("par_bool","name",name.c_str(),"value","no");
};
-void XMLwrapper::addparstr(const string &name,const string &val){
+void XMLwrapper::addparstr(const string &name,const string &val)
+{
mxml_node_t *element=mxmlNewElement(node,"string");
mxmlElementSetAttr(element,"name",name.c_str());
mxmlNewText(element,0,val.c_str());
};
-void XMLwrapper::beginbranch(const string &name){
+void XMLwrapper::beginbranch(const string &name)
+{
push(node);
node=addparams0(name.c_str());
};
-void XMLwrapper::beginbranch(const string &name,int id){
+void XMLwrapper::beginbranch(const string &name,int id)
+{
push(node);
node=addparams1(name.c_str(),"id",int2str(id));
};
-void XMLwrapper::endbranch(){
+void XMLwrapper::endbranch()
+{
node=pop();
};
@@ -255,7 +270,8 @@ void XMLwrapper::endbranch(){
/* LOAD XML members */
-int XMLwrapper::loadXMLfile(const string &filename){
+int XMLwrapper::loadXMLfile(const string &filename)
+{
if (tree!=NULL) mxmlDelete(tree);
tree=NULL;
@@ -264,16 +280,16 @@ int XMLwrapper::loadXMLfile(const string &filename){
stackpos=0;
- const char *xmldata=doloadfile(filename.c_str());
+ const char *xmldata=doloadfile(filename.c_str());
if (xmldata==NULL) return(-1);//the file could not be loaded or uncompressed
-
+
root=tree=mxmlLoadString(NULL,xmldata,MXML_OPAQUE_CALLBACK);
delete []xmldata;
if (tree==NULL) return(-2);//this is not XML
-
-
+
+
node=root=mxmlFindElement(tree,tree,"ZynAddSubFX-data",NULL,NULL,MXML_DESCEND);
if (root==NULL) return(-3);//the XML doesnt embbed zynaddsubfx data
push(root);
@@ -285,51 +301,53 @@ int XMLwrapper::loadXMLfile(const string &filename){
};
-char *XMLwrapper::doloadfile(const string &filename){
+char *XMLwrapper::doloadfile(const string &filename)
+{
char *xmldata=NULL;
int filesize=-1;
-
+
//try get filesize as gzip data (first)
gzFile gzfile=gzopen(filename.c_str(),"rb");
- if (gzfile!=NULL){//this is a gzip file
- // first check it's size
- while(!gzeof(gzfile)) {
- gzseek (gzfile,1024*1024,SEEK_CUR);
- if (gztell(gzfile)>10000000) {
- gzclose(gzfile);
- goto notgzip;//the file is too big
- };
- };
- filesize=gztell(gzfile);
-
- //rewind the file and load the data
- xmldata=new char[filesize+1];
- ZERO(xmldata,filesize+1);
-
- gzrewind(gzfile);
- gzread(gzfile,xmldata,filesize);
-
- gzclose(gzfile);
- return (xmldata);
+ if (gzfile!=NULL) {//this is a gzip file
+ // first check it's size
+ while (!gzeof(gzfile)) {
+ gzseek (gzfile,1024*1024,SEEK_CUR);
+ if (gztell(gzfile)>10000000) {
+ gzclose(gzfile);
+ goto notgzip;//the file is too big
+ };
+ };
+ filesize=gztell(gzfile);
+
+ //rewind the file and load the data
+ xmldata=new char[filesize+1];
+ ZERO(xmldata,filesize+1);
+
+ gzrewind(gzfile);
+ gzread(gzfile,xmldata,filesize);
+
+ gzclose(gzfile);
+ return (xmldata);
} else {//this is not a gzip file
- notgzip:
- FILE *file=fopen(filename.c_str(),"rb");
- if (file==NULL) return(NULL);
- fseek(file,0,SEEK_END);
- filesize=ftell(file);
-
- xmldata=new char [filesize+1];
- ZERO(xmldata,filesize+1);
-
- rewind(file);
- fread(xmldata,filesize,1,file);
-
- fclose(file);
- return(xmldata);
- };
-};
-
-bool XMLwrapper::putXMLdata(const char *xmldata){
+notgzip:
+ FILE *file=fopen(filename.c_str(),"rb");
+ if (file==NULL) return(NULL);
+ fseek(file,0,SEEK_END);
+ filesize=ftell(file);
+
+ xmldata=new char [filesize+1];
+ ZERO(xmldata,filesize+1);
+
+ rewind(file);
+ fread(xmldata,filesize,1,file);
+
+ fclose(file);
+ return(xmldata);
+ };
+};
+
+bool XMLwrapper::putXMLdata(const char *xmldata)
+{
if (tree!=NULL) mxmlDelete(tree);
tree=NULL;
@@ -339,11 +357,11 @@ bool XMLwrapper::putXMLdata(const char *xmldata){
stackpos=0;
if (xmldata==NULL) return (false);
-
+
root=tree=mxmlLoadString(NULL,xmldata,MXML_OPAQUE_CALLBACK);
if (tree==NULL) return(false);
-
+
node=root=mxmlFindElement(tree,tree,"ZynAddSubFX-data",NULL,NULL,MXML_DESCEND);
if (root==NULL) return (false);;
push(root);
@@ -353,7 +371,8 @@ bool XMLwrapper::putXMLdata(const char *xmldata){
-int XMLwrapper::enterbranch(const string &name){
+int XMLwrapper::enterbranch(const string &name)
+{
node=mxmlFindElement(peek(),peek(),name.c_str(),NULL,NULL,MXML_DESCEND_FIRST);
if (node==NULL) return(0);
@@ -361,7 +380,8 @@ int XMLwrapper::enterbranch(const string &name){
return(1);
};
-int XMLwrapper::enterbranch(const string &name,int id){
+int XMLwrapper::enterbranch(const string &name,int id)
+{
snprintf(tmpstr,TMPSTR_SIZE,"%d",id);
node=mxmlFindElement(peek(),peek(),name.c_str(),"id",tmpstr,MXML_DESCEND_FIRST);
if (node==NULL) return(0);
@@ -371,118 +391,133 @@ int XMLwrapper::enterbranch(const string &name,int id){
};
-void XMLwrapper::exitbranch(){
+void XMLwrapper::exitbranch()
+{
pop();
};
-int XMLwrapper::getbranchid(int min, int max){
+int XMLwrapper::getbranchid(int min, int max)
+{
int id=str2int(mxmlElementGetAttr(node,"id"));
if ((min==0)&&(max==0)) return(id);
-
+
if (id<min) id=min;
- else if (id>max) id=max;
+ else if (id>max) id=max;
return(id);
};
-int XMLwrapper::getpar(const string &name,int defaultpar,int min,int max){
+int XMLwrapper::getpar(const string &name,int defaultpar,int min,int max)
+{
node=mxmlFindElement(peek(),peek(),"par","name",name.c_str(),MXML_DESCEND_FIRST);
if (node==NULL) return(defaultpar);
const char *strval=mxmlElementGetAttr(node,"value");
if (strval==NULL) return(defaultpar);
-
+
int val=str2int(strval);
if (val<min) val=min;
- else if (val>max) val=max;
-
+ else if (val>max) val=max;
+
return(val);
};
-int XMLwrapper::getpar127(const string &name,int defaultpar){
+int XMLwrapper::getpar127(const string &name,int defaultpar)
+{
return(getpar(name,defaultpar,0,127));
};
-int XMLwrapper::getparbool(const string &name,int defaultpar){
+int XMLwrapper::getparbool(const string &name,int defaultpar)
+{
node=mxmlFindElement(peek(),peek(),"par_bool","name",name.c_str(),MXML_DESCEND_FIRST);
if (node==NULL) return(defaultpar);
const char *strval=mxmlElementGetAttr(node,"value");
if (strval==NULL) return(defaultpar);
-
+
if ((strval[0]=='Y')||(strval[0]=='y')) return(1);
- else return(0);
+ else return(0);
};
-void XMLwrapper::getparstr(const string &name,char *par,int maxstrlen){
+void XMLwrapper::getparstr(const string &name,char *par,int maxstrlen)
+{
ZERO(par,maxstrlen);
node=mxmlFindElement(peek(),peek(),"string","name",name.c_str(),MXML_DESCEND_FIRST);
-
+
if (node==NULL) return;
if (node->child==NULL) return;
if (node->child->type!=MXML_OPAQUE) return;
-
+
snprintf(par,maxstrlen,"%s",node->child->value.element.name);
-
+
};
-REALTYPE XMLwrapper::getparreal(const char *name,REALTYPE defaultpar){
+REALTYPE XMLwrapper::getparreal(const char *name,REALTYPE defaultpar)
+{
node=mxmlFindElement(peek(),peek(),"par_real","name",name,MXML_DESCEND_FIRST);
if (node==NULL) return(defaultpar);
const char *strval=mxmlElementGetAttr(node,"value");
if (strval==NULL) return(defaultpar);
-
+
return(str2real(strval));
};
-REALTYPE XMLwrapper::getparreal(const char *name,REALTYPE defaultpar,REALTYPE min,REALTYPE max){
+REALTYPE XMLwrapper::getparreal(const char *name,REALTYPE defaultpar,REALTYPE min,REALTYPE max)
+{
REALTYPE result=getparreal(name,defaultpar);
-
+
if (result<min) result=min;
- else if (result>max) result=max;
+ else if (result>max) result=max;
return(result);
};
/** Private members **/
-char *XMLwrapper::int2str(int x){
+char *XMLwrapper::int2str(int x)
+{
snprintf(tmpstr,TMPSTR_SIZE,"%d",x);
return(tmpstr);
};
-char *XMLwrapper::real2str(REALTYPE x){
+char *XMLwrapper::real2str(REALTYPE x)
+{
snprintf(tmpstr,TMPSTR_SIZE,"%g",x);
return(tmpstr);
};
-int XMLwrapper::str2int(const char *str){
+int XMLwrapper::str2int(const char *str)
+{
if (str==NULL) return(0);
int result=strtol(str,NULL,10);
return(result);
};
-REALTYPE XMLwrapper::str2real(const char *str){
+REALTYPE XMLwrapper::str2real(const char *str)
+{
if (str==NULL) return(0.0);
REALTYPE result=strtod(str,NULL);
return(result);
};
-mxml_node_t *XMLwrapper::addparams0(const char *name){
+mxml_node_t *XMLwrapper::addparams0(const char *name)
+{
mxml_node_t *element=mxmlNewElement(node,name);
return(element);
};
-mxml_node_t *XMLwrapper::addparams1(const char *name,const char *par1,const char *val1){
+mxml_node_t *XMLwrapper::addparams1(const char *name,const char *par1,const char *val1)
+{
mxml_node_t *element=mxmlNewElement(node,name);
mxmlElementSetAttr(element,par1,val1);
return(element);
};
-mxml_node_t *XMLwrapper::addparams2(const char *name,const char *par1,const char *val1,const char *par2, const char *val2){
+mxml_node_t *XMLwrapper::addparams2(const char *name,const char *par1,const char *val1,const char *par2, const char *val2)
+{
mxml_node_t *element=mxmlNewElement(node,name);
mxmlElementSetAttr(element,par1,val1);
mxmlElementSetAttr(element,par2,val2);
@@ -492,21 +527,23 @@ mxml_node_t *XMLwrapper::addparams2(const char *name,const char *par1,const char
-void XMLwrapper::push(mxml_node_t *node){
+void XMLwrapper::push(mxml_node_t *node)
+{
if (stackpos>=STACKSIZE-1) {
- printf("BUG!: XMLwrapper::push() - full parentstack\n");
- return;
+ printf("BUG!: XMLwrapper::push() - full parentstack\n");
+ return;
};
stackpos++;
parentstack[stackpos]=node;
-
+
// printf("push %d - %s\n",stackpos,node->value.element.name);
-
+
};
-mxml_node_t *XMLwrapper::pop(){
+mxml_node_t *XMLwrapper::pop()
+{
if (stackpos<=0) {
- printf("BUG!: XMLwrapper::pop() - empty parentstack\n");
- return (root);
+ printf("BUG!: XMLwrapper::pop() - empty parentstack\n");
+ return (root);
};
mxml_node_t *node=parentstack[stackpos];
parentstack[stackpos]=NULL;
@@ -517,10 +554,11 @@ mxml_node_t *XMLwrapper::pop(){
return(node);
};
-mxml_node_t *XMLwrapper::peek(){
+mxml_node_t *XMLwrapper::peek()
+{
if (stackpos<=0) {
- printf("BUG!: XMLwrapper::peek() - empty parentstack\n");
- return (root);
+ printf("BUG!: XMLwrapper::peek() - empty parentstack\n");
+ return (root);
};
return(parentstack[stackpos]);
};
diff --git a/src/Misc/XMLwrapper.h b/src/Misc/XMLwrapper.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
XML.h - XML wrapper
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -34,143 +34,144 @@
//the maxim tree depth
#define STACKSIZE 100
-class XMLwrapper{
- public:
- XMLwrapper();
- ~XMLwrapper();
-
- /********************************/
- /* SAVE to XML */
- /********************************/
-
- //returns 0 if ok or -1 if the file cannot be saved
- int saveXMLfile(const std::string &filename);
-
- //returns the new allocated string that contains the XML data (used for clipboard)
- //the string is NULL terminated
- char *getXMLdata();
-
- //add simple parameter (name and value)
- void addpar(const std::string &name,int val);
- void addparreal(const std::string &name,REALTYPE val);
-
- //add boolean parameter (name and boolean value)
- //if the value is 0 => "yes", else "no"
- void addparbool(const std::string &name,int val);
-
- //add string parameter (name and string)
- void addparstr(const std::string &name,const std::string &val);
-
- //add a branch
- void beginbranch(const std::string &name);
- void beginbranch(const std::string &name, int id);
-
- //this must be called after each branch (nodes that contains child nodes)
- void endbranch();
-
- /********************************/
- /* LOAD from XML */
- /********************************/
-
- //returns 0 if ok or -1 if the file cannot be loaded
- int loadXMLfile(const std::string &filename);
-
- //used by the clipboard
- bool putXMLdata(const char *xmldata);
-
- //enter into the branch
- //returns 1 if is ok, or 0 otherwise
- int enterbranch(const std::string &name);
-
-
- //enter into the branch with id
- //returns 1 if is ok, or 0 otherwise
- int enterbranch(const std::string &name, int id);
-
- //exits from a branch
- void exitbranch();
-
- //get the the branch_id and limits it between the min and max
- //if min==max==0, it will not limit it
- //if there isn't any id, will return min
- //this must be called only imediately after enterbranch()
- int getbranchid(int min, int max);
-
- //it returns the parameter and limits it between min and max
- //if min==max==0, it will not limit it
- //if no parameter will be here, the defaultpar will be returned
- int getpar(const std::string &name,int defaultpar,int min,int max);
-
- //the same as getpar, but the limits are 0 and 127
- int getpar127(const std::string &name,int defaultpar);
-
- int getparbool(const std::string &name,int defaultpar);
-
- void getparstr(const std::string &name,char *par,int maxstrlen);
- REALTYPE getparreal(const char *name,REALTYPE defaultpar);
- REALTYPE getparreal(const char *name,REALTYPE defaultpar,REALTYPE min,REALTYPE max);
-
- bool minimal;//false if all parameters will be stored (used only for clipboard)
-
- struct {
- bool PADsynth_used;
- }information;
-
- //opens a file and parse only the "information" data on it
- //returns "true" if all went ok or "false" on errors
- bool checkfileinformation(const char *filename);
-
- private:
-
- int dosavefile(const char *filename,int compression,const char *xmldata);
- char *doloadfile(const std::string &filename);
-
-
- mxml_node_t *tree;//all xml data
- mxml_node_t *root;//xml data used by zynaddsubfx
- mxml_node_t *node;//current node
- mxml_node_t *info;//this node is used to store the information about the data
-
- //adds params like this:
- // <name>
- //returns the node
- mxml_node_t *addparams0(const char *name);
-
- //adds params like this:
- // <name par1="val1">
- //returns the node
- mxml_node_t *addparams1(const char *name,const char *par1,const char *val1);
-
- //adds params like this:
- // <name par1="val1" par2="val2">
- //returns the node
- mxml_node_t *addparams2(const char *name,const char *par1,const char *val1,const char *par2, const char *val2);
-
- char *int2str(int x);
- char *real2str(REALTYPE x);
-
- int str2int(const char *str);
- REALTYPE str2real(const char *str);
-
- char tmpstr[TMPSTR_SIZE];
-
-
- //this is used to store the parents
- mxml_node_t *parentstack[STACKSIZE];
- int stackpos;
-
-
- void push(mxml_node_t *node);
- mxml_node_t *pop();
- mxml_node_t *peek();
-
- //theese are used to store the values
- struct{
- struct {
- int major,minor;
- }xml_version;
- }values;
-
+class XMLwrapper
+{
+public:
+ XMLwrapper();
+ ~XMLwrapper();
+
+ /********************************/
+ /* SAVE to XML */
+ /********************************/
+
+ //returns 0 if ok or -1 if the file cannot be saved
+ int saveXMLfile(const std::string &filename);
+
+ //returns the new allocated string that contains the XML data (used for clipboard)
+ //the string is NULL terminated
+ char *getXMLdata();
+
+ //add simple parameter (name and value)
+ void addpar(const std::string &name,int val);
+ void addparreal(const std::string &name,REALTYPE val);
+
+ //add boolean parameter (name and boolean value)
+ //if the value is 0 => "yes", else "no"
+ void addparbool(const std::string &name,int val);
+
+ //add string parameter (name and string)
+ void addparstr(const std::string &name,const std::string &val);
+
+ //add a branch
+ void beginbranch(const std::string &name);
+ void beginbranch(const std::string &name, int id);
+
+ //this must be called after each branch (nodes that contains child nodes)
+ void endbranch();
+
+ /********************************/
+ /* LOAD from XML */
+ /********************************/
+
+ //returns 0 if ok or -1 if the file cannot be loaded
+ int loadXMLfile(const std::string &filename);
+
+ //used by the clipboard
+ bool putXMLdata(const char *xmldata);
+
+ //enter into the branch
+ //returns 1 if is ok, or 0 otherwise
+ int enterbranch(const std::string &name);
+
+
+ //enter into the branch with id
+ //returns 1 if is ok, or 0 otherwise
+ int enterbranch(const std::string &name, int id);
+
+ //exits from a branch
+ void exitbranch();
+
+ //get the the branch_id and limits it between the min and max
+ //if min==max==0, it will not limit it
+ //if there isn't any id, will return min
+ //this must be called only imediately after enterbranch()
+ int getbranchid(int min, int max);
+
+ //it returns the parameter and limits it between min and max
+ //if min==max==0, it will not limit it
+ //if no parameter will be here, the defaultpar will be returned
+ int getpar(const std::string &name,int defaultpar,int min,int max);
+
+ //the same as getpar, but the limits are 0 and 127
+ int getpar127(const std::string &name,int defaultpar);
+
+ int getparbool(const std::string &name,int defaultpar);
+
+ void getparstr(const std::string &name,char *par,int maxstrlen);
+ REALTYPE getparreal(const char *name,REALTYPE defaultpar);
+ REALTYPE getparreal(const char *name,REALTYPE defaultpar,REALTYPE min,REALTYPE max);
+
+ bool minimal;//false if all parameters will be stored (used only for clipboard)
+
+ struct {
+ bool PADsynth_used;
+ }information;
+
+ //opens a file and parse only the "information" data on it
+ //returns "true" if all went ok or "false" on errors
+ bool checkfileinformation(const char *filename);
+
+private:
+
+ int dosavefile(const char *filename,int compression,const char *xmldata);
+ char *doloadfile(const std::string &filename);
+
+
+ mxml_node_t *tree;//all xml data
+ mxml_node_t *root;//xml data used by zynaddsubfx
+ mxml_node_t *node;//current node
+ mxml_node_t *info;//this node is used to store the information about the data
+
+ //adds params like this:
+ // <name>
+ //returns the node
+ mxml_node_t *addparams0(const char *name);
+
+ //adds params like this:
+ // <name par1="val1">
+ //returns the node
+ mxml_node_t *addparams1(const char *name,const char *par1,const char *val1);
+
+ //adds params like this:
+ // <name par1="val1" par2="val2">
+ //returns the node
+ mxml_node_t *addparams2(const char *name,const char *par1,const char *val1,const char *par2, const char *val2);
+
+ char *int2str(int x);
+ char *real2str(REALTYPE x);
+
+ int str2int(const char *str);
+ REALTYPE str2real(const char *str);
+
+ char tmpstr[TMPSTR_SIZE];
+
+
+ //this is used to store the parents
+ mxml_node_t *parentstack[STACKSIZE];
+ int stackpos;
+
+
+ void push(mxml_node_t *node);
+ mxml_node_t *pop();
+ mxml_node_t *peek();
+
+ //theese are used to store the values
+ struct {
+ struct {
+ int major,minor;
+ }xml_version;
+ }values;
+
};
#endif
diff --git a/src/Output/DSSIaudiooutput.C b/src/Output/DSSIaudiooutput.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
DSSIaudiooutput.C - Audio functions for DSSI
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -20,7 +20,7 @@
*/
-//this file contains code used from trivial_synth.c from
+//this file contains code used from trivial_synth.c from
//the DSSI (published by Steve Harris under public domain) as a template
//the code is incomplete
#include <string.h>
@@ -37,54 +37,58 @@ typedef struct {
} TS;
-static void cleanupTS(LADSPA_Handle instance){
+static void cleanupTS(LADSPA_Handle instance)
+{
free(instance);
}
static void connectPortTS(LADSPA_Handle instance, unsigned long port,
- LADSPA_Data * data){
+ LADSPA_Data * data)
+{
TS *plugin;
plugin = (TS *) instance;
switch (port) {
case 0:
- plugin->outl = data;
- break;
+ plugin->outl = data;
+ break;
case 1:
- plugin->outr = data;
- break;
+ plugin->outr = data;
+ break;
}
}
-const LADSPA_Descriptor *ladspa_descriptor(unsigned long index){
+const LADSPA_Descriptor *ladspa_descriptor(unsigned long index)
+{
switch (index) {
case 0:
- return tsLDescriptor;
+ return tsLDescriptor;
default:
- return NULL;
+ return NULL;
}
}
-const DSSI_Descriptor *dssi_descriptor(unsigned long index){
+const DSSI_Descriptor *dssi_descriptor(unsigned long index)
+{
// FILE *a=fopen("/tmp/zzzzz11z","w");
// fprintf(a,"aaaaaaaaaaa TEST\n");
// fclose(a);
switch (index) {
case 0:
- return tsDDescriptor;
+ return tsDDescriptor;
default:
- return NULL;
+ return NULL;
}
}
static LADSPA_Handle instantiateTS(const LADSPA_Descriptor * descriptor,
- unsigned long s_rate)
+ unsigned long s_rate)
{
TS *plugin_data = (TS *) malloc(sizeof(TS));
-/* for (i=0; i<MIDI_NOTES; i++) {
- plugin_data->omega[i] = M_PI * 2.0 / (double)s_rate *
- pow(2.0, (i-69.0) / 12.0);
- }
-*/
+ /* for (i=0; i<MIDI_NOTES; i++) {
+ plugin_data->omega[i] = M_PI * 2.0 / (double)s_rate *
+ pow(2.0, (i-69.0) / 12.0);
+ }
+ */
return (LADSPA_Handle) plugin_data;
}
@@ -99,7 +103,8 @@ static void activateTS(LADSPA_Handle instance)
static void runTS(LADSPA_Handle instance, unsigned long sample_count,
- snd_seq_event_t *events, unsigned long event_count){
+ snd_seq_event_t *events, unsigned long event_count)
+{
TS *plugin_data = (TS *) instance;
// LADSPA_Data *const output = plugin_data->output;
// LADSPA_Data freq = *(plugin_data->freq);
@@ -109,67 +114,70 @@ static void runTS(LADSPA_Handle instance, unsigned long sample_count,
unsigned long event_pos;
unsigned long note;
-/* if (freq < 1.0) {
- freq = 440.0f;
- }
- if (vol < 0.000001) {
- vol = 1.0f;
- }
-
- if (event_count > 0) {
- printf("trivial_synth: have %ld events\n", event_count);
- }
-
- for (pos = 0, event_pos = 0; pos < sample_count; pos++) {
-
- while (event_pos < event_count
- && pos == events[event_pos].time.tick) {
-
- printf("trivial_synth: event type %d\n", events[event_pos].type);
-
- if (events[event_pos].type == SND_SEQ_EVENT_NOTEON) {
- data[events[event_pos].data.note.note].amp =
- events[event_pos].data.note.velocity / 512.0f;
- data[events[event_pos].data.note.note].
- active = events[event_pos].data.note.velocity > 0;
- data[events[event_pos].data.note.note].
- phase = 0.0;
- } else if (events[event_pos].type == SND_SEQ_EVENT_NOTEOFF) {
- data[events[event_pos].data.note.note].
- active = 0;
- }
- event_pos++;
- }
-
- output[pos] = 0.0f;
- for (note = 0; note < MIDI_NOTES; note++) {
- if (data[note].active) {
- output[pos] += sin(data[note].phase) * data[note].amp * vol;
- data[note].phase += plugin_data->omega[note] * freq;
- if (data[note].phase > M_PI * 2.0) {
- data[note].phase -= M_PI * 2.0;
- }
- }
- }
- }
- */
+ /* if (freq < 1.0) {
+ freq = 440.0f;
+ }
+ if (vol < 0.000001) {
+ vol = 1.0f;
+ }
+
+ if (event_count > 0) {
+ printf("trivial_synth: have %ld events\n", event_count);
+ }
+
+ for (pos = 0, event_pos = 0; pos < sample_count; pos++) {
+
+ while (event_pos < event_count
+ && pos == events[event_pos].time.tick) {
+
+ printf("trivial_synth: event type %d\n", events[event_pos].type);
+
+ if (events[event_pos].type == SND_SEQ_EVENT_NOTEON) {
+ data[events[event_pos].data.note.note].amp =
+ events[event_pos].data.note.velocity / 512.0f;
+ data[events[event_pos].data.note.note].
+ active = events[event_pos].data.note.velocity > 0;
+ data[events[event_pos].data.note.note].
+ phase = 0.0;
+ } else if (events[event_pos].type == SND_SEQ_EVENT_NOTEOFF) {
+ data[events[event_pos].data.note.note].
+ active = 0;
+ }
+ event_pos++;
+ }
+
+ output[pos] = 0.0f;
+ for (note = 0; note < MIDI_NOTES; note++) {
+ if (data[note].active) {
+ output[pos] += sin(data[note].phase) * data[note].amp * vol;
+ data[note].phase += plugin_data->omega[note] * freq;
+ if (data[note].phase > M_PI * 2.0) {
+ data[note].phase -= M_PI * 2.0;
+ }
+ }
+ }
+ }
+ */
}
static void runTSWrapper(LADSPA_Handle instance,
- unsigned long sample_count){
+ unsigned long sample_count)
+{
runTS(instance, sample_count, NULL, 0);
}
-int getControllerTS(LADSPA_Handle instance, unsigned long port){
+int getControllerTS(LADSPA_Handle instance, unsigned long port)
+{
return -1;
}
-void _init(){
+void _init()
+{
char **port_names;
LADSPA_PortDescriptor *port_descriptors;
LADSPA_PortRangeHint *port_range_hints;
-
+
FILE *a=fopen("/tmp/zzzzzz","w");
fprintf(a,"aaaaaaaaaaa TEST\n");
fclose(a);
@@ -177,63 +185,64 @@ void _init(){
tsLDescriptor = (LADSPA_Descriptor *) malloc(sizeof(LADSPA_Descriptor));
if (tsLDescriptor) {
- tsLDescriptor->UniqueID = 100;
- tsLDescriptor->Label = "ZASF";
- tsLDescriptor->Properties = 0;
- tsLDescriptor->Name = "ZynAddSubFX";
- tsLDescriptor->Maker = "Nasca Octavian Paul <[email protected]>";
- tsLDescriptor->Copyright = "GNU General Public License v.2";
- tsLDescriptor->PortCount = 2;
-
- port_descriptors = (LADSPA_PortDescriptor *)
- calloc(tsLDescriptor->PortCount, sizeof
- (LADSPA_PortDescriptor));
- tsLDescriptor->PortDescriptors =
- (const LADSPA_PortDescriptor *) port_descriptors;
-
- port_range_hints = (LADSPA_PortRangeHint *)
- calloc(tsLDescriptor->PortCount, sizeof
- (LADSPA_PortRangeHint));
- tsLDescriptor->PortRangeHints =
- (const LADSPA_PortRangeHint *) port_range_hints;
-
- port_names = (char **) calloc(tsLDescriptor->PortCount, sizeof(char *));
- tsLDescriptor->PortNames = (const char **) port_names;
-
- port_descriptors[0] = LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO;
- port_names[0] = "Output L";
- port_range_hints[0].HintDescriptor = 0;
- port_descriptors[1] = LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO;
- port_names[1] = "Output R";
- port_range_hints[1].HintDescriptor = 0;
-
- tsLDescriptor->activate = activateTS;
- tsLDescriptor->cleanup = cleanupTS;
- tsLDescriptor->connect_port = connectPortTS;
- tsLDescriptor->deactivate = NULL;
- tsLDescriptor->instantiate = instantiateTS;
- tsLDescriptor->run = runTSWrapper;
- tsLDescriptor->run_adding = NULL;
- tsLDescriptor->set_run_adding_gain = NULL;
+ tsLDescriptor->UniqueID = 100;
+ tsLDescriptor->Label = "ZASF";
+ tsLDescriptor->Properties = 0;
+ tsLDescriptor->Name = "ZynAddSubFX";
+ tsLDescriptor->Maker = "Nasca Octavian Paul <[email protected]>";
+ tsLDescriptor->Copyright = "GNU General Public License v.2";
+ tsLDescriptor->PortCount = 2;
+
+ port_descriptors = (LADSPA_PortDescriptor *)
+ calloc(tsLDescriptor->PortCount, sizeof
+ (LADSPA_PortDescriptor));
+ tsLDescriptor->PortDescriptors =
+ (const LADSPA_PortDescriptor *) port_descriptors;
+
+ port_range_hints = (LADSPA_PortRangeHint *)
+ calloc(tsLDescriptor->PortCount, sizeof
+ (LADSPA_PortRangeHint));
+ tsLDescriptor->PortRangeHints =
+ (const LADSPA_PortRangeHint *) port_range_hints;
+
+ port_names = (char **) calloc(tsLDescriptor->PortCount, sizeof(char *));
+ tsLDescriptor->PortNames = (const char **) port_names;
+
+ port_descriptors[0] = LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO;
+ port_names[0] = "Output L";
+ port_range_hints[0].HintDescriptor = 0;
+ port_descriptors[1] = LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO;
+ port_names[1] = "Output R";
+ port_range_hints[1].HintDescriptor = 0;
+
+ tsLDescriptor->activate = activateTS;
+ tsLDescriptor->cleanup = cleanupTS;
+ tsLDescriptor->connect_port = connectPortTS;
+ tsLDescriptor->deactivate = NULL;
+ tsLDescriptor->instantiate = instantiateTS;
+ tsLDescriptor->run = runTSWrapper;
+ tsLDescriptor->run_adding = NULL;
+ tsLDescriptor->set_run_adding_gain = NULL;
}
tsDDescriptor = (DSSI_Descriptor *) malloc(sizeof(DSSI_Descriptor));
if (tsDDescriptor) {
- tsDDescriptor->DSSI_API_Version = 1;
- tsDDescriptor->LADSPA_Plugin = tsLDescriptor;
- tsDDescriptor->configure = NULL;
- tsDDescriptor->get_program = NULL;
- tsDDescriptor->get_midi_controller_for_port = getControllerTS;
- tsDDescriptor->select_program = NULL;
- tsDDescriptor->run_synth = runTS;
- tsDDescriptor->run_synth_adding = NULL;
- tsDDescriptor->run_multiple_synths = NULL;
- tsDDescriptor->run_multiple_synths_adding = NULL;
+ tsDDescriptor->DSSI_API_Version = 1;
+ tsDDescriptor->LADSPA_Plugin = tsLDescriptor;
+ tsDDescriptor->configure = NULL;
+ tsDDescriptor->get_program = NULL;
+ tsDDescriptor->get_midi_controller_for_port = getControllerTS;
+ tsDDescriptor->select_program = NULL;
+ tsDDescriptor->run_synth = runTS;
+ tsDDescriptor->run_synth_adding = NULL;
+ tsDDescriptor->run_multiple_synths = NULL;
+ tsDDescriptor->run_multiple_synths_adding = NULL;
}
-
+
};
-void _fini(){
+void _fini()
+{
};
diff --git a/src/Output/DSSIaudiooutput.h b/src/Output/DSSIaudiooutput.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
VSTaudiooutput.h - Audio output for VST
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -36,7 +36,7 @@ class VSTSynth:public AudioEffectX{
public:
VSTSynth (audioMasterCallback audioMaster);
~VSTSynth();
-
+
virtual void process (float **inputs, float **outputs, long sampleframes);
virtual void processReplacing (float **inputs, float **outputs, long sampleframes);
virtual long processEvents(VstEvents *events);//this is used for Midi input
@@ -44,9 +44,9 @@ class VSTSynth:public AudioEffectX{
virtual bool getVendorString(char *txt);
virtual bool getProductString(char *txt);
virtual void resume();
-
+
virtual long getChunk(void** data,bool isPreset=false);
- virtual void setChunk(void *data,long size,bool isPreset=false);
+ virtual void setChunk(void *data,long size,bool isPreset=false);
MasterUI *ui;
int Pexitprogram;
diff --git a/src/Output/JACK_RTaudiooutput.C b/src/Output/JACK_RTaudiooutput.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
JACKaudiooutput.C - Audio output for JACK
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,7 +26,8 @@
#include <unistd.h>
-extern "C" {
+extern "C"
+{
#include <jack/ringbuffer.h>
};
#include "JACKaudiooutput.h"
@@ -50,89 +51,92 @@ pthread_mutex_t zyn_thread_lock=PTHREAD_MUTEX_INITIALIZER;
pthread_t bthr;
-bool JACKaudiooutputinit(Master *master_){
+bool JACKaudiooutputinit(Master *master_)
+{
jackmaster=master_;
jackclient=0;
char tmpstr[100];
jackoutl=new REALTYPE [SOUND_BUFFER_SIZE];
jackoutr=new REALTYPE [SOUND_BUFFER_SIZE];
-
+
int rbbufsize=SOUND_BUFFER_SIZE*sizeof (REALTYPE)*2*2;
printf("%d\n",rbbufsize);
rb=jack_ringbuffer_create(rbbufsize);
for (int i=0;i<rbbufsize;i++) rb->buf[i]=0.0;
- for (int i=0;i<15;i++){
- if (i!=0) snprintf(tmpstr,100,"ZynAddSubFX_%d",i);
- else snprintf(tmpstr,100,"ZynAddSubFX");
- jackclient=jack_client_new(tmpstr);
- if (jackclient!=0) break;
+ for (int i=0;i<15;i++) {
+ if (i!=0) snprintf(tmpstr,100,"ZynAddSubFX_%d",i);
+ else snprintf(tmpstr,100,"ZynAddSubFX");
+ jackclient=jack_client_new(tmpstr);
+ if (jackclient!=0) break;
};
if (jackclient==0) {
- fprintf(stderr,"\nERROR: Cannot make a jack client (possible reasons: JACK server is not running or jackd is launched by root and zynaddsubfx by another user.).\n\n\n");
- return(false);
+ fprintf(stderr,"\nERROR: Cannot make a jack client (possible reasons: JACK server is not running or jackd is launched by root and zynaddsubfx by another user.).\n\n\n");
+ return(false);
};
fprintf(stderr,"Internal SampleRate = %d\nJack Output SampleRate= %d\n",SAMPLE_RATE,jack_get_sample_rate(jackclient));
- if ((unsigned int)jack_get_sample_rate(jackclient)!=(unsigned int) SAMPLE_RATE)
- fprintf(stderr,"It is recomanded that the both samplerates to be equal.\n");
-
- jack_set_process_callback(jackclient,jackprocess,0);
- jack_set_sample_rate_callback(jackclient,jacksrate,0);
- jack_on_shutdown(jackclient,jackshutdown,0);
-
+ if ((unsigned int)jack_get_sample_rate(jackclient)!=(unsigned int) SAMPLE_RATE)
+ fprintf(stderr,"It is recomanded that the both samplerates to be equal.\n");
+
+ jack_set_process_callback(jackclient,jackprocess,0);
+ jack_set_sample_rate_callback(jackclient,jacksrate,0);
+ jack_on_shutdown(jackclient,jackshutdown,0);
+
outport_left=jack_port_register(jackclient,"out_1",
- JACK_DEFAULT_AUDIO_TYPE,JackPortIsOutput|JackPortIsTerminal,0);
+ JACK_DEFAULT_AUDIO_TYPE,JackPortIsOutput|JackPortIsTerminal,0);
outport_right=jack_port_register(jackclient,"out_2",
- JACK_DEFAULT_AUDIO_TYPE,JackPortIsOutput|JackPortIsTerminal,0);
+ JACK_DEFAULT_AUDIO_TYPE,JackPortIsOutput|JackPortIsTerminal,0);
- if (jack_activate(jackclient)){
- fprintf(stderr,"Cannot activate jack client\n");
- return(false);
+ if (jack_activate(jackclient)) {
+ fprintf(stderr,"Cannot activate jack client\n");
+ return(false);
};
pthread_create(&bthr,NULL,thread_blocked,NULL);
-
+
/*
jack_connect(jackclient,jack_port_name(outport_left),"alsa_pcm:out_1");
jack_connect(jackclient,jack_port_name(outport_right),"alsa_pcm:out_2");
*/
-
- return(true);
+
+ return(true);
};
-void *thread_blocked(void *arg){
+void *thread_blocked(void *arg)
+{
int datasize=SOUND_BUFFER_SIZE*sizeof (REALTYPE);
//try to get realtime
sched_param sc;
sc.sched_priority=50;
int err=sched_setscheduler(0,SCHED_FIFO,&sc);
-
+
pthread_setcanceltype(PTHREAD_CANCEL_ASYNCHRONOUS,NULL);
pthread_mutex_lock(&zyn_thread_lock);
- while (jackfinish==0){
- while (jack_ringbuffer_write_space(rb)>=datasize){
- pthread_mutex_lock(&jackmaster->mutex);
- jackmaster->GetAudioOutSamples(SOUND_BUFFER_SIZE,jack_get_sample_rate(jackclient),jackoutl,jackoutr);
- pthread_mutex_unlock(&jackmaster->mutex);
-
- jack_ringbuffer_write(rb, (char *) jackoutl,datasize);
- jack_ringbuffer_write(rb, (char *) jackoutr,datasize);
- };
- pthread_cond_wait(&more_data,&zyn_thread_lock);
+ while (jackfinish==0) {
+ while (jack_ringbuffer_write_space(rb)>=datasize) {
+ pthread_mutex_lock(&jackmaster->mutex);
+ jackmaster->GetAudioOutSamples(SOUND_BUFFER_SIZE,jack_get_sample_rate(jackclient),jackoutl,jackoutr);
+ pthread_mutex_unlock(&jackmaster->mutex);
+
+ jack_ringbuffer_write(rb, (char *) jackoutl,datasize);
+ jack_ringbuffer_write(rb, (char *) jackoutr,datasize);
+ };
+ pthread_cond_wait(&more_data,&zyn_thread_lock);
};
pthread_mutex_unlock(&zyn_thread_lock);
-
+
return(0);
};
-int jackprocess(jack_nframes_t nframes,void *arg){
+int jackprocess(jack_nframes_t nframes,void *arg)
+{
jack_default_audio_sample_t *outl=(jack_default_audio_sample_t *) jack_port_get_buffer (outport_left, nframes);
jack_default_audio_sample_t *outr=(jack_default_audio_sample_t *) jack_port_get_buffer (outport_right, nframes);
@@ -141,42 +145,44 @@ int jackprocess(jack_nframes_t nframes,void *arg){
int data_read=0;
- if (jack_ringbuffer_read_space(rb)>=(2*incoming_datasize)){
- if (datasize>incoming_datasize){
- data_read=0;
- while (data_read < datasize){
- jack_ringbuffer_read(rb, (char *) outl+data_read,datasize);
- jack_ringbuffer_read(rb, (char *) outr+data_read,datasize);
- data_read+=incoming_datasize;
- };
- } else if (datasize==incoming_datasize){
- jack_ringbuffer_read(rb, (char *) outl,datasize);
- jack_ringbuffer_read(rb, (char *) outr,datasize);
- } else {
- };
- } else {//the ringbuffer is empty or there are too small amount of samples in it
- for (int i=0;i<nframes;i++){
- outl[i]=0.0;outr[i]=0.0;
- };
- };
-/* if (jack_ringbuffer_read_space(rb)>=datasize){
- jack_ringbuffer_read(rb, (char *) outl,datasize);
- jack_ringbuffer_read(rb, (char *) outr,datasize);
+ if (jack_ringbuffer_read_space(rb)>=(2*incoming_datasize)) {
+ if (datasize>incoming_datasize) {
+ data_read=0;
+ while (data_read < datasize) {
+ jack_ringbuffer_read(rb, (char *) outl+data_read,datasize);
+ jack_ringbuffer_read(rb, (char *) outr+data_read,datasize);
+ data_read+=incoming_datasize;
+ };
+ } else if (datasize==incoming_datasize) {
+ jack_ringbuffer_read(rb, (char *) outl,datasize);
+ jack_ringbuffer_read(rb, (char *) outr,datasize);
+ } else {
+ };
} else {//the ringbuffer is empty or there are too small amount of samples in it
- for (int i=0;i<nframes;i++){
- outl[i]=0.0;outr[i]=0.0;
- };
+ for (int i=0;i<nframes;i++) {
+ outl[i]=0.0;
+ outr[i]=0.0;
+ };
};
-*/
- if (pthread_mutex_trylock(&zyn_thread_lock)==0){
- pthread_cond_signal(&more_data);
- pthread_mutex_unlock(&zyn_thread_lock);
+ /* if (jack_ringbuffer_read_space(rb)>=datasize){
+ jack_ringbuffer_read(rb, (char *) outl,datasize);
+ jack_ringbuffer_read(rb, (char *) outr,datasize);
+ } else {//the ringbuffer is empty or there are too small amount of samples in it
+ for (int i=0;i<nframes;i++){
+ outl[i]=0.0;outr[i]=0.0;
+ };
+ };
+ */
+ if (pthread_mutex_trylock(&zyn_thread_lock)==0) {
+ pthread_cond_signal(&more_data);
+ pthread_mutex_unlock(&zyn_thread_lock);
};
-
+
return(0);
};
-void JACKfinish(){
+void JACKfinish()
+{
jackfinish=1;
jack_ringbuffer_free(rb);
jack_client_close(jackclient);
@@ -186,12 +192,14 @@ void JACKfinish(){
delete(jackoutr);
};
-int jacksrate(jack_nframes_t nframes,void *arg){
-
+int jacksrate(jack_nframes_t nframes,void *arg)
+{
+
return(0);
};
-void jackshutdown(void *arg){
+void jackshutdown(void *arg)
+{
};
diff --git a/src/Output/JACKaudiooutput.C b/src/Output/JACKaudiooutput.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
JACKaudiooutput.C - Audio output for JACK
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -33,125 +33,131 @@ int jackprocess(jack_nframes_t nframes,void *arg);
int jacksrate(jack_nframes_t nframes,void *arg);
void jackshutdown(void *arg);
-bool JACKaudiooutputinit(Master *master_){
+bool JACKaudiooutputinit(Master *master_)
+{
jackmaster=master_;
jackclient=0;
- for (int i=0;i<15;i++){
- if (i!=0) snprintf(jackname,100,"ZynAddSubFX_%d",i);
- else snprintf(jackname,100,"ZynAddSubFX");
- jackclient=jack_client_new(jackname);
- if (jackclient!=0) break;
+ for (int i=0;i<15;i++) {
+ if (i!=0) snprintf(jackname,100,"ZynAddSubFX_%d",i);
+ else snprintf(jackname,100,"ZynAddSubFX");
+ jackclient=jack_client_new(jackname);
+ if (jackclient!=0) break;
};
if (jackclient==0) {
- fprintf(stderr,"\nERROR: Cannot make a jack client (possible reasons: JACK server is not running or jackd is launched by root and zynaddsubfx by another user.).\n");
- return(false);
+ fprintf(stderr,"\nERROR: Cannot make a jack client (possible reasons: JACK server is not running or jackd is launched by root and zynaddsubfx by another user.).\n");
+ return(false);
};
fprintf(stderr,"Internal SampleRate = %d\nJack Output SampleRate= %d\n",SAMPLE_RATE,jack_get_sample_rate(jackclient));
- if ((unsigned int)jack_get_sample_rate(jackclient)!=(unsigned int) SAMPLE_RATE)
- fprintf(stderr,"It is recomanded that the both samplerates to be equal.\n");
-
- jack_set_process_callback(jackclient,jackprocess,0);
- jack_set_sample_rate_callback(jackclient,jacksrate,0);
- jack_on_shutdown(jackclient,jackshutdown,0);
-
+ if ((unsigned int)jack_get_sample_rate(jackclient)!=(unsigned int) SAMPLE_RATE)
+ fprintf(stderr,"It is recomanded that the both samplerates to be equal.\n");
+
+ jack_set_process_callback(jackclient,jackprocess,0);
+ jack_set_sample_rate_callback(jackclient,jacksrate,0);
+ jack_on_shutdown(jackclient,jackshutdown,0);
+
outport_left=jack_port_register(jackclient,"out_1",
- JACK_DEFAULT_AUDIO_TYPE,JackPortIsOutput|JackPortIsTerminal,0);
+ JACK_DEFAULT_AUDIO_TYPE,JackPortIsOutput|JackPortIsTerminal,0);
outport_right=jack_port_register(jackclient,"out_2",
- JACK_DEFAULT_AUDIO_TYPE,JackPortIsOutput|JackPortIsTerminal,0);
+ JACK_DEFAULT_AUDIO_TYPE,JackPortIsOutput|JackPortIsTerminal,0);
midi_inport=jack_port_register(jackclient,"midi_input",
- JACK_DEFAULT_MIDI_TYPE,JackPortIsInput|JackPortIsTerminal,0);
+ JACK_DEFAULT_MIDI_TYPE,JackPortIsInput|JackPortIsTerminal,0);
- if (jack_activate(jackclient)){
- fprintf(stderr,"Cannot activate jack client\n");
- return(false);
+ if (jack_activate(jackclient)) {
+ fprintf(stderr,"Cannot activate jack client\n");
+ return(false);
};
-
+
/*
jack_connect(jackclient,jack_port_name(outport_left),"alsa_pcm:out_1");
jack_connect(jackclient,jack_port_name(outport_right),"alsa_pcm:out_2");
*/
- return(true);
+ return(true);
};
-int jackprocess(jack_nframes_t nframes,void *arg){
+int jackprocess(jack_nframes_t nframes,void *arg)
+{
jack_default_audio_sample_t *outl=(jack_default_audio_sample_t *) jack_port_get_buffer (outport_left, nframes);
jack_default_audio_sample_t *outr=(jack_default_audio_sample_t *) jack_port_get_buffer (outport_right, nframes);
if (!pthread_mutex_trylock(&jackmaster->mutex)) {
- JACKhandlemidi(nframes);
- jackmaster->GetAudioOutSamples(nframes,jack_get_sample_rate(jackclient),outl,outr);
- pthread_mutex_unlock(&jackmaster->mutex);
- }
- else {
- memset(outl, 0, sizeof(jack_default_audio_sample_t) * nframes);
- memset(outr, 0, sizeof(jack_default_audio_sample_t) * nframes);
+ JACKhandlemidi(nframes);
+ jackmaster->GetAudioOutSamples(nframes,jack_get_sample_rate(jackclient),outl,outr);
+ pthread_mutex_unlock(&jackmaster->mutex);
+ } else {
+ memset(outl, 0, sizeof(jack_default_audio_sample_t) * nframes);
+ memset(outr, 0, sizeof(jack_default_audio_sample_t) * nframes);
}
-
+
return(0);
};
-void JACKfinish(){
+void JACKfinish()
+{
jack_client_close(jackclient);
};
-int jacksrate(jack_nframes_t nframes,void *arg){
-
+int jacksrate(jack_nframes_t nframes,void *arg)
+{
+
return(0);
};
-void jackshutdown(void *arg){
+void jackshutdown(void *arg)
+{
};
-void JACKhandlemidi(unsigned long frames) {
-
- // We must have the master mutex before we run this function
-
- // XXX This is really nasty, not only do we lose the sample accuracy of
- // JACK MIDI, but any accuracy at all below the buffer size
-
- void* midi_buf = jack_port_get_buffer(midi_inport, frames);
- jack_midi_event_t jack_midi_event;
- jack_nframes_t event_index = 0;
- unsigned char* midi_data;
- unsigned char type, chan;
-
- while (jack_midi_event_get(&jack_midi_event,midi_buf, event_index++) == 0) {
- midi_data = jack_midi_event.buffer;
- type = midi_data[0] & 0xF0;
- chan = midi_data[0] & 0x0F;
-
- switch (type) {
-
- case 0x80: /* note-off */
- jackmaster->NoteOff(chan, midi_data[1]);
- break;
-
- case 0x90: /* note-on */
- jackmaster->NoteOn(chan, midi_data[1], midi_data[2]);
- break;
-
- case 0xB0: /* controller */
- jackmaster->SetController(chan, midi_data[1], midi_data[2]);
- break;
-
- case 0xE0: /* pitch bend */
- jackmaster->SetController(chan, C_pitchwheel,
- ((midi_data[2] << 7) | midi_data[1]));
- break;
-
- /* XXX TODO: handle MSB/LSB controllers and RPNs and NRPNs */
- }
- }
-
+void JACKhandlemidi(unsigned long frames)
+{
+
+ // We must have the master mutex before we run this function
+
+ // XXX This is really nasty, not only do we lose the sample accuracy of
+ // JACK MIDI, but any accuracy at all below the buffer size
+
+ void* midi_buf = jack_port_get_buffer(midi_inport, frames);
+ jack_midi_event_t jack_midi_event;
+ jack_nframes_t event_index = 0;
+ unsigned char* midi_data;
+ unsigned char type, chan;
+
+ while (jack_midi_event_get(&jack_midi_event,midi_buf, event_index++) == 0) {
+ midi_data = jack_midi_event.buffer;
+ type = midi_data[0] & 0xF0;
+ chan = midi_data[0] & 0x0F;
+
+ switch (type) {
+
+ case 0x80: /* note-off */
+ jackmaster->NoteOff(chan, midi_data[1]);
+ break;
+
+ case 0x90: /* note-on */
+ jackmaster->NoteOn(chan, midi_data[1], midi_data[2]);
+ break;
+
+ case 0xB0: /* controller */
+ jackmaster->SetController(chan, midi_data[1], midi_data[2]);
+ break;
+
+ case 0xE0: /* pitch bend */
+ jackmaster->SetController(chan, C_pitchwheel,
+ ((midi_data[2] << 7) | midi_data[1]));
+ break;
+
+ /* XXX TODO: handle MSB/LSB controllers and RPNs and NRPNs */
+ }
+ }
+
}
-const char* JACKgetname() {
- if (jackclient != NULL)
- return jackname;
- return NULL;
+const char* JACKgetname()
+{
+ if (jackclient != NULL)
+ return jackname;
+ return NULL;
}
diff --git a/src/Output/JACKaudiooutput.h b/src/Output/JACKaudiooutput.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
JACKaudiooutput.h - Audio output for JACK
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
diff --git a/src/Output/OSSaudiooutput.C b/src/Output/OSSaudiooutput.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
OSSaudiooutput.C - Audio output for Open Sound System
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -34,22 +34,23 @@
#include "../globals.h"
using namespace std;
-OSSaudiooutput::OSSaudiooutput(){
+OSSaudiooutput::OSSaudiooutput()
+{
int i;
int snd_bitsize=16;
snd_fragment=0x00080009;//fragment size (?)
snd_stereo=1;//stereo
snd_format=AFMT_S16_LE;
snd_samplerate=SAMPLE_RATE;
-
+
smps=new short int[SOUND_BUFFER_SIZE*2];
for (i=0;i<SOUND_BUFFER_SIZE*2;i++) smps[i]=0;
-
+
snd_handle=open(config.cfg.LinuxOSSWaveOutDev,O_WRONLY,0);
if (snd_handle == -1) {
cerr << "ERROR - I can't open the ";
cerr << config.cfg.LinuxOSSWaveOutDev << '.'<< endl;
- return;
+ return;
};
ioctl(snd_handle,SNDCTL_DSP_RESET,NULL);
@@ -58,7 +59,7 @@ OSSaudiooutput::OSSaudiooutput(){
ioctl(snd_handle,SNDCTL_DSP_SPEED,&snd_samplerate);
ioctl(snd_handle,SNDCTL_DSP_SAMPLESIZE,&snd_bitsize);
ioctl(snd_handle,SNDCTL_DSP_SETFRAGMENT,&snd_fragment);
-
+
};
@@ -66,25 +67,29 @@ OSSaudiooutput::OSSaudiooutput(){
* Output the samples to the soundcard
* The samples are bigger than -1.0 and smaller 1.0
*/
-void OSSaudiooutput::OSSout(REALTYPE *smp_left,REALTYPE *smp_right){
+void OSSaudiooutput::OSSout(REALTYPE *smp_left,REALTYPE *smp_right)
+{
int i;
REALTYPE l,r;
if (snd_handle<0) return;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- l=smp_left[i];
- r=smp_right[i];
-
- if (l<-1.0) l=-1.0; else if (l>1.0) l=1.0;
- if (r<-1.0) r=-1.0; else if (r>1.0) r=1.0;
-
- smps[i*2]=(short int) (l*32767.0);
- smps[i*2+1]=(short int) (r*32767.0);
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ l=smp_left[i];
+ r=smp_right[i];
+
+ if (l<-1.0) l=-1.0;
+ else if (l>1.0) l=1.0;
+ if (r<-1.0) r=-1.0;
+ else if (r>1.0) r=1.0;
+
+ smps[i*2]=(short int) (l*32767.0);
+ smps[i*2+1]=(short int) (r*32767.0);
};
write(snd_handle,smps,SOUND_BUFFER_SIZE*4);// *2 because is 16 bit, again * 2 because is stereo
};
-OSSaudiooutput::~OSSaudiooutput(){
+OSSaudiooutput::~OSSaudiooutput()
+{
close(snd_handle);
delete [] smps;
};
diff --git a/src/Output/OSSaudiooutput.h b/src/Output/OSSaudiooutput.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
OSSaudiooutput.h - Audio output for Open Sound System
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,22 +25,23 @@
#include "../globals.h"
-class OSSaudiooutput{
- public:
- OSSaudiooutput();
- ~OSSaudiooutput();
-
- //the out is [-1.0 .. 1.0]
- /* smp_left[] and smp_right[] has the size of SOUND_BUFFER_SIZE */
- void OSSout(REALTYPE *smp_left,REALTYPE *smp_right);
- private:
- int snd_handle;
- int snd_fragment;
- int snd_stereo;
- int snd_format;
- int snd_samplerate;
-
- short int *smps;//Samples to be sent to soundcard
+class OSSaudiooutput
+{
+public:
+ OSSaudiooutput();
+ ~OSSaudiooutput();
+
+ //the out is [-1.0 .. 1.0]
+ /* smp_left[] and smp_right[] has the size of SOUND_BUFFER_SIZE */
+ void OSSout(REALTYPE *smp_left,REALTYPE *smp_right);
+private:
+ int snd_handle;
+ int snd_fragment;
+ int snd_stereo;
+ int snd_format;
+ int snd_samplerate;
+
+ short int *smps;//Samples to be sent to soundcard
};
#endif
diff --git a/src/Output/PAaudiooutput.C b/src/Output/PAaudiooutput.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
PAaudiooutput.C - Audio output for PortAudio
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,30 +27,32 @@ PaStream *stream;
REALTYPE *outl,*outr;
int PAprocess(void *inputBuffer,void *outputBuffer,
- unsigned long framesPerBuffer,
- PaTimestamp outTime,void *userData){
+ unsigned long framesPerBuffer,
+ PaTimestamp outTime,void *userData)
+{
if (framesPerBuffer!=SOUND_BUFFER_SIZE) {
- fprintf(stderr,"Bug: PAudioOutput::PAprocess SOUND_BUFFER_SIZE!=framesPerBuffer");
+ fprintf(stderr,"Bug: PAudioOutput::PAprocess SOUND_BUFFER_SIZE!=framesPerBuffer");
fprintf(stderr,"%d %d\n",framesPerBuffer,SOUND_BUFFER_SIZE);
};
pthread_mutex_lock(&PAmaster->mutex);
- PAmaster->GetAudioOutSamples(SOUND_BUFFER_SIZE,SAMPLE_RATE,outl,outr);
+ PAmaster->GetAudioOutSamples(SOUND_BUFFER_SIZE,SAMPLE_RATE,outl,outr);
pthread_mutex_unlock(&PAmaster->mutex);
float *out=(float *)outputBuffer;
- for (int i=0;i<framesPerBuffer;i++){
- if (i>=SOUND_BUFFER_SIZE) break;//this should never happens, except only when framesPerBuffer!>SOUND_BUFFER_SIZE
- out[i*2]=outl[i];
- out[i*2+1]=outr[i];
+ for (int i=0;i<framesPerBuffer;i++) {
+ if (i>=SOUND_BUFFER_SIZE) break;//this should never happens, except only when framesPerBuffer!>SOUND_BUFFER_SIZE
+ out[i*2]=outl[i];
+ out[i*2+1]=outr[i];
};
return(0);
};
-void PAaudiooutputinit(Master *master_){
+void PAaudiooutputinit(Master *master_)
+{
PAmaster=master_;
outl=new REALTYPE [SOUND_BUFFER_SIZE];
outr=new REALTYPE [SOUND_BUFFER_SIZE];
@@ -59,7 +61,8 @@ void PAaudiooutputinit(Master *master_){
Pa_StartStream(stream);
};
-void PAfinish(){
+void PAfinish()
+{
Pa_StopStream(stream);
delete (outl);
delete (outr);
diff --git a/src/Output/PAaudiooutput.h b/src/Output/PAaudiooutput.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
PAaudiooutput.h - Audio output for PortAudio
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
diff --git a/src/Output/Recorder.C b/src/Output/Recorder.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Recorder.C - Records sound to a file
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -23,60 +23,68 @@
#include <sys/stat.h>
#include "Recorder.h"
-Recorder::Recorder(){
+Recorder::Recorder()
+{
recordbuf_16bit=new short int [SOUND_BUFFER_SIZE*2];
status=0;
notetrigger=0;
- for (int i=0;i<SOUND_BUFFER_SIZE*2;i++){
+ for (int i=0;i<SOUND_BUFFER_SIZE*2;i++) {
recordbuf_16bit[i]=0;
}
};
-Recorder::~Recorder(){
+Recorder::~Recorder()
+{
if (recording()==1) stop();
delete [] recordbuf_16bit;
};
-int Recorder::preparefile(std::string filename_,int overwrite){
- if (!overwrite){
+int Recorder::preparefile(std::string filename_,int overwrite)
+{
+ if (!overwrite) {
struct stat fileinfo;
int statr;
statr = stat(filename_.c_str(),&fileinfo);
- if(statr == 0) {//file exists
+ if (statr == 0) {//file exists
return 1;
}
}
-
+
if (!wav.newfile(filename_, SAMPLE_RATE,2)) return 2;
-
+
status=1;//ready
-
+
return(0);
};
-void Recorder::start(){
+void Recorder::start()
+{
notetrigger=0;
status=2;//recording
};
-void Recorder::stop(){
- wav.close();
+void Recorder::stop()
+{
+ wav.close();
status=0;
};
-void Recorder::pause(){
+void Recorder::pause()
+{
status=0;
};
-int Recorder::recording(){
+int Recorder::recording()
+{
if ((status==2)&&(notetrigger!=0)) return(1);
else return(0);
};
-void Recorder::recordbuffer(REALTYPE *outl,REALTYPE *outr){
+void Recorder::recordbuffer(REALTYPE *outl,REALTYPE *outr)
+{
int tmp;
if (status!=2) return;
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
tmp=(int)(outl[i]*32767.0);
if (tmp<-32768) tmp=-32768;
if (tmp>32767) tmp=32767;
@@ -90,6 +98,7 @@ void Recorder::recordbuffer(REALTYPE *outl,REALTYPE *outr){
wav.write_stereo_samples(SOUND_BUFFER_SIZE,recordbuf_16bit);
};
-void Recorder::triggernow(){
+void Recorder::triggernow()
+{
if (status==2) notetrigger=1;
};
diff --git a/src/Output/Recorder.h b/src/Output/Recorder.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Recorder.h - Records sound to a file
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,29 +27,30 @@
#include "WAVaudiooutput.h"
/**Records sound to a file*/
-class Recorder{
- public:
-
- Recorder();
- ~Recorder();
- int preparefile(std::string filename_,int overwrite);//returns 1 if the file exists
- void start();
- void stop();
- void pause();
- int recording();
- void triggernow();
- void recordbuffer(REALTYPE *outl,REALTYPE *outr);
-
- /** Status:
- * 0 - not ready(no file selected),
- * 1 - ready
- * 2 - recording */
- int status;
-
- private:
- WAVaudiooutput wav;
- short int *recordbuf_16bit;
- int notetrigger;
+class Recorder
+{
+public:
+
+ Recorder();
+ ~Recorder();
+ int preparefile(std::string filename_,int overwrite);//returns 1 if the file exists
+ void start();
+ void stop();
+ void pause();
+ int recording();
+ void triggernow();
+ void recordbuffer(REALTYPE *outl,REALTYPE *outr);
+
+ /** Status:
+ * 0 - not ready(no file selected),
+ * 1 - ready
+ * 2 - recording */
+ int status;
+
+private:
+ WAVaudiooutput wav;
+ short int *recordbuf_16bit;
+ int notetrigger;
};
#endif
diff --git a/src/Output/VSTaudiooutput.C b/src/Output/VSTaudiooutput.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
VSTaudiooutput.C - Audio output for VST
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,36 +24,42 @@
//the constructor and the destructor are defined in main.C
-void VSTSynth::process (float **inputs, float **outputs, long sampleframes){
+void VSTSynth::process (float **inputs, float **outputs, long sampleframes)
+{
float *outl=outputs[0];
float *outr=outputs[1];
pthread_mutex_lock(&vmaster->mutex);
- vmaster->GetAudioOutSamples(sampleframes,(int) getSampleRate(),outl,outr);
+ vmaster->GetAudioOutSamples(sampleframes,(int) getSampleRate(),outl,outr);
pthread_mutex_unlock(&vmaster->mutex);
};
-void VSTSynth::processReplacing (float **inputs, float **outputs, long sampleframes){
+void VSTSynth::processReplacing (float **inputs, float **outputs, long sampleframes)
+{
process(inputs,outputs,sampleframes);
};
-long int VSTSynth::canDo(char *txt){
- if (strcmp(txt,"receiveVstEvents")==0) return (1);
- if (strcmp(txt,"receiveVstMidiEvent")==0) return (1);
- return(-1);
+long int VSTSynth::canDo(char *txt)
+{
+ if (strcmp(txt,"receiveVstEvents")==0) return (1);
+ if (strcmp(txt,"receiveVstMidiEvent")==0) return (1);
+ return(-1);
};
-bool VSTSynth::getVendorString(char *txt){
- strcpy(txt,"Nasca O. Paul");
- return(true);
+bool VSTSynth::getVendorString(char *txt)
+{
+ strcpy(txt,"Nasca O. Paul");
+ return(true);
};
-bool VSTSynth::getProductString(char *txt){
- strcpy(txt,"ZynAddSubFX");
- return(true);
+bool VSTSynth::getProductString(char *txt)
+{
+ strcpy(txt,"ZynAddSubFX");
+ return(true);
};
-void VSTSynth::resume(){
- wantEvents();
+void VSTSynth::resume()
+{
+ wantEvents();
};
diff --git a/src/Output/VSTaudiooutput.h b/src/Output/VSTaudiooutput.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
VSTaudiooutput.h - Audio output for VST
Copyright (C) 2002 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -30,27 +30,28 @@
#include "../../../vstsdk2/source/common/audioeffectx.h"
-class VSTSynth:public AudioEffectX{
- public:
- VSTSynth (audioMasterCallback audioMaster);
- ~VSTSynth();
-
- virtual void process (float **inputs, float **outputs, long sampleframes);
- virtual void processReplacing (float **inputs, float **outputs, long sampleframes);
- virtual long processEvents(VstEvents *events);//this is used for Midi input
- virtual long int canDo(char *txt);
- virtual bool getVendorString(char *txt);
- virtual bool getProductString(char *txt);
- virtual void resume();
-
- virtual long getChunk(void** data,bool isPreset=false);
- virtual long setChunk(void *data,long size,bool isPreset=false);
-
- MasterUI *ui;
- int Pexitprogram;
-
- Master *vmaster;
- pthread_t thr;
+class VSTSynth:public AudioEffectX
+{
+public:
+ VSTSynth (audioMasterCallback audioMaster);
+ ~VSTSynth();
+
+ virtual void process (float **inputs, float **outputs, long sampleframes);
+ virtual void processReplacing (float **inputs, float **outputs, long sampleframes);
+ virtual long processEvents(VstEvents *events);//this is used for Midi input
+ virtual long int canDo(char *txt);
+ virtual bool getVendorString(char *txt);
+ virtual bool getProductString(char *txt);
+ virtual void resume();
+
+ virtual long getChunk(void** data,bool isPreset=false);
+ virtual long setChunk(void *data,long size,bool isPreset=false);
+
+ MasterUI *ui;
+ int Pexitprogram;
+
+ Master *vmaster;
+ pthread_t thr;
};
#endif
diff --git a/src/Output/WAVaudiooutput.C b/src/Output/WAVaudiooutput.C
@@ -3,7 +3,7 @@
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -21,18 +21,21 @@
#include "WAVaudiooutput.h"
using namespace std;
-WAVaudiooutput::WAVaudiooutput(){
+WAVaudiooutput::WAVaudiooutput()
+{
file=NULL;
sampleswritten=0;
samplerate=44100;
};
-WAVaudiooutput::~WAVaudiooutput(){
+WAVaudiooutput::~WAVaudiooutput()
+{
close();
};
-bool WAVaudiooutput::newfile(string filename,int samplerate,int channels){
- /**\todo Move this into the Constructor*/
+bool WAVaudiooutput::newfile(string filename,int samplerate,int channels)
+{
+ /**\todo Move this into the Constructor*/
close();//inchide un posibil fisier existent
file=fopen(filename.c_str(),"w");
if (!file) return false;
@@ -44,14 +47,15 @@ bool WAVaudiooutput::newfile(string filename,int samplerate,int channels){
return(true);
};
-void WAVaudiooutput::close(){
- if (file){
+void WAVaudiooutput::close()
+{
+ if (file) {
unsigned int chunksize;
rewind(file);
fwrite("RIFF",4,1,file);
chunksize=sampleswritten*4+36;
- fwrite(&chunksize,4,1,file);
+ fwrite(&chunksize,4,1,file);
fwrite("WAVEfmt ",8,1,file);
chunksize=16;
@@ -68,7 +72,7 @@ void WAVaudiooutput::close(){
fwrite(&blockalign,2,1,file);
unsigned short int bitspersample=16;
fwrite(&bitspersample,2,1,file);
-
+
fwrite("data",4,1,file);
chunksize=sampleswritten*blockalign;
fwrite(&chunksize,4,1,file);
@@ -78,13 +82,15 @@ void WAVaudiooutput::close(){
}
};
-void WAVaudiooutput::write_stereo_samples(int nsmps,short int *smps){
+void WAVaudiooutput::write_stereo_samples(int nsmps,short int *smps)
+{
if (!file) return;
fwrite(smps,nsmps,4,file);
sampleswritten+=nsmps;
};
-void WAVaudiooutput::write_mono_samples(int nsmps,short int *smps){
+void WAVaudiooutput::write_mono_samples(int nsmps,short int *smps)
+{
if (!file) return;
fwrite(smps,nsmps,2,file);
sampleswritten+=nsmps;
diff --git a/src/Output/WAVaudiooutput.h b/src/Output/WAVaudiooutput.h
@@ -4,7 +4,7 @@
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -21,21 +21,22 @@
#define WAVOUTPUT_H
#include <string>
-class WAVaudiooutput{
- public:
- WAVaudiooutput();
- ~WAVaudiooutput();
-
- bool newfile(std::string filename,int samplerate,int channels);
- void close();
-
- void write_mono_samples(int nsmps, short int *smps);
- void write_stereo_samples(int nsmps, short int *smps);
-
- private:
- int sampleswritten;
- int samplerate;
- int channels;
- FILE *file;
+class WAVaudiooutput
+{
+public:
+ WAVaudiooutput();
+ ~WAVaudiooutput();
+
+ bool newfile(std::string filename,int samplerate,int channels);
+ void close();
+
+ void write_mono_samples(int nsmps, short int *smps);
+ void write_stereo_samples(int nsmps, short int *smps);
+
+private:
+ int sampleswritten;
+ int samplerate;
+ int channels;
+ FILE *file;
};
#endif
diff --git a/src/Params/ADnoteParameters.C b/src/Params/ADnoteParameters.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
ADnoteParameters.C - Parameters for ADnote (ADsynth)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,14 +26,15 @@
#include "ADnoteParameters.h"
-ADnoteParameters::ADnoteParameters(FFTwrapper *fft_):Presets(){
+ADnoteParameters::ADnoteParameters(FFTwrapper *fft_):Presets()
+{
setpresettype("Padsyth");
fft=fft_;
GlobalPar.FreqEnvelope=new EnvelopeParams(0,0);
GlobalPar.FreqEnvelope->ASRinit(64,50,64,60);
GlobalPar.FreqLfo=new LFOParams(70,0,64,0,0,0,0,0);
-
+
GlobalPar.AmpEnvelope=new EnvelopeParams(64,1);
GlobalPar.AmpEnvelope->ADSRinit_dB(0,40,127,25);
GlobalPar.AmpLfo=new LFOParams(80,0,64,0,0,0,0,1);
@@ -45,11 +46,12 @@ ADnoteParameters::ADnoteParameters(FFTwrapper *fft_):Presets(){
GlobalPar.Reson=new Resonance();
for (int nvoice=0;nvoice<NUM_VOICES;nvoice++) EnableVoice(nvoice);
-
+
defaults();
};
-void ADnoteParameters::defaults(){
+void ADnoteParameters::defaults()
+{
//Default Parameters
/* Frequency Global Parameters */
GlobalPar.PStereo=1;//stereo
@@ -59,7 +61,7 @@ void ADnoteParameters::defaults(){
GlobalPar.FreqEnvelope->defaults();
GlobalPar.FreqLfo->defaults();
GlobalPar.PBandwidth=64;
-
+
/* Amplitude Global Parameters */
GlobalPar.PVolume=90;
GlobalPar.PPanning=64;//center
@@ -71,7 +73,7 @@ void ADnoteParameters::defaults(){
GlobalPar.PPunchStretch=64;
GlobalPar.PPunchVelocitySensing=72;
GlobalPar.Hrandgrouping=0;
-
+
/* Filter Global Parameters*/
GlobalPar.PFilterVelocityScale=64;
GlobalPar.PFilterVelocityScaleFunction=64;
@@ -81,8 +83,8 @@ void ADnoteParameters::defaults(){
GlobalPar.Reson->defaults();
- for (int nvoice=0;nvoice<NUM_VOICES;nvoice++){
- defaults(nvoice);
+ for (int nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ defaults(nvoice);
};
VoicePar[0].Enabled=1;
};
@@ -90,62 +92,63 @@ void ADnoteParameters::defaults(){
/*
* Defaults a voice
*/
-void ADnoteParameters::defaults(int n){
- int nvoice=n;
- VoicePar[nvoice].Enabled=0;
- VoicePar[nvoice].Type=0;
- VoicePar[nvoice].Pfixedfreq=0;
- VoicePar[nvoice].PfixedfreqET=0;
- VoicePar[nvoice].Presonance=1;
- VoicePar[nvoice].Pfilterbypass=0;
- VoicePar[nvoice].Pextoscil=-1;
- VoicePar[nvoice].PextFMoscil=-1;
- VoicePar[nvoice].Poscilphase=64;
- VoicePar[nvoice].PFMoscilphase=64;
- VoicePar[nvoice].PDelay=0;
- VoicePar[nvoice].PVolume=100;
- VoicePar[nvoice].PVolumeminus=0;
- VoicePar[nvoice].PPanning=64;//center
- VoicePar[nvoice].PDetune=8192;//8192=0
- VoicePar[nvoice].PCoarseDetune=0;
- VoicePar[nvoice].PDetuneType=0;
- VoicePar[nvoice].PFreqLfoEnabled=0;
- VoicePar[nvoice].PFreqEnvelopeEnabled=0;
- VoicePar[nvoice].PAmpEnvelopeEnabled=0;
- VoicePar[nvoice].PAmpLfoEnabled=0;
- VoicePar[nvoice].PAmpVelocityScaleFunction=127;
- VoicePar[nvoice].PFilterEnabled=0;
- VoicePar[nvoice].PFilterEnvelopeEnabled=0;
- VoicePar[nvoice].PFilterLfoEnabled=0;
- VoicePar[nvoice].PFMEnabled=0;
-
- //I use the internal oscillator (-1)
- VoicePar[nvoice].PFMVoice=-1;
-
- VoicePar[nvoice].PFMVolume=90;
- VoicePar[nvoice].PFMVolumeDamp=64;
- VoicePar[nvoice].PFMDetune=8192;
- VoicePar[nvoice].PFMCoarseDetune=0;
- VoicePar[nvoice].PFMDetuneType=0;
- VoicePar[nvoice].PFMFreqEnvelopeEnabled=0;
- VoicePar[nvoice].PFMAmpEnvelopeEnabled=0;
- VoicePar[nvoice].PFMVelocityScaleFunction=64;
-
- VoicePar[nvoice].OscilSmp->defaults();
- VoicePar[nvoice].FMSmp->defaults();
-
- VoicePar[nvoice].AmpEnvelope->defaults();
- VoicePar[nvoice].AmpLfo->defaults();
-
- VoicePar[nvoice].FreqEnvelope->defaults();
- VoicePar[nvoice].FreqLfo->defaults();
-
- VoicePar[nvoice].VoiceFilter->defaults();
- VoicePar[nvoice].FilterEnvelope->defaults();
- VoicePar[nvoice].FilterLfo->defaults();
-
- VoicePar[nvoice].FMFreqEnvelope->defaults();
- VoicePar[nvoice].FMAmpEnvelope->defaults();
+void ADnoteParameters::defaults(int n)
+{
+ int nvoice=n;
+ VoicePar[nvoice].Enabled=0;
+ VoicePar[nvoice].Type=0;
+ VoicePar[nvoice].Pfixedfreq=0;
+ VoicePar[nvoice].PfixedfreqET=0;
+ VoicePar[nvoice].Presonance=1;
+ VoicePar[nvoice].Pfilterbypass=0;
+ VoicePar[nvoice].Pextoscil=-1;
+ VoicePar[nvoice].PextFMoscil=-1;
+ VoicePar[nvoice].Poscilphase=64;
+ VoicePar[nvoice].PFMoscilphase=64;
+ VoicePar[nvoice].PDelay=0;
+ VoicePar[nvoice].PVolume=100;
+ VoicePar[nvoice].PVolumeminus=0;
+ VoicePar[nvoice].PPanning=64;//center
+ VoicePar[nvoice].PDetune=8192;//8192=0
+ VoicePar[nvoice].PCoarseDetune=0;
+ VoicePar[nvoice].PDetuneType=0;
+ VoicePar[nvoice].PFreqLfoEnabled=0;
+ VoicePar[nvoice].PFreqEnvelopeEnabled=0;
+ VoicePar[nvoice].PAmpEnvelopeEnabled=0;
+ VoicePar[nvoice].PAmpLfoEnabled=0;
+ VoicePar[nvoice].PAmpVelocityScaleFunction=127;
+ VoicePar[nvoice].PFilterEnabled=0;
+ VoicePar[nvoice].PFilterEnvelopeEnabled=0;
+ VoicePar[nvoice].PFilterLfoEnabled=0;
+ VoicePar[nvoice].PFMEnabled=0;
+
+ //I use the internal oscillator (-1)
+ VoicePar[nvoice].PFMVoice=-1;
+
+ VoicePar[nvoice].PFMVolume=90;
+ VoicePar[nvoice].PFMVolumeDamp=64;
+ VoicePar[nvoice].PFMDetune=8192;
+ VoicePar[nvoice].PFMCoarseDetune=0;
+ VoicePar[nvoice].PFMDetuneType=0;
+ VoicePar[nvoice].PFMFreqEnvelopeEnabled=0;
+ VoicePar[nvoice].PFMAmpEnvelopeEnabled=0;
+ VoicePar[nvoice].PFMVelocityScaleFunction=64;
+
+ VoicePar[nvoice].OscilSmp->defaults();
+ VoicePar[nvoice].FMSmp->defaults();
+
+ VoicePar[nvoice].AmpEnvelope->defaults();
+ VoicePar[nvoice].AmpLfo->defaults();
+
+ VoicePar[nvoice].FreqEnvelope->defaults();
+ VoicePar[nvoice].FreqLfo->defaults();
+
+ VoicePar[nvoice].VoiceFilter->defaults();
+ VoicePar[nvoice].FilterEnvelope->defaults();
+ VoicePar[nvoice].FilterLfo->defaults();
+
+ VoicePar[nvoice].FMFreqEnvelope->defaults();
+ VoicePar[nvoice].FMAmpEnvelope->defaults();
};
@@ -153,12 +156,13 @@ void ADnoteParameters::defaults(int n){
/*
* Init the voice parameters
*/
-void ADnoteParameters::EnableVoice(int nvoice){
+void ADnoteParameters::EnableVoice(int nvoice)
+{
VoicePar[nvoice].OscilSmp=new OscilGen(fft,GlobalPar.Reson);
VoicePar[nvoice].FMSmp=new OscilGen(fft,NULL);
VoicePar[nvoice].AmpEnvelope=new EnvelopeParams(64,1);
- VoicePar[nvoice].AmpEnvelope->ADSRinit_dB(0,100,127,100);
+ VoicePar[nvoice].AmpEnvelope->ADSRinit_dB(0,100,127,100);
VoicePar[nvoice].AmpLfo=new LFOParams(90,32,64,0,0,30,0,1);
VoicePar[nvoice].FreqEnvelope=new EnvelopeParams(0,0);
@@ -179,10 +183,11 @@ void ADnoteParameters::EnableVoice(int nvoice){
/*
* Get the Multiplier of the fine detunes of the voices
*/
-REALTYPE ADnoteParameters::getBandwidthDetuneMultiplier(){
+REALTYPE ADnoteParameters::getBandwidthDetuneMultiplier()
+{
REALTYPE bw=(GlobalPar.PBandwidth-64.0)/64.0;
bw=pow(2.0,bw*pow(fabs(bw),0.2)*5.0);
-
+
return(bw);
};
@@ -190,7 +195,8 @@ REALTYPE ADnoteParameters::getBandwidthDetuneMultiplier(){
/*
* Kill the voice
*/
-void ADnoteParameters::KillVoice(int nvoice){
+void ADnoteParameters::KillVoice(int nvoice)
+{
delete (VoicePar[nvoice].OscilSmp);
delete (VoicePar[nvoice].FMSmp);
@@ -208,7 +214,8 @@ void ADnoteParameters::KillVoice(int nvoice){
delete (VoicePar[nvoice].FMAmpEnvelope);
};
-ADnoteParameters::~ADnoteParameters(){
+ADnoteParameters::~ADnoteParameters()
+{
delete(GlobalPar.FreqEnvelope);
delete(GlobalPar.FreqLfo);
delete(GlobalPar.AmpEnvelope);
@@ -218,7 +225,7 @@ ADnoteParameters::~ADnoteParameters(){
delete(GlobalPar.FilterLfo);
delete(GlobalPar.Reson);
- for (int nvoice=0;nvoice<NUM_VOICES;nvoice++){
+ for (int nvoice=0;nvoice<NUM_VOICES;nvoice++) {
KillVoice(nvoice);
};
};
@@ -226,24 +233,25 @@ ADnoteParameters::~ADnoteParameters(){
-void ADnoteParameters::add2XMLsection(XMLwrapper *xml,int n){
+void ADnoteParameters::add2XMLsection(XMLwrapper *xml,int n)
+{
int nvoice=n;
if (nvoice>=NUM_VOICES) return;
-
+
int oscilused=0,fmoscilused=0;//if the oscil or fmoscil are used by another voice
- for (int i=0;i<NUM_VOICES;i++){
- if (VoicePar[i].Pextoscil==nvoice) oscilused=1;
- if (VoicePar[i].PextFMoscil==nvoice) fmoscilused=1;
+ for (int i=0;i<NUM_VOICES;i++) {
+ if (VoicePar[i].Pextoscil==nvoice) oscilused=1;
+ if (VoicePar[i].PextFMoscil==nvoice) fmoscilused=1;
};
-
+
xml->addparbool("enabled",VoicePar[nvoice].Enabled);
if (((VoicePar[nvoice].Enabled==0)&&(oscilused==0)&&(fmoscilused==0))&&(xml->minimal)) return;
xml->addpar("type",VoicePar[nvoice].Type);
xml->addpar("delay",VoicePar[nvoice].PDelay);
xml->addparbool("resonance",VoicePar[nvoice].Presonance);
-
+
xml->addpar("ext_oscil",VoicePar[nvoice].Pextoscil);
xml->addpar("ext_fm_oscil",VoicePar[nvoice].PextFMoscil);
@@ -256,259 +264,262 @@ void ADnoteParameters::add2XMLsection(XMLwrapper *xml,int n){
xml->addpar("fm_enabled",VoicePar[nvoice].PFMEnabled);
xml->beginbranch("OSCIL");
- VoicePar[nvoice].OscilSmp->add2XML(xml);
+ VoicePar[nvoice].OscilSmp->add2XML(xml);
xml->endbranch();
xml->beginbranch("AMPLITUDE_PARAMETERS");
- xml->addpar("panning",VoicePar[nvoice].PPanning);
- xml->addpar("volume",VoicePar[nvoice].PVolume);
- xml->addparbool("volume_minus",VoicePar[nvoice].PVolumeminus);
- xml->addpar("velocity_sensing",VoicePar[nvoice].PAmpVelocityScaleFunction);
-
- xml->addparbool("amp_envelope_enabled",VoicePar[nvoice].PAmpEnvelopeEnabled);
- if ((VoicePar[nvoice].PAmpEnvelopeEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("AMPLITUDE_ENVELOPE");
- VoicePar[nvoice].AmpEnvelope->add2XML(xml);
- xml->endbranch();
- };
- xml->addparbool("amp_lfo_enabled",VoicePar[nvoice].PAmpLfoEnabled);
- if ((VoicePar[nvoice].PAmpLfoEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("AMPLITUDE_LFO");
- VoicePar[nvoice].AmpLfo->add2XML(xml);
- xml->endbranch();
- };
+ xml->addpar("panning",VoicePar[nvoice].PPanning);
+ xml->addpar("volume",VoicePar[nvoice].PVolume);
+ xml->addparbool("volume_minus",VoicePar[nvoice].PVolumeminus);
+ xml->addpar("velocity_sensing",VoicePar[nvoice].PAmpVelocityScaleFunction);
+
+ xml->addparbool("amp_envelope_enabled",VoicePar[nvoice].PAmpEnvelopeEnabled);
+ if ((VoicePar[nvoice].PAmpEnvelopeEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("AMPLITUDE_ENVELOPE");
+ VoicePar[nvoice].AmpEnvelope->add2XML(xml);
+ xml->endbranch();
+ };
+ xml->addparbool("amp_lfo_enabled",VoicePar[nvoice].PAmpLfoEnabled);
+ if ((VoicePar[nvoice].PAmpLfoEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("AMPLITUDE_LFO");
+ VoicePar[nvoice].AmpLfo->add2XML(xml);
+ xml->endbranch();
+ };
xml->endbranch();
xml->beginbranch("FREQUENCY_PARAMETERS");
- xml->addparbool("fixed_freq",VoicePar[nvoice].Pfixedfreq);
- xml->addpar("fixed_freq_et",VoicePar[nvoice].PfixedfreqET);
- xml->addpar("detune",VoicePar[nvoice].PDetune);
- xml->addpar("coarse_detune",VoicePar[nvoice].PCoarseDetune);
- xml->addpar("detune_type",VoicePar[nvoice].PDetuneType);
-
- xml->addparbool("freq_envelope_enabled",VoicePar[nvoice].PFreqEnvelopeEnabled);
- if ((VoicePar[nvoice].PFreqEnvelopeEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("FREQUENCY_ENVELOPE");
- VoicePar[nvoice].FreqEnvelope->add2XML(xml);
- xml->endbranch();
- };
- xml->addparbool("freq_lfo_enabled",VoicePar[nvoice].PFreqLfoEnabled);
- if ((VoicePar[nvoice].PFreqLfoEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("FREQUENCY_LFO");
- VoicePar[nvoice].FreqLfo->add2XML(xml);
- xml->endbranch();
- };
+ xml->addparbool("fixed_freq",VoicePar[nvoice].Pfixedfreq);
+ xml->addpar("fixed_freq_et",VoicePar[nvoice].PfixedfreqET);
+ xml->addpar("detune",VoicePar[nvoice].PDetune);
+ xml->addpar("coarse_detune",VoicePar[nvoice].PCoarseDetune);
+ xml->addpar("detune_type",VoicePar[nvoice].PDetuneType);
+
+ xml->addparbool("freq_envelope_enabled",VoicePar[nvoice].PFreqEnvelopeEnabled);
+ if ((VoicePar[nvoice].PFreqEnvelopeEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("FREQUENCY_ENVELOPE");
+ VoicePar[nvoice].FreqEnvelope->add2XML(xml);
+ xml->endbranch();
+ };
+ xml->addparbool("freq_lfo_enabled",VoicePar[nvoice].PFreqLfoEnabled);
+ if ((VoicePar[nvoice].PFreqLfoEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("FREQUENCY_LFO");
+ VoicePar[nvoice].FreqLfo->add2XML(xml);
+ xml->endbranch();
+ };
xml->endbranch();
- if ((VoicePar[nvoice].PFilterEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("FILTER_PARAMETERS");
- xml->beginbranch("FILTER");
- VoicePar[nvoice].VoiceFilter->add2XML(xml);
- xml->endbranch();
-
- xml->addparbool("filter_envelope_enabled",VoicePar[nvoice].PFilterEnvelopeEnabled);
- if ((VoicePar[nvoice].PFilterEnvelopeEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("FILTER_ENVELOPE");
- VoicePar[nvoice].FilterEnvelope->add2XML(xml);
- xml->endbranch();
- };
-
- xml->addparbool("filter_lfo_enabled",VoicePar[nvoice].PFilterLfoEnabled);
- if ((VoicePar[nvoice].PFilterLfoEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("FILTER_LFO");
- VoicePar[nvoice].FilterLfo->add2XML(xml);
- xml->endbranch();
- };
- xml->endbranch();
+ if ((VoicePar[nvoice].PFilterEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("FILTER_PARAMETERS");
+ xml->beginbranch("FILTER");
+ VoicePar[nvoice].VoiceFilter->add2XML(xml);
+ xml->endbranch();
+
+ xml->addparbool("filter_envelope_enabled",VoicePar[nvoice].PFilterEnvelopeEnabled);
+ if ((VoicePar[nvoice].PFilterEnvelopeEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("FILTER_ENVELOPE");
+ VoicePar[nvoice].FilterEnvelope->add2XML(xml);
+ xml->endbranch();
+ };
+
+ xml->addparbool("filter_lfo_enabled",VoicePar[nvoice].PFilterLfoEnabled);
+ if ((VoicePar[nvoice].PFilterLfoEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("FILTER_LFO");
+ VoicePar[nvoice].FilterLfo->add2XML(xml);
+ xml->endbranch();
+ };
+ xml->endbranch();
};
- if ((VoicePar[nvoice].PFMEnabled!=0)||(fmoscilused!=0)||(!xml->minimal)){
- xml->beginbranch("FM_PARAMETERS");
- xml->addpar("input_voice",VoicePar[nvoice].PFMVoice);
-
- xml->addpar("volume",VoicePar[nvoice].PFMVolume);
- xml->addpar("volume_damp",VoicePar[nvoice].PFMVolumeDamp);
- xml->addpar("velocity_sensing",VoicePar[nvoice].PFMVelocityScaleFunction);
-
- xml->addparbool("amp_envelope_enabled",VoicePar[nvoice].PFMAmpEnvelopeEnabled);
- if ((VoicePar[nvoice].PFMAmpEnvelopeEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("AMPLITUDE_ENVELOPE");
- VoicePar[nvoice].FMAmpEnvelope->add2XML(xml);
- xml->endbranch();
- };
- xml->beginbranch("MODULATOR");
- xml->addpar("detune",VoicePar[nvoice].PFMDetune);
- xml->addpar("coarse_detune",VoicePar[nvoice].PFMCoarseDetune);
- xml->addpar("detune_type",VoicePar[nvoice].PFMDetuneType);
-
- xml->addparbool("freq_envelope_enabled",VoicePar[nvoice].PFMFreqEnvelopeEnabled);
- if ((VoicePar[nvoice].PFMFreqEnvelopeEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("FREQUENCY_ENVELOPE");
- VoicePar[nvoice].FMFreqEnvelope->add2XML(xml);
- xml->endbranch();
- };
-
- xml->beginbranch("OSCIL");
- VoicePar[nvoice].FMSmp->add2XML(xml);
- xml->endbranch();
-
- xml->endbranch();
- xml->endbranch();
+ if ((VoicePar[nvoice].PFMEnabled!=0)||(fmoscilused!=0)||(!xml->minimal)) {
+ xml->beginbranch("FM_PARAMETERS");
+ xml->addpar("input_voice",VoicePar[nvoice].PFMVoice);
+
+ xml->addpar("volume",VoicePar[nvoice].PFMVolume);
+ xml->addpar("volume_damp",VoicePar[nvoice].PFMVolumeDamp);
+ xml->addpar("velocity_sensing",VoicePar[nvoice].PFMVelocityScaleFunction);
+
+ xml->addparbool("amp_envelope_enabled",VoicePar[nvoice].PFMAmpEnvelopeEnabled);
+ if ((VoicePar[nvoice].PFMAmpEnvelopeEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("AMPLITUDE_ENVELOPE");
+ VoicePar[nvoice].FMAmpEnvelope->add2XML(xml);
+ xml->endbranch();
+ };
+ xml->beginbranch("MODULATOR");
+ xml->addpar("detune",VoicePar[nvoice].PFMDetune);
+ xml->addpar("coarse_detune",VoicePar[nvoice].PFMCoarseDetune);
+ xml->addpar("detune_type",VoicePar[nvoice].PFMDetuneType);
+
+ xml->addparbool("freq_envelope_enabled",VoicePar[nvoice].PFMFreqEnvelopeEnabled);
+ if ((VoicePar[nvoice].PFMFreqEnvelopeEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("FREQUENCY_ENVELOPE");
+ VoicePar[nvoice].FMFreqEnvelope->add2XML(xml);
+ xml->endbranch();
+ };
+
+ xml->beginbranch("OSCIL");
+ VoicePar[nvoice].FMSmp->add2XML(xml);
+ xml->endbranch();
+
+ xml->endbranch();
+ xml->endbranch();
};
};
-void ADnoteParameters::add2XML(XMLwrapper *xml){
+void ADnoteParameters::add2XML(XMLwrapper *xml)
+{
xml->addparbool("stereo",GlobalPar.PStereo);
xml->beginbranch("AMPLITUDE_PARAMETERS");
- xml->addpar("volume",GlobalPar.PVolume);
- xml->addpar("panning",GlobalPar.PPanning);
- xml->addpar("velocity_sensing",GlobalPar.PAmpVelocityScaleFunction);
- xml->addpar("punch_strength",GlobalPar.PPunchStrength);
- xml->addpar("punch_time",GlobalPar.PPunchTime);
- xml->addpar("punch_stretch",GlobalPar.PPunchStretch);
- xml->addpar("punch_velocity_sensing",GlobalPar.PPunchVelocitySensing);
- xml->addpar("harmonic_randomness_grouping",GlobalPar.Hrandgrouping);
-
- xml->beginbranch("AMPLITUDE_ENVELOPE");
- GlobalPar.AmpEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("AMPLITUDE_LFO");
- GlobalPar.AmpLfo->add2XML(xml);
- xml->endbranch();
+ xml->addpar("volume",GlobalPar.PVolume);
+ xml->addpar("panning",GlobalPar.PPanning);
+ xml->addpar("velocity_sensing",GlobalPar.PAmpVelocityScaleFunction);
+ xml->addpar("punch_strength",GlobalPar.PPunchStrength);
+ xml->addpar("punch_time",GlobalPar.PPunchTime);
+ xml->addpar("punch_stretch",GlobalPar.PPunchStretch);
+ xml->addpar("punch_velocity_sensing",GlobalPar.PPunchVelocitySensing);
+ xml->addpar("harmonic_randomness_grouping",GlobalPar.Hrandgrouping);
+
+ xml->beginbranch("AMPLITUDE_ENVELOPE");
+ GlobalPar.AmpEnvelope->add2XML(xml);
+ xml->endbranch();
+
+ xml->beginbranch("AMPLITUDE_LFO");
+ GlobalPar.AmpLfo->add2XML(xml);
+ xml->endbranch();
xml->endbranch();
xml->beginbranch("FREQUENCY_PARAMETERS");
- xml->addpar("detune",GlobalPar.PDetune);
-
- xml->addpar("coarse_detune",GlobalPar.PCoarseDetune);
- xml->addpar("detune_type",GlobalPar.PDetuneType);
-
- xml->addpar("bandwidth",GlobalPar.PBandwidth);
-
- xml->beginbranch("FREQUENCY_ENVELOPE");
- GlobalPar.FreqEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FREQUENCY_LFO");
- GlobalPar.FreqLfo->add2XML(xml);
- xml->endbranch();
+ xml->addpar("detune",GlobalPar.PDetune);
+
+ xml->addpar("coarse_detune",GlobalPar.PCoarseDetune);
+ xml->addpar("detune_type",GlobalPar.PDetuneType);
+
+ xml->addpar("bandwidth",GlobalPar.PBandwidth);
+
+ xml->beginbranch("FREQUENCY_ENVELOPE");
+ GlobalPar.FreqEnvelope->add2XML(xml);
xml->endbranch();
-
-
+
+ xml->beginbranch("FREQUENCY_LFO");
+ GlobalPar.FreqLfo->add2XML(xml);
+ xml->endbranch();
+ xml->endbranch();
+
+
xml->beginbranch("FILTER_PARAMETERS");
- xml->addpar("velocity_sensing_amplitude",GlobalPar.PFilterVelocityScale);
- xml->addpar("velocity_sensing",GlobalPar.PFilterVelocityScaleFunction);
-
- xml->beginbranch("FILTER");
- GlobalPar.GlobalFilter->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FILTER_ENVELOPE");
- GlobalPar.FilterEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FILTER_LFO");
- GlobalPar.FilterLfo->add2XML(xml);
- xml->endbranch();
+ xml->addpar("velocity_sensing_amplitude",GlobalPar.PFilterVelocityScale);
+ xml->addpar("velocity_sensing",GlobalPar.PFilterVelocityScaleFunction);
+
+ xml->beginbranch("FILTER");
+ GlobalPar.GlobalFilter->add2XML(xml);
xml->endbranch();
-
+
+ xml->beginbranch("FILTER_ENVELOPE");
+ GlobalPar.FilterEnvelope->add2XML(xml);
+ xml->endbranch();
+
+ xml->beginbranch("FILTER_LFO");
+ GlobalPar.FilterLfo->add2XML(xml);
+ xml->endbranch();
+ xml->endbranch();
+
xml->beginbranch("RESONANCE");
- GlobalPar.Reson->add2XML(xml);
+ GlobalPar.Reson->add2XML(xml);
xml->endbranch();
- for (int nvoice=0;nvoice<NUM_VOICES;nvoice++){
- xml->beginbranch("VOICE",nvoice);
- add2XMLsection(xml,nvoice);
- xml->endbranch();
+ for (int nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ xml->beginbranch("VOICE",nvoice);
+ add2XMLsection(xml,nvoice);
+ xml->endbranch();
};
};
-void ADnoteParameters::getfromXML(XMLwrapper *xml){
+void ADnoteParameters::getfromXML(XMLwrapper *xml)
+{
GlobalPar.PStereo=xml->getparbool("stereo",GlobalPar.PStereo);
-
- if (xml->enterbranch("AMPLITUDE_PARAMETERS")){
- GlobalPar.PVolume=xml->getpar127("volume",GlobalPar.PVolume);
- GlobalPar.PPanning=xml->getpar127("panning",GlobalPar.PPanning);
- GlobalPar.PAmpVelocityScaleFunction=xml->getpar127("velocity_sensing",GlobalPar.PAmpVelocityScaleFunction);
-
- GlobalPar.PPunchStrength=xml->getpar127("punch_strength",GlobalPar.PPunchStrength);
- GlobalPar.PPunchTime=xml->getpar127("punch_time",GlobalPar.PPunchTime);
- GlobalPar.PPunchStretch=xml->getpar127("punch_stretch",GlobalPar.PPunchStretch);
- GlobalPar.PPunchVelocitySensing=xml->getpar127("punch_velocity_sensing",GlobalPar.PPunchVelocitySensing);
- GlobalPar.Hrandgrouping=xml->getpar127("harmonic_randomness_grouping",GlobalPar.Hrandgrouping);
-
- if (xml->enterbranch("AMPLITUDE_ENVELOPE")){
- GlobalPar.AmpEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
- if (xml->enterbranch("AMPLITUDE_LFO")){
- GlobalPar.AmpLfo->getfromXML(xml);
- xml->exitbranch();
- };
-
- xml->exitbranch();
+
+ if (xml->enterbranch("AMPLITUDE_PARAMETERS")) {
+ GlobalPar.PVolume=xml->getpar127("volume",GlobalPar.PVolume);
+ GlobalPar.PPanning=xml->getpar127("panning",GlobalPar.PPanning);
+ GlobalPar.PAmpVelocityScaleFunction=xml->getpar127("velocity_sensing",GlobalPar.PAmpVelocityScaleFunction);
+
+ GlobalPar.PPunchStrength=xml->getpar127("punch_strength",GlobalPar.PPunchStrength);
+ GlobalPar.PPunchTime=xml->getpar127("punch_time",GlobalPar.PPunchTime);
+ GlobalPar.PPunchStretch=xml->getpar127("punch_stretch",GlobalPar.PPunchStretch);
+ GlobalPar.PPunchVelocitySensing=xml->getpar127("punch_velocity_sensing",GlobalPar.PPunchVelocitySensing);
+ GlobalPar.Hrandgrouping=xml->getpar127("harmonic_randomness_grouping",GlobalPar.Hrandgrouping);
+
+ if (xml->enterbranch("AMPLITUDE_ENVELOPE")) {
+ GlobalPar.AmpEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ if (xml->enterbranch("AMPLITUDE_LFO")) {
+ GlobalPar.AmpLfo->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ xml->exitbranch();
};
- if (xml->enterbranch("FREQUENCY_PARAMETERS")){
- GlobalPar.PDetune=xml->getpar("detune",GlobalPar.PDetune,0,16383);
- GlobalPar.PCoarseDetune=xml->getpar("coarse_detune",GlobalPar.PCoarseDetune,0,16383);
+ if (xml->enterbranch("FREQUENCY_PARAMETERS")) {
+ GlobalPar.PDetune=xml->getpar("detune",GlobalPar.PDetune,0,16383);
+ GlobalPar.PCoarseDetune=xml->getpar("coarse_detune",GlobalPar.PCoarseDetune,0,16383);
GlobalPar.PDetuneType=xml->getpar127("detune_type",GlobalPar.PDetuneType);
GlobalPar.PBandwidth=xml->getpar127("bandwidth",GlobalPar.PBandwidth);
-
- xml->enterbranch("FREQUENCY_ENVELOPE");
- GlobalPar.FreqEnvelope->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FREQUENCY_LFO");
- GlobalPar.FreqLfo->getfromXML(xml);
- xml->exitbranch();
-
- xml->exitbranch();
+
+ xml->enterbranch("FREQUENCY_ENVELOPE");
+ GlobalPar.FreqEnvelope->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->enterbranch("FREQUENCY_LFO");
+ GlobalPar.FreqLfo->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->exitbranch();
};
-
-
- if (xml->enterbranch("FILTER_PARAMETERS")){
- GlobalPar.PFilterVelocityScale=xml->getpar127("velocity_sensing_amplitude",GlobalPar.PFilterVelocityScale);
- GlobalPar.PFilterVelocityScaleFunction=xml->getpar127("velocity_sensing",GlobalPar.PFilterVelocityScaleFunction);
-
- xml->enterbranch("FILTER");
- GlobalPar.GlobalFilter->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FILTER_ENVELOPE");
- GlobalPar.FilterEnvelope->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FILTER_LFO");
- GlobalPar.FilterLfo->getfromXML(xml);
- xml->exitbranch();
- xml->exitbranch();
+
+
+ if (xml->enterbranch("FILTER_PARAMETERS")) {
+ GlobalPar.PFilterVelocityScale=xml->getpar127("velocity_sensing_amplitude",GlobalPar.PFilterVelocityScale);
+ GlobalPar.PFilterVelocityScaleFunction=xml->getpar127("velocity_sensing",GlobalPar.PFilterVelocityScaleFunction);
+
+ xml->enterbranch("FILTER");
+ GlobalPar.GlobalFilter->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->enterbranch("FILTER_ENVELOPE");
+ GlobalPar.FilterEnvelope->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->enterbranch("FILTER_LFO");
+ GlobalPar.FilterLfo->getfromXML(xml);
+ xml->exitbranch();
+ xml->exitbranch();
};
- if (xml->enterbranch("RESONANCE")){
- GlobalPar.Reson->getfromXML(xml);
- xml->exitbranch();
+ if (xml->enterbranch("RESONANCE")) {
+ GlobalPar.Reson->getfromXML(xml);
+ xml->exitbranch();
};
- for (int nvoice=0;nvoice<NUM_VOICES;nvoice++){
- VoicePar[nvoice].Enabled=0;
- if (xml->enterbranch("VOICE",nvoice)==0) continue;
- getfromXMLsection(xml,nvoice);
- xml->exitbranch();
+ for (int nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ VoicePar[nvoice].Enabled=0;
+ if (xml->enterbranch("VOICE",nvoice)==0) continue;
+ getfromXMLsection(xml,nvoice);
+ xml->exitbranch();
};
-
-
+
+
};
-void ADnoteParameters::getfromXMLsection(XMLwrapper *xml,int n){
+void ADnoteParameters::getfromXMLsection(XMLwrapper *xml,int n)
+{
int nvoice=n;
if (nvoice>=NUM_VOICES) return;
@@ -517,7 +528,7 @@ void ADnoteParameters::getfromXMLsection(XMLwrapper *xml,int n){
VoicePar[nvoice].Type=xml->getpar127("type",VoicePar[nvoice].Type);
VoicePar[nvoice].PDelay=xml->getpar127("delay",VoicePar[nvoice].PDelay);
VoicePar[nvoice].Presonance=xml->getparbool("resonance",VoicePar[nvoice].Presonance);
-
+
VoicePar[nvoice].Pextoscil=xml->getpar("ext_oscil",-1,-1,nvoice-1);
VoicePar[nvoice].PextFMoscil=xml->getpar("ext_fm_oscil",-1,-1,nvoice-1);
@@ -529,108 +540,108 @@ void ADnoteParameters::getfromXMLsection(XMLwrapper *xml,int n){
VoicePar[nvoice].PFMEnabled=xml->getpar127("fm_enabled",VoicePar[nvoice].PFMEnabled);
- if (xml->enterbranch("OSCIL")){
- VoicePar[nvoice].OscilSmp->getfromXML(xml);
- xml->exitbranch();
+ if (xml->enterbranch("OSCIL")) {
+ VoicePar[nvoice].OscilSmp->getfromXML(xml);
+ xml->exitbranch();
};
- if (xml->enterbranch("AMPLITUDE_PARAMETERS")){
- VoicePar[nvoice].PPanning=xml->getpar127("panning",VoicePar[nvoice].PPanning);
- VoicePar[nvoice].PVolume=xml->getpar127("volume",VoicePar[nvoice].PVolume);
- VoicePar[nvoice].PVolumeminus=xml->getparbool("volume_minus",VoicePar[nvoice].PVolumeminus);
- VoicePar[nvoice].PAmpVelocityScaleFunction=xml->getpar127("velocity_sensing",VoicePar[nvoice].PAmpVelocityScaleFunction);
-
- VoicePar[nvoice].PAmpEnvelopeEnabled=xml->getparbool("amp_envelope_enabled",VoicePar[nvoice].PAmpEnvelopeEnabled);
- if (xml->enterbranch("AMPLITUDE_ENVELOPE")){
- VoicePar[nvoice].AmpEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
- VoicePar[nvoice].PAmpLfoEnabled=xml->getparbool("amp_lfo_enabled",VoicePar[nvoice].PAmpLfoEnabled);
- if (xml->enterbranch("AMPLITUDE_LFO")){
- VoicePar[nvoice].AmpLfo->getfromXML(xml);
- xml->exitbranch();
- };
- xml->exitbranch();
+ if (xml->enterbranch("AMPLITUDE_PARAMETERS")) {
+ VoicePar[nvoice].PPanning=xml->getpar127("panning",VoicePar[nvoice].PPanning);
+ VoicePar[nvoice].PVolume=xml->getpar127("volume",VoicePar[nvoice].PVolume);
+ VoicePar[nvoice].PVolumeminus=xml->getparbool("volume_minus",VoicePar[nvoice].PVolumeminus);
+ VoicePar[nvoice].PAmpVelocityScaleFunction=xml->getpar127("velocity_sensing",VoicePar[nvoice].PAmpVelocityScaleFunction);
+
+ VoicePar[nvoice].PAmpEnvelopeEnabled=xml->getparbool("amp_envelope_enabled",VoicePar[nvoice].PAmpEnvelopeEnabled);
+ if (xml->enterbranch("AMPLITUDE_ENVELOPE")) {
+ VoicePar[nvoice].AmpEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ VoicePar[nvoice].PAmpLfoEnabled=xml->getparbool("amp_lfo_enabled",VoicePar[nvoice].PAmpLfoEnabled);
+ if (xml->enterbranch("AMPLITUDE_LFO")) {
+ VoicePar[nvoice].AmpLfo->getfromXML(xml);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
- if (xml->enterbranch("FREQUENCY_PARAMETERS")){
- VoicePar[nvoice].Pfixedfreq=xml->getparbool("fixed_freq",VoicePar[nvoice].Pfixedfreq);
- VoicePar[nvoice].PfixedfreqET=xml->getpar127("fixed_freq_et",VoicePar[nvoice].PfixedfreqET);
-
-
- VoicePar[nvoice].PDetune=xml->getpar("detune",VoicePar[nvoice].PDetune,0,16383);
-
- VoicePar[nvoice].PCoarseDetune=xml->getpar("coarse_detune",VoicePar[nvoice].PCoarseDetune,0,16383);
- VoicePar[nvoice].PDetuneType=xml->getpar127("detune_type",VoicePar[nvoice].PDetuneType);
-
- VoicePar[nvoice].PFreqEnvelopeEnabled=xml->getparbool("freq_envelope_enabled",VoicePar[nvoice].PFreqEnvelopeEnabled);
- if (xml->enterbranch("FREQUENCY_ENVELOPE")){
- VoicePar[nvoice].FreqEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
- VoicePar[nvoice].PFreqLfoEnabled=xml->getparbool("freq_lfo_enabled",VoicePar[nvoice].PFreqLfoEnabled);
- if (xml->enterbranch("FREQUENCY_LFO")){
- VoicePar[nvoice].FreqLfo->getfromXML(xml);
- xml->exitbranch();
- };
- xml->exitbranch();
+ if (xml->enterbranch("FREQUENCY_PARAMETERS")) {
+ VoicePar[nvoice].Pfixedfreq=xml->getparbool("fixed_freq",VoicePar[nvoice].Pfixedfreq);
+ VoicePar[nvoice].PfixedfreqET=xml->getpar127("fixed_freq_et",VoicePar[nvoice].PfixedfreqET);
+
+
+ VoicePar[nvoice].PDetune=xml->getpar("detune",VoicePar[nvoice].PDetune,0,16383);
+
+ VoicePar[nvoice].PCoarseDetune=xml->getpar("coarse_detune",VoicePar[nvoice].PCoarseDetune,0,16383);
+ VoicePar[nvoice].PDetuneType=xml->getpar127("detune_type",VoicePar[nvoice].PDetuneType);
+
+ VoicePar[nvoice].PFreqEnvelopeEnabled=xml->getparbool("freq_envelope_enabled",VoicePar[nvoice].PFreqEnvelopeEnabled);
+ if (xml->enterbranch("FREQUENCY_ENVELOPE")) {
+ VoicePar[nvoice].FreqEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ VoicePar[nvoice].PFreqLfoEnabled=xml->getparbool("freq_lfo_enabled",VoicePar[nvoice].PFreqLfoEnabled);
+ if (xml->enterbranch("FREQUENCY_LFO")) {
+ VoicePar[nvoice].FreqLfo->getfromXML(xml);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
- if (xml->enterbranch("FILTER_PARAMETERS")){
- if (xml->enterbranch("FILTER")){
- VoicePar[nvoice].VoiceFilter->getfromXML(xml);
- xml->exitbranch();
- };
-
- VoicePar[nvoice].PFilterEnvelopeEnabled=xml->getparbool("filter_envelope_enabled",VoicePar[nvoice].PFilterEnvelopeEnabled);
- if (xml->enterbranch("FILTER_ENVELOPE")){
- VoicePar[nvoice].FilterEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
- VoicePar[nvoice].PFilterLfoEnabled=xml->getparbool("filter_lfo_enabled",VoicePar[nvoice].PFilterLfoEnabled);
- if (xml->enterbranch("FILTER_LFO")){
- VoicePar[nvoice].FilterLfo->getfromXML(xml);
- xml->exitbranch();
- };
- xml->exitbranch();
+ if (xml->enterbranch("FILTER_PARAMETERS")) {
+ if (xml->enterbranch("FILTER")) {
+ VoicePar[nvoice].VoiceFilter->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ VoicePar[nvoice].PFilterEnvelopeEnabled=xml->getparbool("filter_envelope_enabled",VoicePar[nvoice].PFilterEnvelopeEnabled);
+ if (xml->enterbranch("FILTER_ENVELOPE")) {
+ VoicePar[nvoice].FilterEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ VoicePar[nvoice].PFilterLfoEnabled=xml->getparbool("filter_lfo_enabled",VoicePar[nvoice].PFilterLfoEnabled);
+ if (xml->enterbranch("FILTER_LFO")) {
+ VoicePar[nvoice].FilterLfo->getfromXML(xml);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
- if (xml->enterbranch("FM_PARAMETERS")){
- VoicePar[nvoice].PFMVoice=xml->getpar("input_voice",VoicePar[nvoice].PFMVoice,-1,nvoice-1);
-
- VoicePar[nvoice].PFMVolume=xml->getpar127("volume",VoicePar[nvoice].PFMVolume);
- VoicePar[nvoice].PFMVolumeDamp=xml->getpar127("volume_damp",VoicePar[nvoice].PFMVolumeDamp);
- VoicePar[nvoice].PFMVelocityScaleFunction=xml->getpar127("velocity_sensing",VoicePar[nvoice].PFMVelocityScaleFunction);
-
- VoicePar[nvoice].PFMAmpEnvelopeEnabled=xml->getparbool("amp_envelope_enabled",VoicePar[nvoice].PFMAmpEnvelopeEnabled);
- if (xml->enterbranch("AMPLITUDE_ENVELOPE")){
- VoicePar[nvoice].FMAmpEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
- if (xml->enterbranch("MODULATOR")){
- VoicePar[nvoice].PFMDetune=xml->getpar("detune",VoicePar[nvoice].PFMDetune,0,16383);
- VoicePar[nvoice].PFMCoarseDetune=xml->getpar("coarse_detune",VoicePar[nvoice].PFMCoarseDetune,0,16383);
- VoicePar[nvoice].PFMDetuneType=xml->getpar127("detune_type",VoicePar[nvoice].PFMDetuneType);
-
- VoicePar[nvoice].PFMFreqEnvelopeEnabled=xml->getparbool("freq_envelope_enabled",VoicePar[nvoice].PFMFreqEnvelopeEnabled);
- if (xml->enterbranch("FREQUENCY_ENVELOPE")){
- VoicePar[nvoice].FMFreqEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
- if (xml->enterbranch("OSCIL")){
- VoicePar[nvoice].FMSmp->getfromXML(xml);
- xml->exitbranch();
- };
-
- xml->exitbranch();
- };
- xml->exitbranch();
+ if (xml->enterbranch("FM_PARAMETERS")) {
+ VoicePar[nvoice].PFMVoice=xml->getpar("input_voice",VoicePar[nvoice].PFMVoice,-1,nvoice-1);
+
+ VoicePar[nvoice].PFMVolume=xml->getpar127("volume",VoicePar[nvoice].PFMVolume);
+ VoicePar[nvoice].PFMVolumeDamp=xml->getpar127("volume_damp",VoicePar[nvoice].PFMVolumeDamp);
+ VoicePar[nvoice].PFMVelocityScaleFunction=xml->getpar127("velocity_sensing",VoicePar[nvoice].PFMVelocityScaleFunction);
+
+ VoicePar[nvoice].PFMAmpEnvelopeEnabled=xml->getparbool("amp_envelope_enabled",VoicePar[nvoice].PFMAmpEnvelopeEnabled);
+ if (xml->enterbranch("AMPLITUDE_ENVELOPE")) {
+ VoicePar[nvoice].FMAmpEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ if (xml->enterbranch("MODULATOR")) {
+ VoicePar[nvoice].PFMDetune=xml->getpar("detune",VoicePar[nvoice].PFMDetune,0,16383);
+ VoicePar[nvoice].PFMCoarseDetune=xml->getpar("coarse_detune",VoicePar[nvoice].PFMCoarseDetune,0,16383);
+ VoicePar[nvoice].PFMDetuneType=xml->getpar127("detune_type",VoicePar[nvoice].PFMDetuneType);
+
+ VoicePar[nvoice].PFMFreqEnvelopeEnabled=xml->getparbool("freq_envelope_enabled",VoicePar[nvoice].PFMFreqEnvelopeEnabled);
+ if (xml->enterbranch("FREQUENCY_ENVELOPE")) {
+ VoicePar[nvoice].FMFreqEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ if (xml->enterbranch("OSCIL")) {
+ VoicePar[nvoice].FMSmp->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
};
diff --git a/src/Params/ADnoteParameters.h b/src/Params/ADnoteParameters.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
ADnoteParameters.h - Parameters for ADnote (ADsynth)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -35,248 +35,249 @@
#include "../DSP/FFTwrapper.h"
#include "Presets.h"
- enum FMTYPE{NONE,MORPH,RING_MOD,PHASE_MOD,FREQ_MOD,PITCH_MOD};
-
- /*****************************************************************/
- /* GLOBAL PARAMETERS */
- /*****************************************************************/
-
- struct ADnoteGlobalParam{
-
- /* The instrument type - MONO/STEREO
- If the mode is MONO, the panning of voices are not used
- Stereo=1, Mono=0. */
-
- unsigned char PStereo;
-
-
- /******************************************
- * FREQUENCY GLOBAL PARAMETERS *
- ******************************************/
- unsigned short int PDetune;//fine detune
- 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
-
- EnvelopeParams *FreqEnvelope; //Frequency Envelope
-
- LFOParams *FreqLfo;//Frequency LFO
-
- /********************************************
- * AMPLITUDE GLOBAL PARAMETERS *
- ********************************************/
-
- /* Panning - 0 - random
- 1 - left
- 64 - center
- 127 - right */
- unsigned char PPanning;
-
- unsigned char PVolume;
-
- unsigned char PAmpVelocityScaleFunction;
-
- EnvelopeParams *AmpEnvelope;
-
- LFOParams *AmpLfo;
-
- unsigned char PPunchStrength,PPunchTime,PPunchStretch,PPunchVelocitySensing;
-
- /******************************************
- * FILTER GLOBAL PARAMETERS *
- ******************************************/
- FilterParams *GlobalFilter;
-
- // filter velocity sensing
- unsigned char PFilterVelocityScale;
-
- // filter velocity sensing
- unsigned char PFilterVelocityScaleFunction;
-
- EnvelopeParams *FilterEnvelope;
-
- LFOParams *FilterLfo;
-
- // RESONANCE
- Resonance *Reson;
-
- //how the randomness is applied to the harmonics on more voices using the same oscillator
- unsigned char Hrandgrouping;
- };
-
-
-
- /***********************************************************/
- /* VOICE PARAMETERS */
- /***********************************************************/
- struct ADnoteVoiceParam{
-
- /** If the voice is enabled */
- unsigned char Enabled;
-
- /** Type of the voice (0=Sound,1=Noise)*/
- unsigned char Type;
-
- /** Voice Delay */
- unsigned char PDelay;
-
- /** If the resonance is enabled for this voice */
- unsigned char Presonance;
-
- // What external oscil should I use, -1 for internal OscilSmp&FMSmp
- short int Pextoscil,PextFMoscil;
- // it is not allowed that the externoscil,externFMoscil => current voice
-
- // oscillator phases
- unsigned char Poscilphase,PFMoscilphase;
-
- // filter bypass
- unsigned char Pfilterbypass;
-
- /** Voice oscillator */
- OscilGen *OscilSmp;
-
- /**********************************
- * FREQUENCY PARAMETERS *
- **********************************/
-
- /** If the base frequency is fixed to 440 Hz*/
- unsigned char Pfixedfreq;
-
- /* 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;
-
- /** Fine detune */
- unsigned short int PDetune;
-
- /** Coarse detune + octave */
- unsigned short int PCoarseDetune;
-
- /** Detune type */
- unsigned char PDetuneType;
-
- /* Frequency Envelope */
- unsigned char PFreqEnvelopeEnabled;
- EnvelopeParams *FreqEnvelope;
-
- /* Frequency LFO */
- unsigned char PFreqLfoEnabled;
- LFOParams *FreqLfo;
-
-
- /***************************
- * AMPLITUDE PARAMETERS *
- ***************************/
-
- /* Panning 0 - random
- 1 - left
- 64 - center
- 127 - right
- The Panning is ignored if the instrument is mono */
- unsigned char PPanning;
-
- /* Voice Volume */
- unsigned char PVolume;
-
- /* If the Volume negative */
- unsigned char PVolumeminus;
-
- /* Velocity sensing */
- unsigned char PAmpVelocityScaleFunction;
-
- /* Amplitude Envelope */
- unsigned char PAmpEnvelopeEnabled;
- EnvelopeParams *AmpEnvelope;
-
- /* Amplitude LFO */
- unsigned char PAmpLfoEnabled;
- LFOParams *AmpLfo;
-
-
-
- /*************************
- * FILTER PARAMETERS *
- *************************/
-
- /* Voice Filter */
- unsigned char PFilterEnabled;
- FilterParams *VoiceFilter;
-
- /* Filter Envelope */
- unsigned char PFilterEnvelopeEnabled;
- EnvelopeParams *FilterEnvelope;
-
- /* LFO Envelope */
- unsigned char PFilterLfoEnabled;
- LFOParams *FilterLfo;
-
- /****************************
- * MODULLATOR PARAMETERS *
- ****************************/
-
- /* Modullator Parameters (0=off,1=Morph,2=RM,3=PM,4=FM.. */
- unsigned char PFMEnabled;
-
- /* Voice that I use as modullator instead of FMSmp.
- It is -1 if I use FMSmp(default).
- It maynot be equal or bigger than current voice */
- short int PFMVoice;
-
- /* Modullator oscillator */
- OscilGen *FMSmp;
-
- /* Modullator Volume */
- unsigned char PFMVolume;
-
- /* Modullator damping at higher frequencies */
- unsigned char PFMVolumeDamp;
-
- /* Modullator Velocity Sensing */
- unsigned char PFMVelocityScaleFunction;
-
- /* Fine Detune of the Modullator*/
- unsigned short int PFMDetune;
-
- /* Coarse Detune of the Modullator */
- unsigned short int PFMCoarseDetune;
-
- /* The detune type */
- unsigned char PFMDetuneType;
-
- /* Frequency Envelope of the Modullator */
- unsigned char PFMFreqEnvelopeEnabled;
- EnvelopeParams *FMFreqEnvelope;
-
- /* Frequency Envelope of the Modullator */
- unsigned char PFMAmpEnvelopeEnabled;
- EnvelopeParams *FMAmpEnvelope;
- };
-
-class ADnoteParameters:public Presets{
- public:
- ADnoteParameters(FFTwrapper *fft_);
- ~ADnoteParameters();
-
- ADnoteGlobalParam GlobalPar;
- ADnoteVoiceParam VoicePar[NUM_VOICES];
-
- void defaults();
- void add2XML(XMLwrapper *xml);
- void getfromXML(XMLwrapper *xml);
-
- REALTYPE getBandwidthDetuneMultiplier();
- private:
- void defaults(int n);//n is the nvoice
-
- void EnableVoice(int nvoice);
- void KillVoice(int nvoice);
- FFTwrapper *fft;
-
- void add2XMLsection(XMLwrapper *xml,int n);
- void getfromXMLsection(XMLwrapper *xml,int n);
+enum FMTYPE {NONE,MORPH,RING_MOD,PHASE_MOD,FREQ_MOD,PITCH_MOD};
+
+/*****************************************************************/
+/* GLOBAL PARAMETERS */
+/*****************************************************************/
+
+struct ADnoteGlobalParam {
+
+ /* The instrument type - MONO/STEREO
+ If the mode is MONO, the panning of voices are not used
+ Stereo=1, Mono=0. */
+
+ unsigned char PStereo;
+
+
+ /******************************************
+ * FREQUENCY GLOBAL PARAMETERS *
+ ******************************************/
+ unsigned short int PDetune;//fine detune
+ 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
+
+ EnvelopeParams *FreqEnvelope; //Frequency Envelope
+
+ LFOParams *FreqLfo;//Frequency LFO
+
+ /********************************************
+ * AMPLITUDE GLOBAL PARAMETERS *
+ ********************************************/
+
+ /* Panning - 0 - random
+ 1 - left
+ 64 - center
+ 127 - right */
+ unsigned char PPanning;
+
+ unsigned char PVolume;
+
+ unsigned char PAmpVelocityScaleFunction;
+
+ EnvelopeParams *AmpEnvelope;
+
+ LFOParams *AmpLfo;
+
+ unsigned char PPunchStrength,PPunchTime,PPunchStretch,PPunchVelocitySensing;
+
+ /******************************************
+ * FILTER GLOBAL PARAMETERS *
+ ******************************************/
+ FilterParams *GlobalFilter;
+
+ // filter velocity sensing
+ unsigned char PFilterVelocityScale;
+
+ // filter velocity sensing
+ unsigned char PFilterVelocityScaleFunction;
+
+ EnvelopeParams *FilterEnvelope;
+
+ LFOParams *FilterLfo;
+
+ // RESONANCE
+ Resonance *Reson;
+
+ //how the randomness is applied to the harmonics on more voices using the same oscillator
+ unsigned char Hrandgrouping;
+};
+
+
+
+/***********************************************************/
+/* VOICE PARAMETERS */
+/***********************************************************/
+struct ADnoteVoiceParam {
+
+ /** If the voice is enabled */
+ unsigned char Enabled;
+
+ /** Type of the voice (0=Sound,1=Noise)*/
+ unsigned char Type;
+
+ /** Voice Delay */
+ unsigned char PDelay;
+
+ /** If the resonance is enabled for this voice */
+ unsigned char Presonance;
+
+ // What external oscil should I use, -1 for internal OscilSmp&FMSmp
+ short int Pextoscil,PextFMoscil;
+ // it is not allowed that the externoscil,externFMoscil => current voice
+
+ // oscillator phases
+ unsigned char Poscilphase,PFMoscilphase;
+
+ // filter bypass
+ unsigned char Pfilterbypass;
+
+ /** Voice oscillator */
+ OscilGen *OscilSmp;
+
+ /**********************************
+ * FREQUENCY PARAMETERS *
+ **********************************/
+
+ /** If the base frequency is fixed to 440 Hz*/
+ unsigned char Pfixedfreq;
+
+ /* 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;
+
+ /** Fine detune */
+ unsigned short int PDetune;
+
+ /** Coarse detune + octave */
+ unsigned short int PCoarseDetune;
+
+ /** Detune type */
+ unsigned char PDetuneType;
+
+ /* Frequency Envelope */
+ unsigned char PFreqEnvelopeEnabled;
+ EnvelopeParams *FreqEnvelope;
+
+ /* Frequency LFO */
+ unsigned char PFreqLfoEnabled;
+ LFOParams *FreqLfo;
+
+
+ /***************************
+ * AMPLITUDE PARAMETERS *
+ ***************************/
+
+ /* Panning 0 - random
+ 1 - left
+ 64 - center
+ 127 - right
+ The Panning is ignored if the instrument is mono */
+ unsigned char PPanning;
+
+ /* Voice Volume */
+ unsigned char PVolume;
+
+ /* If the Volume negative */
+ unsigned char PVolumeminus;
+
+ /* Velocity sensing */
+ unsigned char PAmpVelocityScaleFunction;
+
+ /* Amplitude Envelope */
+ unsigned char PAmpEnvelopeEnabled;
+ EnvelopeParams *AmpEnvelope;
+
+ /* Amplitude LFO */
+ unsigned char PAmpLfoEnabled;
+ LFOParams *AmpLfo;
+
+
+
+ /*************************
+ * FILTER PARAMETERS *
+ *************************/
+
+ /* Voice Filter */
+ unsigned char PFilterEnabled;
+ FilterParams *VoiceFilter;
+
+ /* Filter Envelope */
+ unsigned char PFilterEnvelopeEnabled;
+ EnvelopeParams *FilterEnvelope;
+
+ /* LFO Envelope */
+ unsigned char PFilterLfoEnabled;
+ LFOParams *FilterLfo;
+
+ /****************************
+ * MODULLATOR PARAMETERS *
+ ****************************/
+
+ /* Modullator Parameters (0=off,1=Morph,2=RM,3=PM,4=FM.. */
+ unsigned char PFMEnabled;
+
+ /* Voice that I use as modullator instead of FMSmp.
+ It is -1 if I use FMSmp(default).
+ It maynot be equal or bigger than current voice */
+ short int PFMVoice;
+
+ /* Modullator oscillator */
+ OscilGen *FMSmp;
+
+ /* Modullator Volume */
+ unsigned char PFMVolume;
+
+ /* Modullator damping at higher frequencies */
+ unsigned char PFMVolumeDamp;
+
+ /* Modullator Velocity Sensing */
+ unsigned char PFMVelocityScaleFunction;
+
+ /* Fine Detune of the Modullator*/
+ unsigned short int PFMDetune;
+
+ /* Coarse Detune of the Modullator */
+ unsigned short int PFMCoarseDetune;
+
+ /* The detune type */
+ unsigned char PFMDetuneType;
+
+ /* Frequency Envelope of the Modullator */
+ unsigned char PFMFreqEnvelopeEnabled;
+ EnvelopeParams *FMFreqEnvelope;
+
+ /* Frequency Envelope of the Modullator */
+ unsigned char PFMAmpEnvelopeEnabled;
+ EnvelopeParams *FMAmpEnvelope;
+};
+
+class ADnoteParameters:public Presets
+{
+public:
+ ADnoteParameters(FFTwrapper *fft_);
+ ~ADnoteParameters();
+
+ ADnoteGlobalParam GlobalPar;
+ ADnoteVoiceParam VoicePar[NUM_VOICES];
+
+ void defaults();
+ void add2XML(XMLwrapper *xml);
+ void getfromXML(XMLwrapper *xml);
+
+ REALTYPE getBandwidthDetuneMultiplier();
+private:
+ void defaults(int n);//n is the nvoice
+
+ void EnableVoice(int nvoice);
+ void KillVoice(int nvoice);
+ FFTwrapper *fft;
+
+ void add2XMLsection(XMLwrapper *xml,int n);
+ void getfromXMLsection(XMLwrapper *xml,int n);
};
#endif
diff --git a/src/Params/Controller.C b/src/Params/Controller.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Controller.C - (Midi) Controllers implementation
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,15 +24,18 @@
#include <math.h>
#include <stdio.h>
-Controller::Controller(){
+Controller::Controller()
+{
defaults();
resetall();
};
-Controller::~Controller(){
+Controller::~Controller()
+{
};
-void Controller::defaults(){
+void Controller::defaults()
+{
setpitchwheelbendrange(200);//2 halftones
expression.receive=1;
panning.depth=64;
@@ -63,7 +66,8 @@ void Controller::defaults(){
};
-void Controller::resetall(){
+void Controller::resetall()
+{
setpitchwheel(0);//center
setexpression(127);
setpanning(64);
@@ -76,7 +80,7 @@ void Controller::resetall(){
setsustain(0);
setresonancecenter(64);
setresonancebw(64);
-
+
//reset the NRPN
NRPN.parhi=-1;
NRPN.parlo=-1;
@@ -84,7 +88,8 @@ void Controller::resetall(){
NRPN.vallo=-1;
};
-void Controller::setpitchwheel(int value){
+void Controller::setpitchwheel(int value)
+{
pitchwheel.data=value;
REALTYPE cents=value/8192.0;
cents*=pitchwheel.bendrange;
@@ -92,106 +97,118 @@ void Controller::setpitchwheel(int value){
//fprintf(stderr,"%ld %ld -> %.3f\n",pitchwheel.bendrange,pitchwheel.data,pitchwheel.relfreq);fflush(stderr);
};
-void Controller::setpitchwheelbendrange(unsigned short int value){
+void Controller::setpitchwheelbendrange(unsigned short int value)
+{
pitchwheel.bendrange=value;
};
-void Controller::setexpression(int value){
+void Controller::setexpression(int value)
+{
expression.data=value;
if (expression.receive!=0) expression.relvolume=value/127.0;
- else expression.relvolume=1.0;
+ else expression.relvolume=1.0;
};
-void Controller::setpanning(int value){
+void Controller::setpanning(int value)
+{
panning.data=value;
panning.pan=(value/128.0-0.5)*(panning.depth/64.0);
};
-void Controller::setfiltercutoff(int value){
+void Controller::setfiltercutoff(int value)
+{
filtercutoff.data=value;
filtercutoff.relfreq=(value-64.0)*filtercutoff.depth/4096.0*3.321928;//3.3219..=ln2(10)
};
-void Controller::setfilterq(int value){
+void Controller::setfilterq(int value)
+{
filterq.data=value;
filterq.relq=pow(30.0,(value-64.0)/64.0*(filterq.depth/64.0));
};
-void Controller::setbandwidth(int value){
+void Controller::setbandwidth(int value)
+{
bandwidth.data=value;
if (bandwidth.exponential==0) {
- REALTYPE tmp=pow(25.0,pow(bandwidth.depth/127.0,1.5))-1.0;
- if ((value<64)&&(bandwidth.depth>=64)) tmp=1.0;
- bandwidth.relbw=(value/64.0-1.0)*tmp+1.0;
- if (bandwidth.relbw<0.01) bandwidth.relbw=0.01;
+ REALTYPE tmp=pow(25.0,pow(bandwidth.depth/127.0,1.5))-1.0;
+ if ((value<64)&&(bandwidth.depth>=64)) tmp=1.0;
+ bandwidth.relbw=(value/64.0-1.0)*tmp+1.0;
+ if (bandwidth.relbw<0.01) bandwidth.relbw=0.01;
} else {
- bandwidth.relbw=pow(25.0,(value-64.0)/64.0*(bandwidth.depth/64.0));
+ bandwidth.relbw=pow(25.0,(value-64.0)/64.0*(bandwidth.depth/64.0));
};
};
-void Controller::setmodwheel(int value){
+void Controller::setmodwheel(int value)
+{
modwheel.data=value;
if (modwheel.exponential==0) {
- REALTYPE tmp=pow(25.0,pow(modwheel.depth/127.0,1.5)*2.0)/25.0;
- if ((value<64)&&(modwheel.depth>=64)) tmp=1.0;
- modwheel.relmod=(value/64.0-1.0)*tmp+1.0;
- if (modwheel.relmod<0.0) modwheel.relmod=0.0;
- } else modwheel.relmod=pow(25.0,(value-64.0)/64.0*(modwheel.depth/80.0));
+ REALTYPE tmp=pow(25.0,pow(modwheel.depth/127.0,1.5)*2.0)/25.0;
+ if ((value<64)&&(modwheel.depth>=64)) tmp=1.0;
+ modwheel.relmod=(value/64.0-1.0)*tmp+1.0;
+ if (modwheel.relmod<0.0) modwheel.relmod=0.0;
+ } else modwheel.relmod=pow(25.0,(value-64.0)/64.0*(modwheel.depth/80.0));
};
-void Controller::setfmamp(int value){
+void Controller::setfmamp(int value)
+{
fmamp.data=value;
fmamp.relamp=value/127.0;
if (fmamp.receive!=0) fmamp.relamp=value/127.0;
- else fmamp.relamp=1.0;
+ else fmamp.relamp=1.0;
};
-void Controller::setvolume(int value){
+void Controller::setvolume(int value)
+{
volume.data=value;
if (volume.receive!=0) volume.volume=pow(0.1,(127-value)/127.0*2.0);
- else volume.volume=1.0;
+ else volume.volume=1.0;
};
-void Controller::setsustain(int value){
+void Controller::setsustain(int value)
+{
sustain.data=value;
if (sustain.receive!=0) sustain.sustain=((value<64) ? 0 : 1 );
- else sustain.sustain=0;
+ else sustain.sustain=0;
};
-void Controller::setportamento(int value){
+void Controller::setportamento(int value)
+{
portamento.data=value;
if (portamento.receive!=0) portamento.portamento=((value<64) ? 0 : 1 );
};
// I added a third argument to pass legato status,
// when legatoflag is true it means "there's a legato in progress".
-int Controller::initportamento(REALTYPE oldfreq,REALTYPE newfreq,bool legatoflag){
+int Controller::initportamento(REALTYPE oldfreq,REALTYPE newfreq,bool legatoflag)
+{
portamento.x=0.0;
- if (legatoflag){ // Legato in progress
- if (portamento.portamento==0) return(0);
+ if (legatoflag) { // Legato in progress
+ if (portamento.portamento==0) return(0);
} else { // No legato, do the original if...return
- if ((portamento.used!=0) || (portamento.portamento==0)) return(0);
+ if ((portamento.used!=0) || (portamento.portamento==0)) return(0);
};
REALTYPE portamentotime=pow(100.0,portamento.time/127.0)/50.0;//portamento time in seconds
- if ((portamento.updowntimestretch>=64)&&(newfreq<oldfreq)){
- if (portamento.updowntimestretch==127) return(0);
- portamentotime*=pow(0.1,(portamento.updowntimestretch-64)/63.0);
- }
- if ((portamento.updowntimestretch<64)&&(newfreq>oldfreq)){
- if (portamento.updowntimestretch==0) return(0);
- portamentotime*=pow(0.1,(64.0-portamento.updowntimestretch)/64.0);
+ if ((portamento.updowntimestretch>=64)&&(newfreq<oldfreq)) {
+ if (portamento.updowntimestretch==127) return(0);
+ portamentotime*=pow(0.1,(portamento.updowntimestretch-64)/63.0);
+ }
+ if ((portamento.updowntimestretch<64)&&(newfreq>oldfreq)) {
+ if (portamento.updowntimestretch==0) return(0);
+ portamentotime*=pow(0.1,(64.0-portamento.updowntimestretch)/64.0);
};
-
+
portamento.dx=SOUND_BUFFER_SIZE/(portamentotime*SAMPLE_RATE);
portamento.origfreqrap=oldfreq/newfreq;
-
- REALTYPE tmprap=( (portamento.origfreqrap>1.0) ?
- (portamento.origfreqrap) :
- (1.0/portamento.origfreqrap) );
-
+
+ REALTYPE tmprap=( (portamento.origfreqrap>1.0) ?
+ (portamento.origfreqrap) :
+ (1.0/portamento.origfreqrap) );
+
REALTYPE thresholdrap=pow(2.0,portamento.pitchthresh/12.0);
if ((portamento.pitchthreshtype==0) && (tmprap-0.00001>thresholdrap) ) return(0);
if ((portamento.pitchthreshtype==1) && (tmprap+0.00001<thresholdrap) ) return(0);
@@ -201,34 +218,38 @@ int Controller::initportamento(REALTYPE oldfreq,REALTYPE newfreq,bool legatoflag
return (1);
};
-void Controller::updateportamento(){
+void Controller::updateportamento()
+{
if (portamento.used==0) return;
-
+
portamento.x+=portamento.dx;
if (portamento.x>1.0) {
- portamento.x=1.0;
- portamento.used=0;
+ portamento.x=1.0;
+ portamento.used=0;
};
portamento.freqrap=(1.0-portamento.x)*portamento.origfreqrap+portamento.x;
};
-void Controller::setresonancecenter(int value){
+void Controller::setresonancecenter(int value)
+{
resonancecenter.data=value;
resonancecenter.relcenter=pow(3.0,(value-64.0)/64.0*(resonancecenter.depth/64.0));
};
-void Controller::setresonancebw(int value){
+void Controller::setresonancebw(int value)
+{
resonancebandwidth.data=value;
resonancebandwidth.relbw=pow(1.5,(value-64.0)/64.0*(resonancebandwidth.depth/127.0));
};
//Returns 0 if there is NRPN or 1 if there is not
-int Controller::getnrpn(int *parhi, int *parlo, int *valhi, int *vallo){
+int Controller::getnrpn(int *parhi, int *parlo, int *valhi, int *vallo)
+{
if (NRPN.receive==0) return(1);
- if ((NRPN.parhi<0)||(NRPN.parlo<0)||(NRPN.valhi<0)||(NRPN.vallo<0))
- return(1);
-
+ if ((NRPN.parhi<0)||(NRPN.parlo<0)||(NRPN.valhi<0)||(NRPN.vallo<0))
+ return(1);
+
*parhi=NRPN.parhi;
*parlo=NRPN.parlo;
*valhi=NRPN.valhi;
@@ -237,24 +258,32 @@ 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;
+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;
};
};
-void Controller::add2XML(XMLwrapper *xml){
+void Controller::add2XML(XMLwrapper *xml)
+{
xml->addpar("pitchwheel_bendrange",pitchwheel.bendrange);
xml->addparbool("expression_receive",expression.receive);
@@ -279,7 +308,8 @@ void Controller::add2XML(XMLwrapper *xml){
xml->addpar("resonance_bandwidth_depth",resonancebandwidth.depth);
};
-void Controller::getfromXML(XMLwrapper *xml){
+void Controller::getfromXML(XMLwrapper *xml)
+{
pitchwheel.bendrange=xml->getpar("pitchwheel_bendrange",pitchwheel.bendrange,-6400,6400);
expression.receive=xml->getparbool("expression_receive",expression.receive);
diff --git a/src/Params/Controller.h b/src/Params/Controller.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Controller.h - (Midi) Controllers implementation
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,152 +28,153 @@
#include "../Misc/XMLwrapper.h"
/**(Midi) Controllers implementation*/
-class Controller{
- public:
- Controller();
- ~Controller();
- void resetall();
+class Controller
+{
+public:
+ Controller();
+ ~Controller();
+ void resetall();
- void add2XML(XMLwrapper *xml);
- void defaults();
- void getfromXML(XMLwrapper *xml);
+ void add2XML(XMLwrapper *xml);
+ void defaults();
+ void getfromXML(XMLwrapper *xml);
//Controllers functions
- void setpitchwheel(int value);
- void setpitchwheelbendrange(unsigned short int value);
- void setexpression(int value);
- void setpanning(int value);
- void setfiltercutoff(int value);
- void setfilterq(int value);
- void setbandwidth(int value);
- void setmodwheel(int value);
- void setfmamp(int value);
- void setvolume(int value);
- void setsustain(int value);
- void setportamento(int value);
- void setresonancecenter(int value);
- void setresonancebw(int value);
-
-
- void setparameternumber(unsigned int type,int value);//used for RPN and NRPN's
- int getnrpn(int *parhi, int *parlo, int *valhi, int *vallo);
-
- int initportamento(REALTYPE oldfreq,REALTYPE newfreq,bool legatoflag);//returns 1 if the portamento's conditions are true, else return 0
- void updateportamento(); //update portamento values
-
- // Controllers values
+ void setpitchwheel(int value);
+ void setpitchwheelbendrange(unsigned short int value);
+ void setexpression(int value);
+ void setpanning(int value);
+ void setfiltercutoff(int value);
+ void setfilterq(int value);
+ void setbandwidth(int value);
+ void setmodwheel(int value);
+ void setfmamp(int value);
+ void setvolume(int value);
+ void setsustain(int value);
+ void setportamento(int value);
+ void setresonancecenter(int value);
+ void setresonancebw(int value);
+
+
+ void setparameternumber(unsigned int type,int value);//used for RPN and NRPN's
+ int getnrpn(int *parhi, int *parlo, int *valhi, int *vallo);
+
+ int initportamento(REALTYPE oldfreq,REALTYPE newfreq,bool legatoflag);//returns 1 if the portamento's conditions are true, else return 0
+ void updateportamento(); //update portamento values
+
+ // Controllers values
struct {//Pitch Wheel
- int data;
- short int bendrange;//bendrange is in cents
- REALTYPE relfreq;//the relative frequency (default is 1.0)
+ int data;
+ short int bendrange;//bendrange is in cents
+ REALTYPE relfreq;//the relative frequency (default is 1.0)
} pitchwheel;
-
- struct{//Expression
- int data;
- REALTYPE relvolume;
- unsigned char receive;
+
+ struct {//Expression
+ int data;
+ REALTYPE relvolume;
+ unsigned char receive;
} expression;
- struct{//Panning
- int data;
- REALTYPE pan;
- unsigned char depth;
+ struct {//Panning
+ int data;
+ REALTYPE pan;
+ unsigned char depth;
} panning;
-
- struct{//Filter cutoff
- int data;
- REALTYPE relfreq;
- unsigned char depth;
+
+ struct {//Filter cutoff
+ int data;
+ REALTYPE relfreq;
+ unsigned char depth;
} filtercutoff;
- struct{//Filter Q
- int data;
- REALTYPE relq;
- unsigned char depth;
+ struct {//Filter Q
+ int data;
+ REALTYPE relq;
+ unsigned char depth;
} filterq;
- struct{//Bandwidth
- int data;
- REALTYPE relbw;
- unsigned char depth;
- unsigned char exponential;
+ struct {//Bandwidth
+ int data;
+ REALTYPE relbw;
+ unsigned char depth;
+ unsigned char exponential;
} bandwidth;
struct {//Modulation Wheel
- int data;
- REALTYPE relmod;
- unsigned char depth;
- unsigned char exponential;
+ int data;
+ REALTYPE relmod;
+ unsigned char depth;
+ unsigned char exponential;
} modwheel;
- struct{//FM amplitude
- int data;
- REALTYPE relamp;
- unsigned char receive;
+ struct {//FM amplitude
+ int data;
+ REALTYPE relamp;
+ unsigned char receive;
} fmamp;
- struct{//Volume
- int data;
- REALTYPE volume;
- unsigned char receive;
+ struct {//Volume
+ int data;
+ REALTYPE volume;
+ unsigned char receive;
} volume;
- struct{//Sustain
- int data,sustain;
- unsigned char receive;
+ struct {//Sustain
+ int data,sustain;
+ unsigned char receive;
} sustain;
- struct{/**<Portamento*/
- //parameters
- int data;
- unsigned char portamento;
-
- unsigned char receive,time;
- /**pitchthresh is the threshold of enabling protamento \todo see if this should be an int*/
- unsigned char pitchthresh;
- /**pitchthreshtype -> enable the portamento only below(0)/above(1) the threshold*/
- unsigned char pitchthreshtype;
+ struct {/**<Portamento*/
+ //parameters
+ int data;
+ unsigned char portamento;
+
+ unsigned char receive,time;
+ /**pitchthresh is the threshold of enabling protamento \todo see if this should be an int*/
+ unsigned char pitchthresh;
+ /**pitchthreshtype -> enable the portamento only below(0)/above(1) the threshold*/
+ unsigned char pitchthreshtype;
/**this value represent how the portamento time is reduced
* 0 - for down portamento, 1..63 - the up portamento's time is smaller than the down portamento
- * 64 - the portamento time is always the same
- * 64-126 - the down portamento's time is smaller than the up portamento
- * 127 - for upper portamento
+ * 64 - the portamento time is always the same
+ * 64-126 - the down portamento's time is smaller than the up portamento
+ * 127 - for upper portamento
* 'up portanemto' means when the frequency is rising (eg: the portamento is from 200Hz to 300 Hz)
* 'down portanemto' means when the frequency is lowering (eg: the portamento is from 300Hz to 200 Hz)
*/
- unsigned char updowntimestretch;
-
- REALTYPE freqrap;/**<this value is used to compute the actual portamento*/
- int noteusing;/**this is used by the Part for knowing which note uses the portamento*/
- int used;/**<if a the portamento is used by a note \todo see if this can be a bool*/
- //internal data
- REALTYPE x,dx;//x is from 0.0 (start portamento) to 1.0 (finished portamento), dx is x increment
- REALTYPE origfreqrap;// this is used for computing oldfreq value from x
+ unsigned char updowntimestretch;
+
+ REALTYPE freqrap;/**<this value is used to compute the actual portamento*/
+ int noteusing;/**this is used by the Part for knowing which note uses the portamento*/
+ int used;/**<if a the portamento is used by a note \todo see if this can be a bool*/
+ //internal data
+ REALTYPE x,dx;//x is from 0.0 (start portamento) to 1.0 (finished portamento), dx is x increment
+ REALTYPE origfreqrap;// this is used for computing oldfreq value from x
} portamento;
-
- struct{//Resonance Center Frequency
- int data;
- REALTYPE relcenter;
- unsigned char depth;
+
+ struct {//Resonance Center Frequency
+ int data;
+ REALTYPE relcenter;
+ unsigned char depth;
} resonancecenter;
- struct{//Resonance Bandwidth
- int data;
- REALTYPE relbw;
- unsigned char depth;
+ struct {//Resonance Bandwidth
+ int data;
+ REALTYPE relbw;
+ unsigned char depth;
} resonancebandwidth;
-
+
/** RPN and NPRPN */
- struct{//nrpn
- int parhi,parlo;
- int valhi,vallo;
- unsigned char receive;//this is saved to disk by Master
+ struct {//nrpn
+ int parhi,parlo;
+ int valhi,vallo;
+ unsigned char receive;//this is saved to disk by Master
} NRPN;
-
- private:
+
+private:
};
#endif
diff --git a/src/Params/EnvelopeParams.C b/src/Params/EnvelopeParams.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EnvelopeParams.C - Parameters for Envelope
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,31 +26,40 @@
#include <stdlib.h>
#include "EnvelopeParams.h"
-EnvelopeParams::EnvelopeParams(unsigned char Penvstretch_,unsigned char Pforcedrelease_):Presets(){
+EnvelopeParams::EnvelopeParams(unsigned char Penvstretch_,unsigned char Pforcedrelease_):Presets()
+{
int i;
-
- PA_dt=10;PD_dt=10;PR_dt=10;PA_val=64;PD_val=64;PS_val=64;PR_val=64;
-
- for (i=0;i<MAX_ENVELOPE_POINTS;i++){
- Penvdt[i]=32;
- Penvval[i]=64;
+
+ PA_dt=10;
+ PD_dt=10;
+ PR_dt=10;
+ PA_val=64;
+ PD_val=64;
+ PS_val=64;
+ PR_val=64;
+
+ for (i=0;i<MAX_ENVELOPE_POINTS;i++) {
+ Penvdt[i]=32;
+ Penvval[i]=64;
};
Penvdt[0]=0;//no used
Penvsustain=1;
Penvpoints=1;
Envmode=1;
Penvstretch=Penvstretch_;
- Pforcedrelease=Pforcedrelease_;
+ Pforcedrelease=Pforcedrelease_;
Pfreemode=1;
Plinearenvelope=0;
-
+
store2defaults();
};
-EnvelopeParams::~EnvelopeParams(){
+EnvelopeParams::~EnvelopeParams()
+{
};
-REALTYPE EnvelopeParams::getdt(char i){
+REALTYPE EnvelopeParams::getdt(char i)
+{
REALTYPE result=(pow(2.0,Penvdt[(int)i]/127.0*12.0)-1.0)*10.0;//miliseconds
return(result);
};
@@ -59,49 +68,71 @@ REALTYPE EnvelopeParams::getdt(char i){
/*
* ADSR/ASR... initialisations
*/
-void EnvelopeParams::ADSRinit(char A_dt,char D_dt,char S_val,char R_dt){
+void EnvelopeParams::ADSRinit(char A_dt,char D_dt,char S_val,char R_dt)
+{
setpresettype("Penvamplitude");
Envmode=1;
- PA_dt=A_dt;PD_dt=D_dt;PS_val=S_val;PR_dt=R_dt;
+ PA_dt=A_dt;
+ PD_dt=D_dt;
+ PS_val=S_val;
+ PR_dt=R_dt;
Pfreemode=0;
converttofree();
store2defaults();
};
-void EnvelopeParams::ADSRinit_dB(char A_dt,char D_dt,char S_val,char R_dt){
+void EnvelopeParams::ADSRinit_dB(char A_dt,char D_dt,char S_val,char R_dt)
+{
setpresettype("Penvamplitude");
Envmode=2;
- PA_dt=A_dt;PD_dt=D_dt;PS_val=S_val;PR_dt=R_dt;
+ PA_dt=A_dt;
+ PD_dt=D_dt;
+ PS_val=S_val;
+ PR_dt=R_dt;
Pfreemode=0;
converttofree();
store2defaults();
};
-void EnvelopeParams::ASRinit(char A_val,char A_dt,char R_val,char R_dt){
+void EnvelopeParams::ASRinit(char A_val,char A_dt,char R_val,char R_dt)
+{
setpresettype("Penvfrequency");
Envmode=3;
- PA_val=A_val;PA_dt=A_dt;PR_val=R_val;PR_dt=R_dt;
+ PA_val=A_val;
+ PA_dt=A_dt;
+ PR_val=R_val;
+ PR_dt=R_dt;
Pfreemode=0;
converttofree();
store2defaults();
};
-void EnvelopeParams::ADSRinit_filter(char A_val,char A_dt,char D_val,char D_dt,char R_dt,char R_val){
+void EnvelopeParams::ADSRinit_filter(char A_val,char A_dt,char D_val,char D_dt,char R_dt,char R_val)
+{
setpresettype("Penvfilter");
Envmode=4;
- PA_val=A_val;PA_dt=A_dt;PD_val=D_val;PD_dt=D_dt;PR_dt=R_dt;PR_val=R_val;
+ PA_val=A_val;
+ PA_dt=A_dt;
+ PD_val=D_val;
+ PD_dt=D_dt;
+ PR_dt=R_dt;
+ PR_val=R_val;
Pfreemode=0;
converttofree();
store2defaults();
};
-void EnvelopeParams::ASRinit_bw(char A_val,char A_dt,char R_val,char R_dt){
+void EnvelopeParams::ASRinit_bw(char A_val,char A_dt,char R_val,char R_dt)
+{
setpresettype("Penvbandwidth");
Envmode=5;
- PA_val=A_val;PA_dt=A_dt;PR_val=R_val;PR_dt=R_dt;
+ PA_val=A_val;
+ PA_dt=A_dt;
+ PR_val=R_val;
+ PR_dt=R_dt;
Pfreemode=0;
converttofree();
store2defaults();
@@ -110,39 +141,69 @@ void EnvelopeParams::ASRinit_bw(char A_val,char A_dt,char R_val,char R_dt){
/*
* Convert the Envelope to freemode
*/
-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;
+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;
};
};
-void EnvelopeParams::add2XML(XMLwrapper *xml){
- xml->addparbool("free_mode",Pfreemode);
+void EnvelopeParams::add2XML(XMLwrapper *xml)
+{
+ xml->addparbool("free_mode",Pfreemode);
xml->addpar("env_points",Penvpoints);
xml->addpar("env_sustain",Penvsustain);
xml->addpar("env_stretch",Penvstretch);
@@ -156,20 +217,21 @@ void EnvelopeParams::add2XML(XMLwrapper *xml){
xml->addpar("S_val",PS_val);
xml->addpar("R_val",PR_val);
- if ((Pfreemode!=0)||(!xml->minimal)){
- for (int i=0;i<Penvpoints;i++){
- xml->beginbranch("POINT",i);
- if (i!=0) xml->addpar("dt",Penvdt[i]);
- xml->addpar("val",Penvval[i]);
- xml->endbranch();
- };
+ if ((Pfreemode!=0)||(!xml->minimal)) {
+ for (int i=0;i<Penvpoints;i++) {
+ xml->beginbranch("POINT",i);
+ if (i!=0) xml->addpar("dt",Penvdt[i]);
+ xml->addpar("val",Penvval[i]);
+ xml->endbranch();
+ };
};
};
-void EnvelopeParams::getfromXML(XMLwrapper *xml){
- Pfreemode=xml->getparbool("free_mode",Pfreemode);
+void EnvelopeParams::getfromXML(XMLwrapper *xml)
+{
+ Pfreemode=xml->getparbool("free_mode",Pfreemode);
Penvpoints=xml->getpar127("env_points",Penvpoints);
Penvsustain=xml->getpar127("env_sustain",Penvsustain);
Penvstretch=xml->getpar127("env_stretch",Penvstretch);
@@ -184,18 +246,19 @@ void EnvelopeParams::getfromXML(XMLwrapper *xml){
PS_val=xml->getpar127("S_val",PS_val);
PR_val=xml->getpar127("R_val",PR_val);
- for (int i=0;i<Penvpoints;i++){
+ for (int i=0;i<Penvpoints;i++) {
if (xml->enterbranch("POINT",i)==0) continue;
- if (i!=0) Penvdt[i]=xml->getpar127("dt",Penvdt[i]);
- Penvval[i]=xml->getpar127("val",Penvval[i]);
- xml->exitbranch();
+ if (i!=0) Penvdt[i]=xml->getpar127("dt",Penvdt[i]);
+ Penvval[i]=xml->getpar127("val",Penvval[i]);
+ xml->exitbranch();
};
-
+
if (!Pfreemode) converttofree();
};
-void EnvelopeParams::defaults(){
+void EnvelopeParams::defaults()
+{
Penvstretch=Denvstretch;
Pforcedrelease=Dforcedrelease;
Plinearenvelope=Dlinearenvelope;
@@ -210,7 +273,8 @@ void EnvelopeParams::defaults(){
converttofree();
};
-void EnvelopeParams::store2defaults(){
+void EnvelopeParams::store2defaults()
+{
Denvstretch=Penvstretch;
Dforcedrelease=Pforcedrelease;
Dlinearenvelope=Plinearenvelope;
diff --git a/src/Params/EnvelopeParams.h b/src/Params/EnvelopeParams.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EnvelopeParams.h - Parameters for Envelope
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -30,53 +30,54 @@
#define MAX_ENVELOPE_POINTS 40
#define MIN_ENVELOPE_DB -40
-class EnvelopeParams:public Presets{
- public:
- EnvelopeParams(unsigned char Penvstretch_,unsigned char Pforcedrelease_);
- ~EnvelopeParams();
- void ADSRinit(char A_dt,char D_dt,char S_val,char R_dt);
- void ADSRinit_dB(char A_dt,char D_dt,char S_val,char R_dt);
- void ASRinit(char A_val,char A_dt,char R_val,char R_dt);
- void ADSRinit_filter(char A_val,char A_dt,char D_val,char D_dt,char R_dt,char R_val);
- void ASRinit_bw(char A_val,char A_dt,char R_val,char R_dt);
- void converttofree();
-
- void add2XML(XMLwrapper *xml);
- void defaults();
- void getfromXML(XMLwrapper *xml);
-
- REALTYPE getdt(char i);
-
- /* MIDI Parameters */
- unsigned char Pfreemode;//1 daca este in modul free sau 0 daca este in mod ADSR,ASR,...
- unsigned char Penvpoints;
- unsigned char Penvsustain;//127 pentru dezactivat
- unsigned char Penvdt[MAX_ENVELOPE_POINTS];
- unsigned char Penvval[MAX_ENVELOPE_POINTS];
- unsigned char Penvstretch;//64=normal stretch (piano-like), 0=no stretch
- unsigned char Pforcedrelease;//0 - OFF, 1 - ON
- unsigned char Plinearenvelope;//if the amplitude envelope is linear
-
- unsigned char PA_dt,PD_dt,PR_dt,
- PA_val,PD_val,PS_val,PR_val;
-
-
-
- int Envmode;// 1 for ADSR parameters (linear amplitude)
- // 2 for ADSR_dB parameters (dB amplitude)
- // 3 for ASR parameters (frequency LFO)
- // 4 for ADSR_filter parameters (filter parameters)
- // 5 for ASR_bw parameters (bandwidth parameters)
-
- private:
- void store2defaults();
-
- /* Default parameters */
- unsigned char Denvstretch;
- unsigned char Dforcedrelease;
- unsigned char Dlinearenvelope;
- unsigned char DA_dt,DD_dt,DR_dt,
- DA_val,DD_val,DS_val,DR_val;
+class EnvelopeParams:public Presets
+{
+public:
+ EnvelopeParams(unsigned char Penvstretch_,unsigned char Pforcedrelease_);
+ ~EnvelopeParams();
+ void ADSRinit(char A_dt,char D_dt,char S_val,char R_dt);
+ void ADSRinit_dB(char A_dt,char D_dt,char S_val,char R_dt);
+ void ASRinit(char A_val,char A_dt,char R_val,char R_dt);
+ void ADSRinit_filter(char A_val,char A_dt,char D_val,char D_dt,char R_dt,char R_val);
+ void ASRinit_bw(char A_val,char A_dt,char R_val,char R_dt);
+ void converttofree();
+
+ void add2XML(XMLwrapper *xml);
+ void defaults();
+ void getfromXML(XMLwrapper *xml);
+
+ REALTYPE getdt(char i);
+
+ /* MIDI Parameters */
+ unsigned char Pfreemode;//1 daca este in modul free sau 0 daca este in mod ADSR,ASR,...
+ unsigned char Penvpoints;
+ unsigned char Penvsustain;//127 pentru dezactivat
+ unsigned char Penvdt[MAX_ENVELOPE_POINTS];
+ unsigned char Penvval[MAX_ENVELOPE_POINTS];
+ unsigned char Penvstretch;//64=normal stretch (piano-like), 0=no stretch
+ unsigned char Pforcedrelease;//0 - OFF, 1 - ON
+ unsigned char Plinearenvelope;//if the amplitude envelope is linear
+
+ unsigned char PA_dt,PD_dt,PR_dt,
+ PA_val,PD_val,PS_val,PR_val;
+
+
+
+ int Envmode;// 1 for ADSR parameters (linear amplitude)
+ // 2 for ADSR_dB parameters (dB amplitude)
+ // 3 for ASR parameters (frequency LFO)
+ // 4 for ADSR_filter parameters (filter parameters)
+ // 5 for ASR_bw parameters (bandwidth parameters)
+
+private:
+ void store2defaults();
+
+ /* Default parameters */
+ unsigned char Denvstretch;
+ unsigned char Dforcedrelease;
+ unsigned char Dlinearenvelope;
+ unsigned char DA_dt,DD_dt,DR_dt,
+ DA_val,DD_val,DS_val,DR_val;
};
diff --git a/src/Params/FilterParams.C b/src/Params/FilterParams.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
FilterParams.C - Parameters for filter
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,21 +25,24 @@
#include <stdlib.h>
#include "FilterParams.h"
-FilterParams::FilterParams(unsigned char Ptype_,unsigned char Pfreq_,unsigned char Pq_):Presets(){
+FilterParams::FilterParams(unsigned char Ptype_,unsigned char Pfreq_,unsigned char Pq_):Presets()
+{
setpresettype("Pfilter");
Dtype=Ptype_;
Dfreq=Pfreq_;
Dq=Pq_;
changed=false;
- defaults();
+ defaults();
};
-FilterParams::~FilterParams(){
+FilterParams::~FilterParams()
+{
};
-void FilterParams::defaults(){
+void FilterParams::defaults()
+{
Ptype=Dtype;
Pfreq=Dfreq;
Pq=Dq;
@@ -48,13 +51,13 @@ void FilterParams::defaults(){
Pfreqtrack=64;
Pgain=64;
Pcategory=0;
-
+
Pnumformants=3;
Pformantslowness=64;
- for (int j=0;j<FF_MAX_VOWELS;j++){
- defaults(j);
+ for (int j=0;j<FF_MAX_VOWELS;j++) {
+ defaults(j);
};
-
+
Psequencesize=3;
for (int i=0;i<FF_MAX_SEQUENCE;i++) Psequence[i].nvowel=i%FF_MAX_VOWELS;
@@ -65,25 +68,27 @@ void FilterParams::defaults(){
Pvowelclearness=64;
};
-void FilterParams::defaults(int n){
+void FilterParams::defaults(int n)
+{
int j=n;
- for (int i=0;i<FF_MAX_FORMANTS;i++){
- Pvowels[j].formants[i].freq=(int)(RND*127.0);//some random freqs
- Pvowels[j].formants[i].q=64;
- Pvowels[j].formants[i].amp=127;
- };
+ for (int i=0;i<FF_MAX_FORMANTS;i++) {
+ Pvowels[j].formants[i].freq=(int)(RND*127.0);//some random freqs
+ Pvowels[j].formants[i].q=64;
+ Pvowels[j].formants[i].amp=127;
+ };
};
/*
* Get the parameters from other FilterParams
*/
-
-void FilterParams::getfromFilterParams(FilterParams *pars){
+
+void FilterParams::getfromFilterParams(FilterParams *pars)
+{
defaults();
-
+
if (pars==NULL) return;
-
+
Ptype=pars->Ptype;
Pfreq=pars->Pfreq;
Pq=pars->Pq;
@@ -92,17 +97,17 @@ void FilterParams::getfromFilterParams(FilterParams *pars){
Pfreqtrack=pars->Pfreqtrack;
Pgain=pars->Pgain;
Pcategory=pars->Pcategory;
-
+
Pnumformants=pars->Pnumformants;
Pformantslowness=pars->Pformantslowness;
- 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;
- };
+ 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++) Psequence[i].nvowel=pars->Psequence[i].nvowel;
@@ -117,39 +122,46 @@ void FilterParams::getfromFilterParams(FilterParams *pars){
/*
* Parameter control
*/
-REALTYPE FilterParams::getfreq(){
+REALTYPE FilterParams::getfreq()
+{
return((Pfreq/64.0-1.0)*5.0);
};
-REALTYPE FilterParams::getq(){
+REALTYPE FilterParams::getq()
+{
return(exp(pow((REALTYPE) Pq/127.0,2)*log(1000.0))-0.9);
};
-REALTYPE FilterParams::getfreqtracking(REALTYPE notefreq){
+REALTYPE FilterParams::getfreqtracking(REALTYPE notefreq)
+{
return(log(notefreq/440.0)*(Pfreqtrack-64.0)/(64.0*LOG_2));
};
-REALTYPE FilterParams::getgain(){
+REALTYPE FilterParams::getgain()
+{
return((Pgain/64.0-1.0)*30.0);//-30..30dB
};
/*
* Get the center frequency of the formant's graph
*/
-REALTYPE FilterParams::getcenterfreq(){
+REALTYPE FilterParams::getcenterfreq()
+{
return(10000.0*pow(10,-(1.0-Pcenterfreq/127.0)*2.0));
};
/*
* Get the number of octave that the formant functions applies to
*/
-REALTYPE FilterParams::getoctavesfreq(){
+REALTYPE FilterParams::getoctavesfreq()
+{
return(0.25+10.0*Poctavesfreq/127.0);
};
/*
* Get the frequency from x, where x is [0..1]
*/
-REALTYPE FilterParams::getfreqx(REALTYPE x){
+REALTYPE FilterParams::getfreqx(REALTYPE x)
+{
if (x>1.0) x=1.0;
REALTYPE octf=pow(2.0,getoctavesfreq());
return(getcenterfreq()/sqrt(octf)*pow(octf,x));
@@ -158,7 +170,8 @@ REALTYPE FilterParams::getfreqx(REALTYPE x){
/*
* Get the x coordinate from frequency (used by the UI)
*/
-REALTYPE FilterParams::getfreqpos(REALTYPE freq){
+REALTYPE FilterParams::getfreqpos(REALTYPE freq)
+{
return((log(freq)-log(getfreqx(0.0)))/log(2.0)/getoctavesfreq());
};
@@ -166,7 +179,8 @@ REALTYPE FilterParams::getfreqpos(REALTYPE freq){
/*
* Get the freq. response of the formant filter
*/
-void FilterParams::formantfilterH(int nvowel,int nfreqs,REALTYPE *freqs){
+void FilterParams::formantfilterH(int nvowel,int nfreqs,REALTYPE *freqs)
+{
REALTYPE c[3],d[3];
REALTYPE filter_freq,filter_q,filter_amp;
REALTYPE omega,sn,cs,alpha;
@@ -174,55 +188,56 @@ void FilterParams::formantfilterH(int nvowel,int nfreqs,REALTYPE *freqs){
for (int i=0;i<nfreqs;i++) freqs[i]=0.0;
//for each formant...
- for (int nformant=0;nformant<Pnumformants;nformant++){
- //compute formant parameters(frequency,amplitude,etc.)
- filter_freq=getformantfreq(Pvowels[nvowel].formants[nformant].freq);
- filter_q=getformantq(Pvowels[nvowel].formants[nformant].q)*getq();
- if (Pstages>0) filter_q=(filter_q>1.0 ? pow(filter_q,1.0/(Pstages+1)) : filter_q);
-
- filter_amp=getformantamp(Pvowels[nvowel].formants[nformant].amp);
-
-
- if (filter_freq<=(SAMPLE_RATE/2-100.0)){
- omega=2*PI*filter_freq/SAMPLE_RATE;
- sn=sin(omega);
- cs=cos(omega);
- alpha=sn/(2*filter_q);
- REALTYPE 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);
- } else continue;
-
-
- for (int i=0;i<nfreqs;i++) {
- REALTYPE freq=getfreqx(i/(REALTYPE) nfreqs);
- if (freq>SAMPLE_RATE/2) {
- for (int tmp=i;tmp<nfreqs;tmp++) freqs[tmp]=0.0;
- break;
- };
- REALTYPE fr=freq/SAMPLE_RATE*PI*2.0;
- REALTYPE x=c[0],y=0.0;
- for (int n=1;n<3;n++){
- x+=cos(n*fr)*c[n];
- y-=sin(n*fr)*c[n];
- };
- REALTYPE h=x*x+y*y;
- x=1.0;y=0.0;
- for (int n=1;n<3;n++){
- x-=cos(n*fr)*d[n];
- y+=sin(n*fr)*d[n];
- };
- h=h/(x*x+y*y);
-
- freqs[i]+=pow(h,(Pstages+1.0)/2.0)*filter_amp;
- };
+ for (int nformant=0;nformant<Pnumformants;nformant++) {
+ //compute formant parameters(frequency,amplitude,etc.)
+ filter_freq=getformantfreq(Pvowels[nvowel].formants[nformant].freq);
+ filter_q=getformantq(Pvowels[nvowel].formants[nformant].q)*getq();
+ if (Pstages>0) filter_q=(filter_q>1.0 ? pow(filter_q,1.0/(Pstages+1)) : filter_q);
+
+ filter_amp=getformantamp(Pvowels[nvowel].formants[nformant].amp);
+
+
+ if (filter_freq<=(SAMPLE_RATE/2-100.0)) {
+ omega=2*PI*filter_freq/SAMPLE_RATE;
+ sn=sin(omega);
+ cs=cos(omega);
+ alpha=sn/(2*filter_q);
+ REALTYPE 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);
+ } else continue;
+
+
+ for (int i=0;i<nfreqs;i++) {
+ REALTYPE freq=getfreqx(i/(REALTYPE) nfreqs);
+ if (freq>SAMPLE_RATE/2) {
+ for (int tmp=i;tmp<nfreqs;tmp++) freqs[tmp]=0.0;
+ break;
+ };
+ REALTYPE fr=freq/SAMPLE_RATE*PI*2.0;
+ REALTYPE x=c[0],y=0.0;
+ for (int n=1;n<3;n++) {
+ x+=cos(n*fr)*c[n];
+ y-=sin(n*fr)*c[n];
+ };
+ REALTYPE h=x*x+y*y;
+ x=1.0;
+ y=0.0;
+ for (int n=1;n<3;n++) {
+ x-=cos(n*fr)*d[n];
+ y+=sin(n*fr)*d[n];
+ };
+ h=h/(x*x+y*y);
+
+ freqs[i]+=pow(h,(Pstages+1.0)/2.0)*filter_amp;
+ };
};
for (int i=0;i<nfreqs;i++) {
- if (freqs[i]>0.000000001) freqs[i]=rap2dB(freqs[i])+getgain();
- else freqs[i]=-90.0;
+ if (freqs[i]>0.000000001) freqs[i]=rap2dB(freqs[i])+getgain();
+ else freqs[i]=-90.0;
};
};
@@ -230,17 +245,20 @@ void FilterParams::formantfilterH(int nvowel,int nfreqs,REALTYPE *freqs){
/*
* Transforms a parameter to the real value
*/
-REALTYPE FilterParams::getformantfreq(unsigned char freq){
+REALTYPE FilterParams::getformantfreq(unsigned char freq)
+{
REALTYPE result=getfreqx(freq/127.0);
return(result);
};
-REALTYPE FilterParams::getformantamp(unsigned char amp){
+REALTYPE FilterParams::getformantamp(unsigned char amp)
+{
REALTYPE result=pow(0.1,(1.0-amp/127.0)*4.0);
return(result);
};
-REALTYPE FilterParams::getformantq(unsigned char q){
+REALTYPE FilterParams::getformantq(unsigned char q)
+{
//temp
REALTYPE result=pow(25.0,(q-32.0)/64.0);
return(result);
@@ -248,18 +266,20 @@ REALTYPE FilterParams::getformantq(unsigned char q){
-void FilterParams::add2XMLsection(XMLwrapper *xml,int n){
- int nvowel=n;
- for (int nformant=0;nformant<FF_MAX_FORMANTS;nformant++){
- xml->beginbranch("FORMANT",nformant);
- xml->addpar("freq",Pvowels[nvowel].formants[nformant].freq);
- xml->addpar("amp",Pvowels[nvowel].formants[nformant].amp);
- xml->addpar("q",Pvowels[nvowel].formants[nformant].q);
- xml->endbranch();
- };
+void FilterParams::add2XMLsection(XMLwrapper *xml,int n)
+{
+ int nvowel=n;
+ for (int nformant=0;nformant<FF_MAX_FORMANTS;nformant++) {
+ xml->beginbranch("FORMANT",nformant);
+ xml->addpar("freq",Pvowels[nvowel].formants[nformant].freq);
+ xml->addpar("amp",Pvowels[nvowel].formants[nformant].amp);
+ xml->addpar("q",Pvowels[nvowel].formants[nformant].q);
+ xml->endbranch();
+ };
};
-void FilterParams::add2XML(XMLwrapper *xml){
+void FilterParams::add2XML(XMLwrapper *xml)
+{
//filter parameters
xml->addpar("category",Pcategory);
xml->addpar("type",Ptype);
@@ -270,43 +290,45 @@ void FilterParams::add2XML(XMLwrapper *xml){
xml->addpar("gain",Pgain);
//formant filter parameters
- if ((Pcategory==1)||(!xml->minimal)){
- xml->beginbranch("FORMANT_FILTER");
- xml->addpar("num_formants",Pnumformants);
- xml->addpar("formant_slowness",Pformantslowness);
- xml->addpar("vowel_clearness",Pvowelclearness);
- xml->addpar("center_freq",Pcenterfreq);
- xml->addpar("octaves_freq",Poctavesfreq);
- for (int nvowel=0;nvowel<FF_MAX_VOWELS;nvowel++){
- xml->beginbranch("VOWEL",nvowel);
- add2XMLsection(xml,nvowel);
- xml->endbranch();
- };
- xml->addpar("sequence_size",Psequencesize);
- xml->addpar("sequence_stretch",Psequencestretch);
- xml->addparbool("sequence_reversed",Psequencereversed);
- for (int nseq=0;nseq<FF_MAX_SEQUENCE;nseq++){
- xml->beginbranch("SEQUENCE_POS",nseq);
- xml->addpar("vowel_id",Psequence[nseq].nvowel);
- xml->endbranch();
- };
- xml->endbranch();
+ if ((Pcategory==1)||(!xml->minimal)) {
+ xml->beginbranch("FORMANT_FILTER");
+ xml->addpar("num_formants",Pnumformants);
+ xml->addpar("formant_slowness",Pformantslowness);
+ xml->addpar("vowel_clearness",Pvowelclearness);
+ xml->addpar("center_freq",Pcenterfreq);
+ xml->addpar("octaves_freq",Poctavesfreq);
+ for (int nvowel=0;nvowel<FF_MAX_VOWELS;nvowel++) {
+ xml->beginbranch("VOWEL",nvowel);
+ add2XMLsection(xml,nvowel);
+ xml->endbranch();
+ };
+ xml->addpar("sequence_size",Psequencesize);
+ xml->addpar("sequence_stretch",Psequencestretch);
+ xml->addparbool("sequence_reversed",Psequencereversed);
+ for (int nseq=0;nseq<FF_MAX_SEQUENCE;nseq++) {
+ xml->beginbranch("SEQUENCE_POS",nseq);
+ xml->addpar("vowel_id",Psequence[nseq].nvowel);
+ xml->endbranch();
+ };
+ xml->endbranch();
};
};
-void FilterParams::getfromXMLsection(XMLwrapper *xml,int n){
- int nvowel=n;
- for (int nformant=0;nformant<FF_MAX_FORMANTS;nformant++){
- if (xml->enterbranch("FORMANT",nformant)==0) continue;
- Pvowels[nvowel].formants[nformant].freq=xml->getpar127("freq",Pvowels[nvowel].formants[nformant].freq);
- Pvowels[nvowel].formants[nformant].amp=xml->getpar127("amp",Pvowels[nvowel].formants[nformant].amp);
- Pvowels[nvowel].formants[nformant].q=xml->getpar127("q",Pvowels[nvowel].formants[nformant].q);
- xml->exitbranch();
- };
+void FilterParams::getfromXMLsection(XMLwrapper *xml,int n)
+{
+ int nvowel=n;
+ for (int nformant=0;nformant<FF_MAX_FORMANTS;nformant++) {
+ if (xml->enterbranch("FORMANT",nformant)==0) continue;
+ Pvowels[nvowel].formants[nformant].freq=xml->getpar127("freq",Pvowels[nvowel].formants[nformant].freq);
+ Pvowels[nvowel].formants[nformant].amp=xml->getpar127("amp",Pvowels[nvowel].formants[nformant].amp);
+ Pvowels[nvowel].formants[nformant].q=xml->getpar127("q",Pvowels[nvowel].formants[nformant].q);
+ xml->exitbranch();
+ };
};
-void FilterParams::getfromXML(XMLwrapper *xml){
+void FilterParams::getfromXML(XMLwrapper *xml)
+{
//filter parameters
Pcategory=xml->getpar127("category",Pcategory);
Ptype=xml->getpar127("type",Ptype);
@@ -317,28 +339,28 @@ void FilterParams::getfromXML(XMLwrapper *xml){
Pgain=xml->getpar127("gain",Pgain);
//formant filter parameters
- if(xml->enterbranch("FORMANT_FILTER")){
- Pnumformants=xml->getpar127("num_formants",Pnumformants);
- Pformantslowness=xml->getpar127("formant_slowness",Pformantslowness);
- Pvowelclearness=xml->getpar127("vowel_clearness",Pvowelclearness);
- Pcenterfreq=xml->getpar127("center_freq",Pcenterfreq);
- Poctavesfreq=xml->getpar127("octaves_freq",Poctavesfreq);
-
- for (int nvowel=0;nvowel<FF_MAX_VOWELS;nvowel++){
- if (xml->enterbranch("VOWEL",nvowel)==0) continue;
- getfromXMLsection(xml,nvowel);
- xml->exitbranch();
- };
- Psequencesize=xml->getpar127("sequence_size",Psequencesize);
- Psequencestretch=xml->getpar127("sequence_stretch",Psequencestretch);
- Psequencereversed=xml->getparbool("sequence_reversed",Psequencereversed);
- for (int nseq=0;nseq<FF_MAX_SEQUENCE;nseq++){
- if (xml->enterbranch("SEQUENCE_POS",nseq)==0) continue;
- Psequence[nseq].nvowel=xml->getpar("vowel_id",Psequence[nseq].nvowel,0,FF_MAX_VOWELS-1);
- xml->exitbranch();
- };
- xml->exitbranch();
+ if (xml->enterbranch("FORMANT_FILTER")) {
+ Pnumformants=xml->getpar127("num_formants",Pnumformants);
+ Pformantslowness=xml->getpar127("formant_slowness",Pformantslowness);
+ Pvowelclearness=xml->getpar127("vowel_clearness",Pvowelclearness);
+ Pcenterfreq=xml->getpar127("center_freq",Pcenterfreq);
+ Poctavesfreq=xml->getpar127("octaves_freq",Poctavesfreq);
+
+ for (int nvowel=0;nvowel<FF_MAX_VOWELS;nvowel++) {
+ if (xml->enterbranch("VOWEL",nvowel)==0) continue;
+ getfromXMLsection(xml,nvowel);
+ xml->exitbranch();
+ };
+ Psequencesize=xml->getpar127("sequence_size",Psequencesize);
+ Psequencestretch=xml->getpar127("sequence_stretch",Psequencestretch);
+ Psequencereversed=xml->getparbool("sequence_reversed",Psequencereversed);
+ for (int nseq=0;nseq<FF_MAX_SEQUENCE;nseq++) {
+ if (xml->enterbranch("SEQUENCE_POS",nseq)==0) continue;
+ Psequence[nseq].nvowel=xml->getpar("vowel_id",Psequence[nseq].nvowel,0,FF_MAX_VOWELS-1);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
-
+
};
diff --git a/src/Params/FilterParams.h b/src/Params/FilterParams.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
FilterParams.h - Parameters for filter
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,73 +27,74 @@
#include "../Misc/XMLwrapper.h"
#include "Presets.h"
-class FilterParams:public Presets{
- public:
- FilterParams(unsigned char Ptype_,unsigned char Pfreq,unsigned char Pq_);
- ~FilterParams();
-
- void add2XML(XMLwrapper *xml);
- void add2XMLsection(XMLwrapper *xml,int n);
- void defaults();
- void getfromXML(XMLwrapper *xml);
- void getfromXMLsection(XMLwrapper *xml,int n);
-
-
- void getfromFilterParams(FilterParams *pars);
-
- REALTYPE getfreq();
- REALTYPE getq();
- REALTYPE getfreqtracking(REALTYPE notefreq);
- REALTYPE getgain();
-
- unsigned char Pcategory;//Filter category (Analog/Formant/StVar)
- unsigned char Ptype;// Filter type (for analog lpf,hpf,bpf..)
- unsigned char Pfreq;// Frequency (64-central frequency)
- unsigned char Pq; // Q parameters (resonance or bandwidth)
- unsigned char Pstages; //filter stages+1
- unsigned char Pfreqtrack;//how the filter frequency is changing according the note frequency
- unsigned char Pgain;//filter's output gain
-
- //Formant filter parameters
- unsigned char Pnumformants;//how many formants are used
- unsigned char Pformantslowness;//how slow varies the formants
- unsigned char Pvowelclearness;//how vowels are kept clean (how much try to avoid "mixed" vowels)
- unsigned char Pcenterfreq,Poctavesfreq;//the center frequency of the res. func., and the number of octaves
-
- struct {
- struct {
- unsigned char freq,amp,q;//frequency,amplitude,Q
- }formants[FF_MAX_FORMANTS];
- }Pvowels[FF_MAX_VOWELS];
-
-
- unsigned char Psequencesize;//how many vowels are in the sequence
- unsigned char Psequencestretch;//how the sequence is stretched (how the input from filter envelopes/LFOs/etc. is "stretched")
- unsigned char Psequencereversed;//if the input from filter envelopes/LFOs/etc. is reversed(negated)
- struct {
- unsigned char nvowel;//the vowel from the position
- } Psequence[FF_MAX_SEQUENCE];
-
- REALTYPE getcenterfreq();
- REALTYPE getoctavesfreq();
- REALTYPE getfreqpos(REALTYPE freq);
- REALTYPE getfreqx(REALTYPE x);
-
- void formantfilterH(int nvowel,int nfreqs,REALTYPE *freqs);//used by UI
-
- REALTYPE getformantfreq(unsigned char freq);
- REALTYPE getformantamp(unsigned char amp);
- REALTYPE getformantq(unsigned char q);
-
- bool changed;
-
- private:
- void defaults(int n);
-
- //stored default parameters
- unsigned char Dtype;
- unsigned char Dfreq;
- unsigned char Dq;
+class FilterParams:public Presets
+{
+public:
+ FilterParams(unsigned char Ptype_,unsigned char Pfreq,unsigned char Pq_);
+ ~FilterParams();
+
+ void add2XML(XMLwrapper *xml);
+ void add2XMLsection(XMLwrapper *xml,int n);
+ void defaults();
+ void getfromXML(XMLwrapper *xml);
+ void getfromXMLsection(XMLwrapper *xml,int n);
+
+
+ void getfromFilterParams(FilterParams *pars);
+
+ REALTYPE getfreq();
+ REALTYPE getq();
+ REALTYPE getfreqtracking(REALTYPE notefreq);
+ REALTYPE getgain();
+
+ unsigned char Pcategory;//Filter category (Analog/Formant/StVar)
+ unsigned char Ptype;// Filter type (for analog lpf,hpf,bpf..)
+ unsigned char Pfreq;// Frequency (64-central frequency)
+ unsigned char Pq; // Q parameters (resonance or bandwidth)
+ unsigned char Pstages; //filter stages+1
+ unsigned char Pfreqtrack;//how the filter frequency is changing according the note frequency
+ unsigned char Pgain;//filter's output gain
+
+ //Formant filter parameters
+ unsigned char Pnumformants;//how many formants are used
+ unsigned char Pformantslowness;//how slow varies the formants
+ unsigned char Pvowelclearness;//how vowels are kept clean (how much try to avoid "mixed" vowels)
+ unsigned char Pcenterfreq,Poctavesfreq;//the center frequency of the res. func., and the number of octaves
+
+ struct {
+ struct {
+ unsigned char freq,amp,q;//frequency,amplitude,Q
+ }formants[FF_MAX_FORMANTS];
+ }Pvowels[FF_MAX_VOWELS];
+
+
+ unsigned char Psequencesize;//how many vowels are in the sequence
+ unsigned char Psequencestretch;//how the sequence is stretched (how the input from filter envelopes/LFOs/etc. is "stretched")
+ unsigned char Psequencereversed;//if the input from filter envelopes/LFOs/etc. is reversed(negated)
+ struct {
+ unsigned char nvowel;//the vowel from the position
+ } Psequence[FF_MAX_SEQUENCE];
+
+ REALTYPE getcenterfreq();
+ REALTYPE getoctavesfreq();
+ REALTYPE getfreqpos(REALTYPE freq);
+ REALTYPE getfreqx(REALTYPE x);
+
+ void formantfilterH(int nvowel,int nfreqs,REALTYPE *freqs);//used by UI
+
+ REALTYPE getformantfreq(unsigned char freq);
+ REALTYPE getformantamp(unsigned char amp);
+ REALTYPE getformantq(unsigned char q);
+
+ bool changed;
+
+private:
+ void defaults(int n);
+
+ //stored default parameters
+ unsigned char Dtype;
+ unsigned char Dfreq;
+ unsigned char Dq;
};
#endif
diff --git a/src/Params/LFOParams.C b/src/Params/LFOParams.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
LFOParams.C - Parameters for LFO
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,14 +27,18 @@
int LFOParams::time;
-LFOParams::LFOParams(char Pfreq_,char Pintensity_,char Pstartphase_, char PLFOtype_,char Prandomness_, char Pdelay_,char Pcontinous_,char fel_):Presets(){
- switch(fel_) {
- case 0:setpresettype("Plfofrequency");
- break;
- case 1:setpresettype("Plfoamplitude");
- break;
- case 2:setpresettype("Plfofilter");
- break;
+LFOParams::LFOParams(char Pfreq_,char Pintensity_,char Pstartphase_, char PLFOtype_,char Prandomness_, char Pdelay_,char Pcontinous_,char fel_):Presets()
+{
+ switch (fel_) {
+ case 0:
+ setpresettype("Plfofrequency");
+ break;
+ case 1:
+ setpresettype("Plfoamplitude");
+ break;
+ case 2:
+ setpresettype("Plfofilter");
+ break;
};
Dfreq=Pfreq_;
Dintensity=Pintensity_;
@@ -45,14 +49,16 @@ LFOParams::LFOParams(char Pfreq_,char Pintensity_,char Pstartphase_, char PLFOty
Dcontinous=Pcontinous_;
fel=fel_;
time=0;
-
+
defaults();
};
-LFOParams::~LFOParams(){
+LFOParams::~LFOParams()
+{
};
-void LFOParams::defaults(){
+void LFOParams::defaults()
+{
Pfreq=Dfreq/127.0;
Pintensity=Dintensity;
Pstartphase=Dstartphase;
@@ -65,7 +71,8 @@ void LFOParams::defaults(){
};
-void LFOParams::add2XML(XMLwrapper *xml){
+void LFOParams::add2XML(XMLwrapper *xml)
+{
xml->addparreal("freq",Pfreq);
xml->addpar("intensity",Pintensity);
xml->addpar("start_phase",Pstartphase);
@@ -77,7 +84,8 @@ void LFOParams::add2XML(XMLwrapper *xml){
xml->addparbool("continous",Pcontinous);
};
-void LFOParams::getfromXML(XMLwrapper *xml){
+void LFOParams::getfromXML(XMLwrapper *xml)
+{
Pfreq=xml->getparreal("freq",Pfreq,0.0,1.0);
Pintensity=xml->getpar127("intensity",Pintensity);
Pstartphase=xml->getpar127("start_phase",Pstartphase);
diff --git a/src/Params/LFOParams.h b/src/Params/LFOParams.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
LFOParams.h - Parameters for LFO
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,38 +26,39 @@
#include "../Misc/XMLwrapper.h"
#include "Presets.h"
-class LFOParams:public Presets{
- public:
- LFOParams(char Pfreq_,char Pintensity_,char Pstartphase_, char PLFOtype_,char Prandomness_, char Pdelay_,char Pcontinous,char fel_);
- ~LFOParams();
-
- void add2XML(XMLwrapper *xml);
- void defaults();
- /**Loads the LFO from the xml*/
- void getfromXML(XMLwrapper *xml);
-
+class LFOParams:public Presets
+{
+public:
+ LFOParams(char Pfreq_,char Pintensity_,char Pstartphase_, char PLFOtype_,char Prandomness_, char Pdelay_,char Pcontinous,char fel_);
+ ~LFOParams();
+
+ void add2XML(XMLwrapper *xml);
+ void defaults();
+ /**Loads the LFO from the xml*/
+ void getfromXML(XMLwrapper *xml);
+
/* MIDI Parameters*/
- REALTYPE Pfreq; /**<frequency*/
- unsigned char Pintensity; /**<intensity*/
- unsigned char Pstartphase;/**<start phase (0=random)*/
- unsigned char PLFOtype; /**<LFO type (sin,triangle,square,ramp,...)*/
- unsigned char Prandomness;/**<randomness (0=off)*/
- unsigned char Pfreqrand; /**<frequency randomness (0=off)*/
- unsigned char Pdelay; /**<delay (0=off)*/
- unsigned char Pcontinous; /**<1 if LFO is continous*/
- unsigned char Pstretch; /**<how the LFO is "stretched" according the note frequency (64=no stretch)*/
+ REALTYPE Pfreq; /**<frequency*/
+ unsigned char Pintensity; /**<intensity*/
+ unsigned char Pstartphase;/**<start phase (0=random)*/
+ unsigned char PLFOtype; /**<LFO type (sin,triangle,square,ramp,...)*/
+ unsigned char Prandomness;/**<randomness (0=off)*/
+ unsigned char Pfreqrand; /**<frequency randomness (0=off)*/
+ unsigned char Pdelay; /**<delay (0=off)*/
+ unsigned char Pcontinous; /**<1 if LFO is continous*/
+ unsigned char Pstretch; /**<how the LFO is "stretched" according the note frequency (64=no stretch)*/
- int fel;//what kind is the LFO (0 - frequency, 1 - amplitude, 2 - filter)
- static int time;//is used by Pcontinous parameter
- private:
- /* Default parameters */
- unsigned char Dfreq;
- unsigned char Dintensity;
- unsigned char Dstartphase;
- unsigned char DLFOtype;
- unsigned char Drandomness;
- unsigned char Ddelay;
- unsigned char Dcontinous;
+ int fel;//what kind is the LFO (0 - frequency, 1 - amplitude, 2 - filter)
+ static int time;//is used by Pcontinous parameter
+private:
+ /* Default parameters */
+ unsigned char Dfreq;
+ unsigned char Dintensity;
+ unsigned char Dstartphase;
+ unsigned char DLFOtype;
+ unsigned char Drandomness;
+ unsigned char Ddelay;
+ unsigned char Dcontinous;
};
diff --git a/src/Params/PADnoteParameters.C b/src/Params/PADnoteParameters.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
- PADnoteParameters.C - Parameters for PADnote (PADsynth)
+
+ PADnoteParameters.C - Parameters for PADnote (PADsynth)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,12 +24,13 @@
#include "../Output/WAVaudiooutput.h"
using namespace std;
-PADnoteParameters::PADnoteParameters(FFTwrapper *fft_,pthread_mutex_t *mutex_):Presets(){
+PADnoteParameters::PADnoteParameters(FFTwrapper *fft_,pthread_mutex_t *mutex_):Presets()
+{
setpresettype("Ppadsyth");
fft=fft_;
mutex=mutex_;
-
+
resonance=new Resonance();
oscilgen=new OscilGen(fft_,resonance);
oscilgen->ADvsPAD=true;
@@ -37,7 +38,7 @@ PADnoteParameters::PADnoteParameters(FFTwrapper *fft_,pthread_mutex_t *mutex_):P
FreqEnvelope=new EnvelopeParams(0,0);
FreqEnvelope->ASRinit(64,50,64,60);
FreqLfo=new LFOParams(70,0,64,0,0,0,0,0);
-
+
AmpEnvelope=new EnvelopeParams(64,1);
AmpEnvelope->ADSRinit_dB(0,40,127,25);
AmpLfo=new LFOParams(80,0,64,0,0,0,0,1);
@@ -46,14 +47,15 @@ PADnoteParameters::PADnoteParameters(FFTwrapper *fft_,pthread_mutex_t *mutex_):P
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);
-
+
for (int i=0;i<PAD_MAX_SAMPLES;i++) sample[i].smp=NULL;
newsample.smp=NULL;
-
+
defaults();
};
-PADnoteParameters::~PADnoteParameters(){
+PADnoteParameters::~PADnoteParameters()
+{
deletesamples();
delete(oscilgen);
delete(resonance);
@@ -65,10 +67,11 @@ PADnoteParameters::~PADnoteParameters(){
delete(GlobalFilter);
delete(FilterEnvelope);
delete(FilterLfo);
-
+
};
-void PADnoteParameters::defaults(){
+void PADnoteParameters::defaults()
+{
Pmode=0;
Php.base.type=0;
Php.base.par1=80;
@@ -85,7 +88,7 @@ void PADnoteParameters::defaults(){
setPbandwidth(500);
Pbwscale=0;
-
+
resonance->defaults();
oscilgen->defaults();
@@ -93,7 +96,7 @@ void PADnoteParameters::defaults(){
Phrpos.par1=64;
Phrpos.par2=64;
Phrpos.par3=0;
-
+
Pquality.samplesize=3;
Pquality.basenote=4;
Pquality.oct=3;
@@ -108,7 +111,7 @@ void PADnoteParameters::defaults(){
PDetuneType=1;
FreqEnvelope->defaults();
FreqLfo->defaults();
-
+
/* Amplitude Global Parameters */
PVolume=90;
PPanning=64;//center
@@ -119,7 +122,7 @@ void PADnoteParameters::defaults(){
PPunchTime=60;
PPunchStretch=64;
PPunchVelocitySensing=72;
-
+
/* Filter Global Parameters*/
PFilterVelocityScale=64;
PFilterVelocityScaleFunction=64;
@@ -130,24 +133,27 @@ void PADnoteParameters::defaults(){
deletesamples();
};
-void PADnoteParameters::deletesample(int n){
+void PADnoteParameters::deletesample(int n)
+{
if ((n<0)||(n>=PAD_MAX_SAMPLES)) return;
- if (sample[n].smp!=NULL){
- delete[]sample[n].smp;
- sample[n].smp=NULL;
+ if (sample[n].smp!=NULL) {
+ delete[]sample[n].smp;
+ sample[n].smp=NULL;
};
sample[n].size=0;
sample[n].basefreq=440.0;
};
-void PADnoteParameters::deletesamples(){
+void PADnoteParameters::deletesamples()
+{
for (int i=0;i<PAD_MAX_SAMPLES;i++) deletesample(i);
};
/*
* Get the harmonic profile (i.e. the frequency distributio of a single harmonic)
*/
-REALTYPE PADnoteParameters::getprofile(REALTYPE *smp,int size){
+REALTYPE PADnoteParameters::getprofile(REALTYPE *smp,int size)
+{
for (int i=0;i<size;i++) smp[i]=0.0;
const int supersample=16;
REALTYPE basepar=pow(2.0,(1.0-Php.base.par1/127.0)*12.0);
@@ -159,104 +165,118 @@ REALTYPE PADnoteParameters::getprofile(REALTYPE *smp,int size){
REALTYPE amppar2=(1.0-Php.amp.par2/127.0)*0.998+0.001;
REALTYPE width=pow(150.0/(Php.width+22.0),2.0);
- for (int i=0;i<size*supersample;i++){
- bool makezero=false;
- REALTYPE x=i*1.0/(size*(REALTYPE) supersample);
-
- REALTYPE origx=x;
-
- //do the sizing (width)
- x=(x-0.5)*width+0.5;
- if (x<0.0) {
- x=0.0;
- makezero=true;
- } else {
- if (x>1.0) {
- x=1.0;
- makezero=true;
- };
- };
-
- //compute the full profile or one half
- switch(Php.onehalf){
- case 1:x=x*0.5+0.5;
- break;
- case 2:x=x*0.5;
- break;
- };
-
- REALTYPE x_before_freq_mult=x;
-
- //do the frequency multiplier
- x*=freqmult;
-
- //do the modulation of the profile
- x+=sin(x_before_freq_mult*3.1415926*modfreq)*modpar1;
- x=fmod(x+1000.0,1.0)*2.0-1.0;
-
-
- //this is the base function of the profile
- REALTYPE f;
- switch (Php.base.type){
- case 1:f=exp(-(x*x)*basepar);if (f<0.4) f=0.0; else f=1.0;
- break;
- case 2:f=exp(-(fabs(x))*sqrt(basepar));
- break;
- default:f=exp(-(x*x)*basepar);
- break;
- };
- if (makezero) f=0.0;
-
- REALTYPE amp=1.0;
- origx=origx*2.0-1.0;
-
- //compute the amplitude multiplier
- switch(Php.amp.type){
- case 1:amp=exp(-(origx*origx)*10.0*amppar1);
- break;
- case 2:amp=0.5*(1.0+cos(3.1415926*origx*sqrt(amppar1*4.0+1.0)));
- break;
- case 3:amp=1.0/(pow(origx*(amppar1*2.0+0.8),14.0)+1.0);
- break;
- };
-
- //apply the amplitude multiplier
- REALTYPE finalsmp=f;
- if (Php.amp.type!=0){
- switch(Php.amp.mode){
- case 0:finalsmp=amp*(1.0-amppar2)+finalsmp*amppar2;
- break;
- case 1:finalsmp*=amp*(1.0-amppar2)+amppar2;
- break;
- case 2:finalsmp=finalsmp/(amp+pow(amppar2,4.0)*20.0+0.0001);
- break;
- case 3:finalsmp=amp/(finalsmp+pow(amppar2,4.0)*20.0+0.0001);
- break;
- };
- };
-
- smp[i/supersample]+=finalsmp/supersample;
- };
+ for (int i=0;i<size*supersample;i++) {
+ bool makezero=false;
+ REALTYPE x=i*1.0/(size*(REALTYPE) supersample);
+
+ REALTYPE origx=x;
+
+ //do the sizing (width)
+ x=(x-0.5)*width+0.5;
+ if (x<0.0) {
+ x=0.0;
+ makezero=true;
+ } else {
+ if (x>1.0) {
+ x=1.0;
+ makezero=true;
+ };
+ };
+
+ //compute the full profile or one half
+ switch (Php.onehalf) {
+ case 1:
+ x=x*0.5+0.5;
+ break;
+ case 2:
+ x=x*0.5;
+ break;
+ };
+
+ REALTYPE x_before_freq_mult=x;
+
+ //do the frequency multiplier
+ x*=freqmult;
+
+ //do the modulation of the profile
+ x+=sin(x_before_freq_mult*3.1415926*modfreq)*modpar1;
+ x=fmod(x+1000.0,1.0)*2.0-1.0;
+
+
+ //this is the base function of the profile
+ REALTYPE f;
+ switch (Php.base.type) {
+ case 1:
+ f=exp(-(x*x)*basepar);
+ if (f<0.4) f=0.0;
+ else f=1.0;
+ break;
+ case 2:
+ f=exp(-(fabs(x))*sqrt(basepar));
+ break;
+ default:
+ f=exp(-(x*x)*basepar);
+ break;
+ };
+ if (makezero) f=0.0;
+
+ REALTYPE amp=1.0;
+ origx=origx*2.0-1.0;
+
+ //compute the amplitude multiplier
+ switch (Php.amp.type) {
+ case 1:
+ amp=exp(-(origx*origx)*10.0*amppar1);
+ break;
+ case 2:
+ amp=0.5*(1.0+cos(3.1415926*origx*sqrt(amppar1*4.0+1.0)));
+ break;
+ case 3:
+ amp=1.0/(pow(origx*(amppar1*2.0+0.8),14.0)+1.0);
+ break;
+ };
+
+ //apply the amplitude multiplier
+ REALTYPE finalsmp=f;
+ if (Php.amp.type!=0) {
+ switch (Php.amp.mode) {
+ case 0:
+ finalsmp=amp*(1.0-amppar2)+finalsmp*amppar2;
+ break;
+ case 1:
+ finalsmp*=amp*(1.0-amppar2)+amppar2;
+ break;
+ case 2:
+ finalsmp=finalsmp/(amp+pow(amppar2,4.0)*20.0+0.0001);
+ break;
+ case 3:
+ finalsmp=amp/(finalsmp+pow(amppar2,4.0)*20.0+0.0001);
+ break;
+ };
+ };
+
+ smp[i/supersample]+=finalsmp/supersample;
+ };
//normalize the profile (make the max. to be equal to 1.0)
REALTYPE max=0.0;
for (int i=0;i<size;i++) {
- if (smp[i]<0.0) smp[i]=0.0;
- if (smp[i]>max) max=smp[i];
+ if (smp[i]<0.0) smp[i]=0.0;
+ if (smp[i]>max) max=smp[i];
};
if (max<0.00001) max=1.0;
for (int i=0;i<size;i++) smp[i]/=max;
-
+
if (!Php.autoscale) return(0.5);
//compute the estimated perceived bandwidth
REALTYPE sum=0.0;
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>=4.0) break;
+ sum+=smp[i]*smp[i]+smp[size-i-1]*smp[size-i-1];
+ if (sum>=4.0) break;
};
-
+
REALTYPE result=1.0-2.0*i/(REALTYPE) size;
return(result);
};
@@ -265,7 +285,8 @@ REALTYPE PADnoteParameters::getprofile(REALTYPE *smp,int size){
* Compute the real bandwidth in cents and returns it
* Also, sets the bandwidth parameter
*/
-REALTYPE PADnoteParameters::setPbandwidth(int Pbandwidth){
+REALTYPE PADnoteParameters::setPbandwidth(int Pbandwidth)
+{
this->Pbandwidth=Pbandwidth;
REALTYPE result=pow(Pbandwidth/1000.0,1.1);
result=pow(10.0,result*4.0)*0.25;
@@ -275,46 +296,47 @@ REALTYPE PADnoteParameters::setPbandwidth(int Pbandwidth){
/*
* Get the harmonic(overtone) position
*/
-REALTYPE PADnoteParameters::getNhr(int n){
+REALTYPE PADnoteParameters::getNhr(int n)
+{
REALTYPE result=1.0;
REALTYPE par1=pow(10.0,-(1.0-Phrpos.par1/255.0)*3.0);
REALTYPE par2=Phrpos.par2/255.0;
-
+
REALTYPE n0=n-1.0;
REALTYPE tmp=0.0;
int thresh=0;
- switch(Phrpos.type){
- case 1:
- thresh=(int)(par2*par2*100.0)+1;
- if (n<thresh) result=n;
- else result=1.0+n0+(n0-thresh+1.0)*par1*8.0;
- break;
- case 2:
- thresh=(int)(par2*par2*100.0)+1;
- if (n<thresh) result=n;
- else result=1.0+n0-(n0-thresh+1.0)*par1*0.90;
- break;
- case 3:
- tmp=par1*100.0+1.0;
- result=pow(n0/tmp,1.0-par2*0.8)*tmp+1.0;
- break;
- case 4:
- result=n0*(1.0-par1)+pow(n0*0.1,par2*3.0+1.0)*par1*10.0+1.0;
- break;
- case 5:
- result=n0+sin(n0*par2*par2*PI*0.999)*sqrt(par1)*2.0+1.0;
- break;
- case 6:
- tmp=pow(par2*2.0,2.0)+0.1;
- result=n0*pow(1.0+par1*pow(n0*0.8,tmp),tmp)+1.0;
- break;
- default:
- result=n;
- break;
+ switch (Phrpos.type) {
+ case 1:
+ thresh=(int)(par2*par2*100.0)+1;
+ if (n<thresh) result=n;
+ else result=1.0+n0+(n0-thresh+1.0)*par1*8.0;
+ break;
+ case 2:
+ thresh=(int)(par2*par2*100.0)+1;
+ if (n<thresh) result=n;
+ else result=1.0+n0-(n0-thresh+1.0)*par1*0.90;
+ break;
+ case 3:
+ tmp=par1*100.0+1.0;
+ result=pow(n0/tmp,1.0-par2*0.8)*tmp+1.0;
+ break;
+ case 4:
+ result=n0*(1.0-par1)+pow(n0*0.1,par2*3.0+1.0)*par1*10.0+1.0;
+ break;
+ case 5:
+ result=n0+sin(n0*par2*par2*PI*0.999)*sqrt(par1)*2.0+1.0;
+ break;
+ case 6:
+ tmp=pow(par2*2.0,2.0)+0.1;
+ result=n0*pow(1.0+par1*pow(n0*0.8,tmp),tmp)+1.0;
+ break;
+ default:
+ result=n;
+ break;
};
REALTYPE par3=Phrpos.par3/255.0;
-
+
REALTYPE iresult=floor(result+0.5);
REALTYPE dresult=result-iresult;
@@ -326,9 +348,10 @@ REALTYPE PADnoteParameters::getNhr(int n){
/*
* Generates the long spectrum for Bandwidth mode (only amplitudes are generated; phases will be random)
*/
-void PADnoteParameters::generatespectrum_bandwidthMode(REALTYPE *spectrum, int size,REALTYPE basefreq,REALTYPE *profile,int profilesize,REALTYPE bwadjust){
+void PADnoteParameters::generatespectrum_bandwidthMode(REALTYPE *spectrum, int size,REALTYPE basefreq,REALTYPE *profile,int profilesize,REALTYPE bwadjust)
+{
for (int i=0;i<size;i++) spectrum[i]=0.0;
-
+
REALTYPE harmonics[OSCIL_SIZE/2];
for (int i=0;i<OSCIL_SIZE/2;i++) harmonics[i]=0.0;
//get the harmonic structure from the oscillator (I am using the frequency amplitudes, only)
@@ -339,66 +362,83 @@ void PADnoteParameters::generatespectrum_bandwidthMode(REALTYPE *spectrum, int s
for (int i=0;i<OSCIL_SIZE/2;i++) if (harmonics[i]>max) max=harmonics[i];
if (max<0.000001) max=1;
for (int i=0;i<OSCIL_SIZE/2;i++) harmonics[i]/=max;
-
- for (int nh=1;nh<OSCIL_SIZE/2;nh++){//for each harmonic
- REALTYPE realfreq=getNhr(nh)*basefreq;
- if (realfreq>SAMPLE_RATE*0.49999) break;
- if (realfreq<20.0) break;
- if (harmonics[nh-1]<1e-4) continue;
-
- //compute the bandwidth of each harmonic
- REALTYPE bandwidthcents=setPbandwidth(Pbandwidth);
- REALTYPE bw=(pow(2.0,bandwidthcents/1200.0)-1.0)*basefreq/bwadjust;
- REALTYPE power=1.0;
- switch (Pbwscale){
- case 0: power=1.0;break;
- case 1: power=0.0;break;
- case 2: power=0.25;break;
- case 3: power=0.5;break;
- case 4: power=0.75;break;
- case 5: power=1.5;break;
- case 6: power=2.0;break;
- case 7: power=-0.5;break;
- };
- bw=bw*pow(realfreq/basefreq,power);
+
+ for (int nh=1;nh<OSCIL_SIZE/2;nh++) {//for each harmonic
+ REALTYPE realfreq=getNhr(nh)*basefreq;
+ if (realfreq>SAMPLE_RATE*0.49999) break;
+ if (realfreq<20.0) break;
+ if (harmonics[nh-1]<1e-4) continue;
+
+ //compute the bandwidth of each harmonic
+ REALTYPE bandwidthcents=setPbandwidth(Pbandwidth);
+ REALTYPE bw=(pow(2.0,bandwidthcents/1200.0)-1.0)*basefreq/bwadjust;
+ REALTYPE power=1.0;
+ switch (Pbwscale) {
+ case 0:
+ power=1.0;
+ break;
+ case 1:
+ power=0.0;
+ break;
+ case 2:
+ power=0.25;
+ break;
+ case 3:
+ power=0.5;
+ break;
+ case 4:
+ power=0.75;
+ break;
+ case 5:
+ power=1.5;
+ break;
+ case 6:
+ power=2.0;
+ break;
+ case 7:
+ power=-0.5;
+ break;
+ };
+ bw=bw*pow(realfreq/basefreq,power);
int ibw=(int)((bw/(SAMPLE_RATE*0.5)*size))+1;
- REALTYPE amp=harmonics[nh-1];
- if (resonance->Penabled) amp*=resonance->getfreqresponse(realfreq);
-
- if (ibw>profilesize){//if the bandwidth is larger than the profilesize
- REALTYPE rap=sqrt((REALTYPE)profilesize/(REALTYPE)ibw);
- int cfreq=(int) (realfreq/(SAMPLE_RATE*0.5)*size)-ibw/2;
- for (int i=0;i<ibw;i++){
- int src=(int)(i*rap*rap);
- int spfreq=i+cfreq;
- if (spfreq<0) continue;
- if (spfreq>=size) break;
- spectrum[spfreq]+=amp*profile[src]*rap;
- };
- }else{//if the bandwidth is smaller than the profilesize
- REALTYPE rap=sqrt((REALTYPE)ibw/(REALTYPE)profilesize);
- REALTYPE ibasefreq=realfreq/(SAMPLE_RATE*0.5)*size;
- for (int i=0;i<profilesize;i++){
- REALTYPE idfreq=i/(REALTYPE)profilesize-0.5;
- idfreq*=ibw;
- int spfreq=(int) (idfreq+ibasefreq);
- REALTYPE fspfreq=fmod(idfreq+ibasefreq,1.0);
- if (spfreq<=0) continue;
- if (spfreq>=size-1) break;
- spectrum[spfreq]+=amp*profile[i]*rap*(1.0-fspfreq);
- spectrum[spfreq+1]+=amp*profile[i]*rap*fspfreq;
- };
- };
+ REALTYPE amp=harmonics[nh-1];
+ if (resonance->Penabled) amp*=resonance->getfreqresponse(realfreq);
+
+ if (ibw>profilesize) {//if the bandwidth is larger than the profilesize
+ REALTYPE rap=sqrt((REALTYPE)profilesize/(REALTYPE)ibw);
+ int cfreq=(int) (realfreq/(SAMPLE_RATE*0.5)*size)-ibw/2;
+ for (int i=0;i<ibw;i++) {
+ int src=(int)(i*rap*rap);
+ int spfreq=i+cfreq;
+ if (spfreq<0) continue;
+ if (spfreq>=size) break;
+ spectrum[spfreq]+=amp*profile[src]*rap;
+ };
+ } else {//if the bandwidth is smaller than the profilesize
+ REALTYPE rap=sqrt((REALTYPE)ibw/(REALTYPE)profilesize);
+ REALTYPE ibasefreq=realfreq/(SAMPLE_RATE*0.5)*size;
+ for (int i=0;i<profilesize;i++) {
+ REALTYPE idfreq=i/(REALTYPE)profilesize-0.5;
+ idfreq*=ibw;
+ int spfreq=(int) (idfreq+ibasefreq);
+ REALTYPE fspfreq=fmod(idfreq+ibasefreq,1.0);
+ if (spfreq<=0) continue;
+ if (spfreq>=size-1) break;
+ spectrum[spfreq]+=amp*profile[i]*rap*(1.0-fspfreq);
+ spectrum[spfreq+1]+=amp*profile[i]*rap*fspfreq;
+ };
+ };
};
};
/*
* Generates the long spectrum for non-Bandwidth modes (only amplitudes are generated; phases will be random)
*/
-void PADnoteParameters::generatespectrum_otherModes(REALTYPE *spectrum, int size,REALTYPE basefreq,REALTYPE *profile,int profilesize,REALTYPE bwadjust){
+void PADnoteParameters::generatespectrum_otherModes(REALTYPE *spectrum, int size,REALTYPE basefreq,REALTYPE *profile,int profilesize,REALTYPE bwadjust)
+{
for (int i=0;i<size;i++) spectrum[i]=0.0;
-
+
REALTYPE harmonics[OSCIL_SIZE/2];
for (int i=0;i<OSCIL_SIZE/2;i++) harmonics[i]=0.0;
//get the harmonic structure from the oscillator (I am using the frequency amplitudes, only)
@@ -410,46 +450,47 @@ void PADnoteParameters::generatespectrum_otherModes(REALTYPE *spectrum, int size
if (max<0.000001) max=1;
for (int i=0;i<OSCIL_SIZE/2;i++) harmonics[i]/=max;
- for (int nh=1;nh<OSCIL_SIZE/2;nh++){//for each harmonic
- REALTYPE realfreq=getNhr(nh)*basefreq;
-
- ///sa fac aici interpolarea si sa am grija daca frecv descresc
-
- if (realfreq>SAMPLE_RATE*0.49999) break;
- if (realfreq<20.0) break;
+ for (int nh=1;nh<OSCIL_SIZE/2;nh++) {//for each harmonic
+ REALTYPE realfreq=getNhr(nh)*basefreq;
+
+ ///sa fac aici interpolarea si sa am grija daca frecv descresc
+
+ if (realfreq>SAMPLE_RATE*0.49999) break;
+ if (realfreq<20.0) break;
// if (harmonics[nh-1]<1e-4) continue;
- REALTYPE amp=harmonics[nh-1];
- if (resonance->Penabled) amp*=resonance->getfreqresponse(realfreq);
- int cfreq=(int) (realfreq/(SAMPLE_RATE*0.5)*size);
+ REALTYPE amp=harmonics[nh-1];
+ if (resonance->Penabled) amp*=resonance->getfreqresponse(realfreq);
+ int cfreq=(int) (realfreq/(SAMPLE_RATE*0.5)*size);
- spectrum[cfreq]=amp+1e-9;
+ spectrum[cfreq]=amp+1e-9;
};
-
- if (Pmode!=1){
+
+ if (Pmode!=1) {
int old=0;
- for (int k=1;k<size;k++){
- if ( (spectrum[k]>1e-10) || (k==(size-1)) ){
- int delta=k-old;
- REALTYPE val1=spectrum[old];
- REALTYPE val2=spectrum[k];
- REALTYPE idelta=1.0/delta;
- for (int i=0;i<delta;i++){
- REALTYPE x=idelta*i;
- spectrum[old+i]=val1*(1.0-x)+val2*x;
- };
- old=k;
- };
- };
+ for (int k=1;k<size;k++) {
+ if ( (spectrum[k]>1e-10) || (k==(size-1)) ) {
+ int delta=k-old;
+ REALTYPE val1=spectrum[old];
+ REALTYPE val2=spectrum[k];
+ REALTYPE idelta=1.0/delta;
+ for (int i=0;i<delta;i++) {
+ REALTYPE x=idelta*i;
+ spectrum[old+i]=val1*(1.0-x)+val2*x;
+ };
+ old=k;
+ };
+ };
};
-
+
};
/*
* Applies the parameters (i.e. computes all the samples, based on parameters);
*/
-void PADnoteParameters::applyparameters(bool lockmutex){
+void PADnoteParameters::applyparameters(bool lockmutex)
+{
const int samplesize=(((int) 1)<<(Pquality.samplesize+14));
int spectrumsize=samplesize/2;
REALTYPE spectrum[spectrumsize];
@@ -467,97 +508,99 @@ void PADnoteParameters::applyparameters(bool lockmutex){
if (Pquality.smpoct==5) smpoct=6;
if (Pquality.smpoct==6) smpoct=12;
if (smpoct!=0) samplemax*=smpoct;
- else samplemax=samplemax/2+1;
+ else samplemax=samplemax/2+1;
if (samplemax==0) samplemax=1;
-
+
//prepare a BIG FFT stuff
FFTwrapper *fft=new FFTwrapper(samplesize);
FFTFREQS fftfreqs;
newFFTFREQS(&fftfreqs,samplesize/2);
-
+
REALTYPE adj[samplemax];//this is used to compute frequency relation to the base frequency
for (int nsample=0;nsample<samplemax;nsample++) adj[nsample]=(Pquality.oct+1.0)*(REALTYPE)nsample/samplemax;
- for (int nsample=0;nsample<samplemax;nsample++){
- REALTYPE tmp=adj[nsample]-adj[samplemax-1]*0.5;
- REALTYPE basefreqadjust=pow(2.0,tmp);
+ for (int nsample=0;nsample<samplemax;nsample++) {
+ REALTYPE tmp=adj[nsample]-adj[samplemax-1]*0.5;
+ REALTYPE basefreqadjust=pow(2.0,tmp);
if (Pmode==0) generatespectrum_bandwidthMode(spectrum,spectrumsize,basefreq*basefreqadjust,profile,profilesize,bwadjust);
- else generatespectrum_otherModes(spectrum,spectrumsize,basefreq*basefreqadjust,profile,profilesize,bwadjust);
+ else generatespectrum_otherModes(spectrum,spectrumsize,basefreq*basefreqadjust,profile,profilesize,bwadjust);
- const int extra_samples=5;//the last samples contains the first samples (used for linear/cubic interpolation)
+ const int extra_samples=5;//the last samples contains the first samples (used for linear/cubic interpolation)
newsample.smp=new REALTYPE[samplesize+extra_samples];
-
- newsample.smp[0]=0.0;
- for (int i=1;i<spectrumsize;i++){//randomize the phases
- REALTYPE phase=RND*6.29;
- fftfreqs.c[i]=spectrum[i]*cos(phase);
- fftfreqs.s[i]=spectrum[i]*sin(phase);
- };
- fft->freqs2smps(fftfreqs,newsample.smp);//that's all; here is the only ifft for the whole sample; no windows are used ;-)
-
+
+ newsample.smp[0]=0.0;
+ for (int i=1;i<spectrumsize;i++) {//randomize the phases
+ REALTYPE phase=RND*6.29;
+ fftfreqs.c[i]=spectrum[i]*cos(phase);
+ fftfreqs.s[i]=spectrum[i]*sin(phase);
+ };
+ fft->freqs2smps(fftfreqs,newsample.smp);//that's all; here is the only ifft for the whole sample; no windows are used ;-)
+
//normalize(rms)
- REALTYPE rms=0.0;
+ REALTYPE rms=0.0;
for (int i=0;i<samplesize;i++) rms+=newsample.smp[i]*newsample.smp[i];
- rms=sqrt(rms);
+ rms=sqrt(rms);
if (rms<0.000001) rms=1.0;
- rms*=sqrt(262144.0/samplesize);
+ rms*=sqrt(262144.0/samplesize);
for (int i=0;i<samplesize;i++) newsample.smp[i]*=1.0/rms*50.0;
-
- //prepare extra samples used by the linear or cubic interpolation
+
+ //prepare extra samples used by the linear or cubic interpolation
for (int i=0;i<extra_samples;i++) newsample.smp[i+samplesize]=newsample.smp[i];
- //replace the current sample with the new computed sample
- if (lockmutex){
- pthread_mutex_lock(mutex);
- deletesample(nsample);
- sample[nsample].smp=newsample.smp;
- sample[nsample].size=samplesize;
- sample[nsample].basefreq=basefreq*basefreqadjust;
- pthread_mutex_unlock(mutex);
- } else {
- deletesample(nsample);
- sample[nsample].smp=newsample.smp;
- sample[nsample].size=samplesize;
- sample[nsample].basefreq=basefreq*basefreqadjust;
- };
- newsample.smp=NULL;
+ //replace the current sample with the new computed sample
+ if (lockmutex) {
+ pthread_mutex_lock(mutex);
+ deletesample(nsample);
+ sample[nsample].smp=newsample.smp;
+ sample[nsample].size=samplesize;
+ sample[nsample].basefreq=basefreq*basefreqadjust;
+ pthread_mutex_unlock(mutex);
+ } else {
+ deletesample(nsample);
+ sample[nsample].smp=newsample.smp;
+ sample[nsample].size=samplesize;
+ sample[nsample].basefreq=basefreq*basefreqadjust;
+ };
+ newsample.smp=NULL;
};
delete(fft);
deleteFFTFREQS(&fftfreqs);
-
+
//delete the additional samples that might exists and are not useful
- if (lockmutex){
+ if (lockmutex) {
pthread_mutex_lock(mutex);
- for (int i=samplemax;i<PAD_MAX_SAMPLES;i++) deletesample(i);
+ for (int i=samplemax;i<PAD_MAX_SAMPLES;i++) deletesample(i);
pthread_mutex_unlock(mutex);
} else {
- for (int i=samplemax;i<PAD_MAX_SAMPLES;i++) deletesample(i);
+ for (int i=samplemax;i<PAD_MAX_SAMPLES;i++) deletesample(i);
};
};
-void PADnoteParameters::export2wav(string basefilename){
- applyparameters(true);
- basefilename+="_PADsynth_";
- for (int k=0;k<PAD_MAX_SAMPLES;k++){
- if (sample[k].smp==NULL) continue;
- char tmpstr[20];
- snprintf(tmpstr,20,"_%02d",k+1);
- string filename=basefilename+string(tmpstr)+".wav";
- WAVaudiooutput wav;
- if (wav.newfile(filename,SAMPLE_RATE,1)) {
- int nsmps=sample[k].size;
- short int *smps=new short int[nsmps];
- for (int i=0;i<nsmps;i++) smps[i]=(short int)(sample[k].smp[i]*32767.0);
- wav.write_mono_samples(nsmps, smps);
- wav.close();
- };
- };
+void PADnoteParameters::export2wav(string basefilename)
+{
+ applyparameters(true);
+ basefilename+="_PADsynth_";
+ for (int k=0;k<PAD_MAX_SAMPLES;k++) {
+ if (sample[k].smp==NULL) continue;
+ char tmpstr[20];
+ snprintf(tmpstr,20,"_%02d",k+1);
+ string filename=basefilename+string(tmpstr)+".wav";
+ WAVaudiooutput wav;
+ if (wav.newfile(filename,SAMPLE_RATE,1)) {
+ int nsmps=sample[k].size;
+ short int *smps=new short int[nsmps];
+ for (int i=0;i<nsmps;i++) smps[i]=(short int)(sample[k].smp[i]*32767.0);
+ wav.write_mono_samples(nsmps, smps);
+ wav.close();
+ };
+ };
};
-void PADnoteParameters::add2XML(XMLwrapper *xml){
+void PADnoteParameters::add2XML(XMLwrapper *xml)
+{
xml->information.PADsynth_used=true;
xml->addparbool("stereo",PStereo);
@@ -566,196 +609,197 @@ void PADnoteParameters::add2XML(XMLwrapper *xml){
xml->addpar("bandwidth_scale",Pbwscale);
xml->beginbranch("HARMONIC_PROFILE");
- xml->addpar("base_type",Php.base.type);
- xml->addpar("base_par1",Php.base.par1);
- xml->addpar("frequency_multiplier",Php.freqmult);
- xml->addpar("modulator_par1",Php.modulator.par1);
- xml->addpar("modulator_frequency",Php.modulator.freq);
- xml->addpar("width",Php.width);
- xml->addpar("amplitude_multiplier_type",Php.amp.type);
- xml->addpar("amplitude_multiplier_mode",Php.amp.mode);
- xml->addpar("amplitude_multiplier_par1",Php.amp.par1);
- xml->addpar("amplitude_multiplier_par2",Php.amp.par2);
- xml->addparbool("autoscale",Php.autoscale);
- xml->addpar("one_half",Php.onehalf);
+ xml->addpar("base_type",Php.base.type);
+ xml->addpar("base_par1",Php.base.par1);
+ xml->addpar("frequency_multiplier",Php.freqmult);
+ xml->addpar("modulator_par1",Php.modulator.par1);
+ xml->addpar("modulator_frequency",Php.modulator.freq);
+ xml->addpar("width",Php.width);
+ xml->addpar("amplitude_multiplier_type",Php.amp.type);
+ xml->addpar("amplitude_multiplier_mode",Php.amp.mode);
+ xml->addpar("amplitude_multiplier_par1",Php.amp.par1);
+ xml->addpar("amplitude_multiplier_par2",Php.amp.par2);
+ xml->addparbool("autoscale",Php.autoscale);
+ xml->addpar("one_half",Php.onehalf);
xml->endbranch();
xml->beginbranch("OSCIL");
- oscilgen->add2XML(xml);
+ oscilgen->add2XML(xml);
xml->endbranch();
xml->beginbranch("RESONANCE");
- resonance->add2XML(xml);
+ resonance->add2XML(xml);
xml->endbranch();
xml->beginbranch("HARMONIC_POSITION");
- xml->addpar("type",Phrpos.type);
- xml->addpar("parameter1",Phrpos.par1);
- xml->addpar("parameter2",Phrpos.par2);
- xml->addpar("parameter3",Phrpos.par3);
+ xml->addpar("type",Phrpos.type);
+ xml->addpar("parameter1",Phrpos.par1);
+ xml->addpar("parameter2",Phrpos.par2);
+ xml->addpar("parameter3",Phrpos.par3);
xml->endbranch();
xml->beginbranch("SAMPLE_QUALITY");
- xml->addpar("samplesize",Pquality.samplesize);
- xml->addpar("basenote",Pquality.basenote);
- xml->addpar("octaves",Pquality.oct);
- xml->addpar("samples_per_octave",Pquality.smpoct);
+ xml->addpar("samplesize",Pquality.samplesize);
+ xml->addpar("basenote",Pquality.basenote);
+ xml->addpar("octaves",Pquality.oct);
+ xml->addpar("samples_per_octave",Pquality.smpoct);
xml->endbranch();
xml->beginbranch("AMPLITUDE_PARAMETERS");
- xml->addpar("volume",PVolume);
- xml->addpar("panning",PPanning);
- xml->addpar("velocity_sensing",PAmpVelocityScaleFunction);
- xml->addpar("punch_strength",PPunchStrength);
- xml->addpar("punch_time",PPunchTime);
- xml->addpar("punch_stretch",PPunchStretch);
- xml->addpar("punch_velocity_sensing",PPunchVelocitySensing);
-
- xml->beginbranch("AMPLITUDE_ENVELOPE");
- AmpEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("AMPLITUDE_LFO");
- AmpLfo->add2XML(xml);
- xml->endbranch();
+ xml->addpar("volume",PVolume);
+ xml->addpar("panning",PPanning);
+ xml->addpar("velocity_sensing",PAmpVelocityScaleFunction);
+ xml->addpar("punch_strength",PPunchStrength);
+ xml->addpar("punch_time",PPunchTime);
+ xml->addpar("punch_stretch",PPunchStretch);
+ xml->addpar("punch_velocity_sensing",PPunchVelocitySensing);
+
+ xml->beginbranch("AMPLITUDE_ENVELOPE");
+ AmpEnvelope->add2XML(xml);
+ xml->endbranch();
+ xml->beginbranch("AMPLITUDE_LFO");
+ AmpLfo->add2XML(xml);
xml->endbranch();
-
+
+ xml->endbranch();
+
xml->beginbranch("FREQUENCY_PARAMETERS");
- xml->addpar("fixed_freq",Pfixedfreq);
- xml->addpar("fixed_freq_et",PfixedfreqET);
- xml->addpar("detune",PDetune);
- xml->addpar("coarse_detune",PCoarseDetune);
- xml->addpar("detune_type",PDetuneType);
-
- xml->beginbranch("FREQUENCY_ENVELOPE");
- FreqEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FREQUENCY_LFO");
- FreqLfo->add2XML(xml);
- xml->endbranch();
+ xml->addpar("fixed_freq",Pfixedfreq);
+ xml->addpar("fixed_freq_et",PfixedfreqET);
+ xml->addpar("detune",PDetune);
+ xml->addpar("coarse_detune",PCoarseDetune);
+ xml->addpar("detune_type",PDetuneType);
+
+ xml->beginbranch("FREQUENCY_ENVELOPE");
+ FreqEnvelope->add2XML(xml);
+ xml->endbranch();
+
+ xml->beginbranch("FREQUENCY_LFO");
+ FreqLfo->add2XML(xml);
+ xml->endbranch();
xml->endbranch();
-
+
xml->beginbranch("FILTER_PARAMETERS");
- xml->addpar("velocity_sensing_amplitude",PFilterVelocityScale);
- xml->addpar("velocity_sensing",PFilterVelocityScaleFunction);
-
- xml->beginbranch("FILTER");
- GlobalFilter->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FILTER_ENVELOPE");
- FilterEnvelope->add2XML(xml);
- xml->endbranch();
-
- xml->beginbranch("FILTER_LFO");
- FilterLfo->add2XML(xml);
- xml->endbranch();
- xml->endbranch();
+ xml->addpar("velocity_sensing_amplitude",PFilterVelocityScale);
+ xml->addpar("velocity_sensing",PFilterVelocityScaleFunction);
+
+ xml->beginbranch("FILTER");
+ GlobalFilter->add2XML(xml);
+ xml->endbranch();
+
+ xml->beginbranch("FILTER_ENVELOPE");
+ FilterEnvelope->add2XML(xml);
+ xml->endbranch();
+
+ xml->beginbranch("FILTER_LFO");
+ FilterLfo->add2XML(xml);
+ xml->endbranch();
+ xml->endbranch();
};
-void PADnoteParameters::getfromXML(XMLwrapper *xml){
+void PADnoteParameters::getfromXML(XMLwrapper *xml)
+{
PStereo=xml->getparbool("stereo",PStereo);
Pmode=xml->getpar127("mode",0);
Pbandwidth=xml->getpar("bandwidth",Pbandwidth,0,1000);
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.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);
- xml->exitbranch();
+ 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.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);
+ xml->exitbranch();
};
- if (xml->enterbranch("OSCIL")){
+ if (xml->enterbranch("OSCIL")) {
oscilgen->getfromXML(xml);
- xml->exitbranch();
+ xml->exitbranch();
};
- if (xml->enterbranch("RESONANCE")){
+ if (xml->enterbranch("RESONANCE")) {
resonance->getfromXML(xml);
- xml->exitbranch();
+ xml->exitbranch();
};
- if (xml->enterbranch("HARMONIC_POSITION")){
- Phrpos.type=xml->getpar127("type",Phrpos.type);
- Phrpos.par1=xml->getpar("parameter1",Phrpos.par1,0,255);
- Phrpos.par2=xml->getpar("parameter2",Phrpos.par2,0,255);
- Phrpos.par3=xml->getpar("parameter3",Phrpos.par3,0,255);
- xml->exitbranch();
+ if (xml->enterbranch("HARMONIC_POSITION")) {
+ Phrpos.type=xml->getpar127("type",Phrpos.type);
+ Phrpos.par1=xml->getpar("parameter1",Phrpos.par1,0,255);
+ Phrpos.par2=xml->getpar("parameter2",Phrpos.par2,0,255);
+ Phrpos.par3=xml->getpar("parameter3",Phrpos.par3,0,255);
+ xml->exitbranch();
};
- 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);
- xml->exitbranch();
+ 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);
+ xml->exitbranch();
};
- if (xml->enterbranch("AMPLITUDE_PARAMETERS")){
- PVolume=xml->getpar127("volume",PVolume);
- PPanning=xml->getpar127("panning",PPanning);
- PAmpVelocityScaleFunction=xml->getpar127("velocity_sensing",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);
-
- xml->enterbranch("AMPLITUDE_ENVELOPE");
- AmpEnvelope->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("AMPLITUDE_LFO");
- AmpLfo->getfromXML(xml);
- xml->exitbranch();
-
- xml->exitbranch();
+ if (xml->enterbranch("AMPLITUDE_PARAMETERS")) {
+ PVolume=xml->getpar127("volume",PVolume);
+ PPanning=xml->getpar127("panning",PPanning);
+ PAmpVelocityScaleFunction=xml->getpar127("velocity_sensing",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);
+
+ xml->enterbranch("AMPLITUDE_ENVELOPE");
+ AmpEnvelope->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->enterbranch("AMPLITUDE_LFO");
+ AmpLfo->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->exitbranch();
};
-
- if (xml->enterbranch("FREQUENCY_PARAMETERS")){
- Pfixedfreq=xml->getpar127("fixed_freq",Pfixedfreq);
+
+ if (xml->enterbranch("FREQUENCY_PARAMETERS")) {
+ Pfixedfreq=xml->getpar127("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);
-
- xml->enterbranch("FREQUENCY_ENVELOPE");
- FreqEnvelope->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FREQUENCY_LFO");
- FreqLfo->getfromXML(xml);
- xml->exitbranch();
- xml->exitbranch();
+
+ xml->enterbranch("FREQUENCY_ENVELOPE");
+ FreqEnvelope->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->enterbranch("FREQUENCY_LFO");
+ FreqLfo->getfromXML(xml);
+ xml->exitbranch();
+ xml->exitbranch();
};
-
- if (xml->enterbranch("FILTER_PARAMETERS")){
- PFilterVelocityScale=xml->getpar127("velocity_sensing_amplitude",PFilterVelocityScale);
- PFilterVelocityScaleFunction=xml->getpar127("velocity_sensing",PFilterVelocityScaleFunction);
-
- xml->enterbranch("FILTER");
- GlobalFilter->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FILTER_ENVELOPE");
- FilterEnvelope->getfromXML(xml);
- xml->exitbranch();
-
- xml->enterbranch("FILTER_LFO");
- FilterLfo->getfromXML(xml);
- xml->exitbranch();
- xml->exitbranch();
+
+ if (xml->enterbranch("FILTER_PARAMETERS")) {
+ PFilterVelocityScale=xml->getpar127("velocity_sensing_amplitude",PFilterVelocityScale);
+ PFilterVelocityScaleFunction=xml->getpar127("velocity_sensing",PFilterVelocityScaleFunction);
+
+ xml->enterbranch("FILTER");
+ GlobalFilter->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->enterbranch("FILTER_ENVELOPE");
+ FilterEnvelope->getfromXML(xml);
+ xml->exitbranch();
+
+ xml->enterbranch("FILTER_LFO");
+ FilterLfo->getfromXML(xml);
+ xml->exitbranch();
+ xml->exitbranch();
};
};
diff --git a/src/Params/PADnoteParameters.h b/src/Params/PADnoteParameters.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
- PADnoteParameters.h - Parameters for PADnote (PADsynth)
+
+ PADnoteParameters.h - Parameters for PADnote (PADsynth)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -36,132 +36,133 @@
#include "Presets.h"
#include <string>
-class PADnoteParameters:public Presets{
- public:
- PADnoteParameters(FFTwrapper *fft_,pthread_mutex_t *mutex_);
- ~PADnoteParameters();
-
- void defaults();
- void add2XML(XMLwrapper *xml);
- void getfromXML(XMLwrapper *xml);
-
- //returns a value between 0.0-1.0 that represents the estimation perceived bandwidth
- REALTYPE getprofile(REALTYPE *smp,int size);
-
- //parameters
-
- //the mode: 0 - bandwidth, 1 - discrete (bandwidth=0), 2 - continous
- //the harmonic profile is used only on mode 0
- unsigned char Pmode;
-
- //Harmonic profile (the frequency distribution of a single harmonic)
- struct {
- struct{//base function
- unsigned char type;
- unsigned char par1;
- }base;
- unsigned char freqmult;//frequency multiplier of the distribution
- struct{//the modulator of the distribution
- unsigned char par1;
- unsigned char freq;
- }modulator;
-
- unsigned char width;//the width of the resulting function after the modulation
- struct{//the amplitude multiplier of the harmonic profile
- unsigned char mode;
- unsigned char type;
- unsigned char par1;
- unsigned char par2;
- }amp;
- bool autoscale;//if the scale of the harmonic profile is computed automaticaly
- unsigned char onehalf;//what part of the base function is used to make the distribution
- }Php;
-
-
- unsigned int Pbandwidth;//the values are from 0 to 1000
- unsigned char Pbwscale;//how the bandwidth is increased according to the harmonic's frequency
-
- struct{//where are positioned the harmonics (on integer multimplier or different places)
- unsigned char type;
- unsigned char par1,par2,par3;//0..255
- }Phrpos;
-
- struct {//quality of the samples (how many samples, the length of them,etc.)
- unsigned char samplesize;
- unsigned char basenote,oct,smpoct;
- } Pquality;
-
- //frequency parameters
- //If the base frequency is fixed to 440 Hz
- unsigned char Pfixedfreq;
-
- /* 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 short int PDetune;//fine detune
- unsigned short int PCoarseDetune;//coarse detune+octave
- unsigned char PDetuneType;//detune type
-
- EnvelopeParams *FreqEnvelope; //Frequency Envelope
- LFOParams *FreqLfo;//Frequency LFO
-
- //Amplitude parameters
- unsigned char PStereo;
- /* Panning - 0 - random
- 1 - left
- 64 - center
- 127 - right */
- unsigned char PPanning;
-
- unsigned char PVolume;
-
- unsigned char PAmpVelocityScaleFunction;
-
- EnvelopeParams *AmpEnvelope;
-
- LFOParams *AmpLfo;
-
- unsigned char PPunchStrength,PPunchTime,PPunchStretch,PPunchVelocitySensing;
-
- //Filter Parameters
- FilterParams *GlobalFilter;
-
- // filter velocity sensing
- unsigned char PFilterVelocityScale;
-
- // filter velocity sensing
- unsigned char PFilterVelocityScaleFunction;
-
- EnvelopeParams *FilterEnvelope;
- LFOParams *FilterLfo;
-
-
-
-
- REALTYPE setPbandwidth(int Pbandwidth);//returns the BandWidth in cents
- REALTYPE getNhr(int n);//gets the n-th overtone position relatively to N harmonic
-
- void applyparameters(bool lockmutex);
- void export2wav(std::string basefilename);
-
- OscilGen *oscilgen;
- Resonance *resonance;
-
- struct{
- int size;
- REALTYPE basefreq;
- REALTYPE *smp;
- }sample[PAD_MAX_SAMPLES],newsample;
-
- private:
- void generatespectrum_bandwidthMode(REALTYPE *spectrum, int size,REALTYPE basefreq,REALTYPE *profile,int profilesize,REALTYPE bwadjust);
- void generatespectrum_otherModes(REALTYPE *spectrum, int size,REALTYPE basefreq,REALTYPE *profile,int profilesize,REALTYPE bwadjust);
- void deletesamples();
- void deletesample(int n);
-
- FFTwrapper *fft;
- pthread_mutex_t *mutex;
+class PADnoteParameters:public Presets
+{
+public:
+ PADnoteParameters(FFTwrapper *fft_,pthread_mutex_t *mutex_);
+ ~PADnoteParameters();
+
+ void defaults();
+ void add2XML(XMLwrapper *xml);
+ void getfromXML(XMLwrapper *xml);
+
+ //returns a value between 0.0-1.0 that represents the estimation perceived bandwidth
+ REALTYPE getprofile(REALTYPE *smp,int size);
+
+ //parameters
+
+ //the mode: 0 - bandwidth, 1 - discrete (bandwidth=0), 2 - continous
+ //the harmonic profile is used only on mode 0
+ unsigned char Pmode;
+
+ //Harmonic profile (the frequency distribution of a single harmonic)
+ struct {
+ struct {//base function
+ unsigned char type;
+ unsigned char par1;
+ }base;
+ unsigned char freqmult;//frequency multiplier of the distribution
+ struct {//the modulator of the distribution
+ unsigned char par1;
+ unsigned char freq;
+ }modulator;
+
+ unsigned char width;//the width of the resulting function after the modulation
+ struct {//the amplitude multiplier of the harmonic profile
+ unsigned char mode;
+ unsigned char type;
+ unsigned char par1;
+ unsigned char par2;
+ }amp;
+ bool autoscale;//if the scale of the harmonic profile is computed automaticaly
+ unsigned char onehalf;//what part of the base function is used to make the distribution
+ }Php;
+
+
+ unsigned int Pbandwidth;//the values are from 0 to 1000
+ unsigned char Pbwscale;//how the bandwidth is increased according to the harmonic's frequency
+
+ struct {//where are positioned the harmonics (on integer multimplier or different places)
+ unsigned char type;
+ unsigned char par1,par2,par3;//0..255
+ }Phrpos;
+
+ struct {//quality of the samples (how many samples, the length of them,etc.)
+ unsigned char samplesize;
+ unsigned char basenote,oct,smpoct;
+ } Pquality;
+
+ //frequency parameters
+ //If the base frequency is fixed to 440 Hz
+ unsigned char Pfixedfreq;
+
+ /* 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 short int PDetune;//fine detune
+ unsigned short int PCoarseDetune;//coarse detune+octave
+ unsigned char PDetuneType;//detune type
+
+ EnvelopeParams *FreqEnvelope; //Frequency Envelope
+ LFOParams *FreqLfo;//Frequency LFO
+
+ //Amplitude parameters
+ unsigned char PStereo;
+ /* Panning - 0 - random
+ 1 - left
+ 64 - center
+ 127 - right */
+ unsigned char PPanning;
+
+ unsigned char PVolume;
+
+ unsigned char PAmpVelocityScaleFunction;
+
+ EnvelopeParams *AmpEnvelope;
+
+ LFOParams *AmpLfo;
+
+ unsigned char PPunchStrength,PPunchTime,PPunchStretch,PPunchVelocitySensing;
+
+ //Filter Parameters
+ FilterParams *GlobalFilter;
+
+ // filter velocity sensing
+ unsigned char PFilterVelocityScale;
+
+ // filter velocity sensing
+ unsigned char PFilterVelocityScaleFunction;
+
+ EnvelopeParams *FilterEnvelope;
+ LFOParams *FilterLfo;
+
+
+
+
+ REALTYPE setPbandwidth(int Pbandwidth);//returns the BandWidth in cents
+ REALTYPE getNhr(int n);//gets the n-th overtone position relatively to N harmonic
+
+ void applyparameters(bool lockmutex);
+ void export2wav(std::string basefilename);
+
+ OscilGen *oscilgen;
+ Resonance *resonance;
+
+ struct {
+ int size;
+ REALTYPE basefreq;
+ REALTYPE *smp;
+ }sample[PAD_MAX_SAMPLES],newsample;
+
+private:
+ void generatespectrum_bandwidthMode(REALTYPE *spectrum, int size,REALTYPE basefreq,REALTYPE *profile,int profilesize,REALTYPE bwadjust);
+ void generatespectrum_otherModes(REALTYPE *spectrum, int size,REALTYPE basefreq,REALTYPE *profile,int profilesize,REALTYPE bwadjust);
+ void deletesamples();
+ void deletesample(int n);
+
+ FFTwrapper *fft;
+ pthread_mutex_t *mutex;
};
diff --git a/src/Params/Presets.C b/src/Params/Presets.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Presets.C - Presets and Clipboard management
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,89 +24,95 @@
#include <string.h>
-Presets::Presets(){
+Presets::Presets()
+{
type[0]=0;
nelement=-1;
};
-Presets::~Presets(){
+Presets::~Presets()
+{
};
-void Presets::setpresettype(const char *type){
+void Presets::setpresettype(const char *type)
+{
strcpy(this->type,type);
};
-void Presets::copy(const char *name){
+void Presets::copy(const char *name)
+{
XMLwrapper *xml=new XMLwrapper();
-
+
//used only for the clipboard
if (name==NULL) xml->minimal=false;
-
+
char type[MAX_PRESETTYPE_SIZE];
strcpy(type,this->type);
if (nelement!=-1) strcat(type,"n");
if (name==NULL) {
- if (strstr(type,"Plfo")!=NULL) strcpy(type,"Plfo");
+ if (strstr(type,"Plfo")!=NULL) strcpy(type,"Plfo");
};
xml->beginbranch(type);
if (nelement==-1) add2XML(xml);
- else add2XMLsection(xml,nelement);
+ else add2XMLsection(xml,nelement);
xml->endbranch();
if (name==NULL) presetsstore.copyclipboard(xml,type);
- else presetsstore.copypreset(xml,type,name);
-
+ else presetsstore.copypreset(xml,type,name);
+
delete(xml);
nelement=-1;
};
-void Presets::paste(int npreset){
+void Presets::paste(int npreset)
+{
char type[MAX_PRESETTYPE_SIZE];
strcpy(type,this->type);
if (nelement!=-1) strcat(type,"n");
- if (npreset==0){
- if (strstr(type,"Plfo")!=NULL) strcpy(type,"Plfo");
+ if (npreset==0) {
+ if (strstr(type,"Plfo")!=NULL) strcpy(type,"Plfo");
};
XMLwrapper *xml=new XMLwrapper();
- if (npreset==0){
- if (!checkclipboardtype()) {
- nelement=-1;
- delete(xml);
- return;
- };
- if (!presetsstore.pasteclipboard(xml)) {
- delete(xml);
- nelement=-1;
- return;
- };
+ if (npreset==0) {
+ if (!checkclipboardtype()) {
+ nelement=-1;
+ delete(xml);
+ return;
+ };
+ if (!presetsstore.pasteclipboard(xml)) {
+ delete(xml);
+ nelement=-1;
+ return;
+ };
} else {
- if (!presetsstore.pastepreset(xml,npreset)) {
- delete(xml);
- nelement=-1;
- return;
- };
+ if (!presetsstore.pastepreset(xml,npreset)) {
+ delete(xml);
+ nelement=-1;
+ return;
+ };
};
-
+
if (xml->enterbranch(type)==0) {
- nelement=-1;
- return;
+ nelement=-1;
+ return;
+ };
+ if (nelement==-1) {
+ defaults();
+ getfromXML(xml);
+ } else {
+ defaults(nelement);
+ getfromXMLsection(xml,nelement);
};
- if (nelement==-1) {
- defaults();
- getfromXML(xml);
- } else {
- defaults(nelement);
- getfromXMLsection(xml,nelement);
- };
xml->exitbranch();
-
+
delete(xml);
nelement=-1;
};
-bool Presets::checkclipboardtype(){
+bool Presets::checkclipboardtype()
+{
char type[MAX_PRESETTYPE_SIZE];
strcpy(type,this->type);
if (nelement!=-1) strcat(type,"n");
@@ -114,16 +120,19 @@ bool Presets::checkclipboardtype(){
return(presetsstore.checkclipboardtype(type));
};
-void Presets::setelement(int n){
+void Presets::setelement(int n)
+{
nelement=n;
};
-void Presets::rescanforpresets(){
+void Presets::rescanforpresets()
+{
presetsstore.rescanforpresets(type);
};
-void Presets::deletepreset(int npreset){
+void Presets::deletepreset(int npreset)
+{
presetsstore.deletepreset(npreset);
};
diff --git a/src/Params/Presets.h b/src/Params/Presets.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Presets.h - Presets and Clipboard management
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,31 +28,32 @@
#include "PresetsStore.h"
/**Presets and Clipboard management*/
-class Presets{
- public:
- Presets();
- virtual ~Presets();
-
- void copy(const char *name);/**<if name==NULL, the clipboard is used*/
- void paste(int npreset);//npreset==0 for clipboard
- bool checkclipboardtype();
- void deletepreset(int npreset);
-
- char type[MAX_PRESETTYPE_SIZE];
- void setelement(int n);
-
- void rescanforpresets();
-
- protected:
- void setpresettype(const char *type);
- private:
- virtual void add2XML(XMLwrapper *xml)=0;
- virtual void getfromXML(XMLwrapper *xml)=0;
- virtual void defaults()=0;
- virtual void add2XMLsection(XMLwrapper *xml,int n){};
- virtual void getfromXMLsection(XMLwrapper *xml,int n){};
- virtual void defaults(int n){};
- int nelement;
+class Presets
+{
+public:
+ Presets();
+ virtual ~Presets();
+
+ void copy(const char *name);/**<if name==NULL, the clipboard is used*/
+ void paste(int npreset);//npreset==0 for clipboard
+ bool checkclipboardtype();
+ void deletepreset(int npreset);
+
+ char type[MAX_PRESETTYPE_SIZE];
+ void setelement(int n);
+
+ void rescanforpresets();
+
+protected:
+ void setpresettype(const char *type);
+private:
+ virtual void add2XML(XMLwrapper *xml)=0;
+ virtual void getfromXML(XMLwrapper *xml)=0;
+ virtual void defaults()=0;
+ virtual void add2XMLsection(XMLwrapper *xml,int n) {};
+ virtual void getfromXMLsection(XMLwrapper *xml,int n) {};
+ virtual void defaults(int n) {};
+ int nelement;
};
#endif
diff --git a/src/Params/PresetsStore.C b/src/Params/PresetsStore.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
PresetsStore.C - Presets and Clipboard store
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,130 +29,140 @@
PresetsStore presetsstore;
-PresetsStore::PresetsStore(){
+PresetsStore::PresetsStore()
+{
clipboard.data=NULL;
clipboard.type[0]=0;
-
- for (int i=0;i<MAX_PRESETS;i++){
- presets[i].file=NULL;
- presets[i].name=NULL;
+
+ for (int i=0;i<MAX_PRESETS;i++) {
+ presets[i].file=NULL;
+ presets[i].name=NULL;
};
-
+
};
-PresetsStore::~PresetsStore(){
+PresetsStore::~PresetsStore()
+{
if (clipboard.data!=NULL) free (clipboard.data);
clearpresets();
};
//Clipboard management
-void PresetsStore::copyclipboard(XMLwrapper *xml,char *type){
+void PresetsStore::copyclipboard(XMLwrapper *xml,char *type)
+{
strcpy(clipboard.type,type);
if (clipboard.data!=NULL) free (clipboard.data);
clipboard.data=xml->getXMLdata();
};
-bool PresetsStore::pasteclipboard(XMLwrapper *xml){
+bool PresetsStore::pasteclipboard(XMLwrapper *xml)
+{
if (clipboard.data!=NULL) xml->putXMLdata(clipboard.data);
- else return(false);
- return(true);
+ else return(false);
+ return(true);
};
-bool PresetsStore::checkclipboardtype(char *type){
+bool PresetsStore::checkclipboardtype(char *type)
+{
//makes LFO's compatible
if ((strstr(type,"Plfo")!=NULL)&&(strstr(clipboard.type,"Plfo")!=NULL)) return(true);
return(strcmp(type,clipboard.type)==0);
};
//Presets management
-void PresetsStore::clearpresets(){
- for (int i=0;i<MAX_PRESETS;i++){
- if (presets[i].file!=NULL) {
- delete(presets[i].file);
- presets[i].file=NULL;
- };
- if (presets[i].name!=NULL) {
- delete(presets[i].name);
- presets[i].name=NULL;
- };
+void PresetsStore::clearpresets()
+{
+ for (int i=0;i<MAX_PRESETS;i++) {
+ if (presets[i].file!=NULL) {
+ delete(presets[i].file);
+ presets[i].file=NULL;
+ };
+ if (presets[i].name!=NULL) {
+ delete(presets[i].name);
+ presets[i].name=NULL;
+ };
};
-
+
};
//a helper function that compares 2 presets[]
-int Presets_compar(const void *a,const void *b){
+int Presets_compar(const void *a,const void *b)
+{
struct PresetsStore::presetstruct *p1= (PresetsStore::presetstruct *)a;
struct PresetsStore::presetstruct *p2= (PresetsStore::presetstruct *)b;
if (((p1->name)==NULL)||((p2->name)==NULL)) return(0);
-
+
return(strcasecmp(p1->name,p2->name)<0);
};
-void PresetsStore::rescanforpresets(char *type){
+void PresetsStore::rescanforpresets(char *type)
+{
clearpresets();
int presetk=0;
char ftype[MAX_STRING_SIZE];
snprintf(ftype,MAX_STRING_SIZE,".%s.xpz",type);
-
- for (int i=0;i<MAX_BANK_ROOT_DIRS;i++){
- if (config.cfg.presetsDirList[i]==NULL) continue;
- char *dirname=config.cfg.presetsDirList[i];
- DIR *dir=opendir(dirname);
- if (dir==NULL) continue;
- struct dirent *fn;
- while((fn=readdir(dir))){
- const char *filename=fn->d_name;
- if (strstr(filename,ftype)==NULL) continue;
-
-
- presets[presetk].file=new char [MAX_STRING_SIZE];
- presets[presetk].name=new char [MAX_STRING_SIZE];
- char tmpc=dirname[strlen(dirname)-1];
- const char *tmps;
- if ((tmpc=='/')||(tmpc=='\\')) tmps="";
- else tmps="/";
- snprintf(presets[presetk].file,MAX_STRING_SIZE,"%s%s%s",dirname,tmps,filename);
- snprintf(presets[presetk].name,MAX_STRING_SIZE,"%s",filename);
-
- char *tmp=strstr(presets[presetk].name,ftype);
- if (tmp!=NULL) tmp[0]='\0';
- presetk++; if (presetk>=MAX_PRESETS) return;
- };
-
- closedir(dir);
+
+ for (int i=0;i<MAX_BANK_ROOT_DIRS;i++) {
+ if (config.cfg.presetsDirList[i]==NULL) continue;
+ char *dirname=config.cfg.presetsDirList[i];
+ DIR *dir=opendir(dirname);
+ if (dir==NULL) continue;
+ struct dirent *fn;
+ while ((fn=readdir(dir))) {
+ const char *filename=fn->d_name;
+ if (strstr(filename,ftype)==NULL) continue;
+
+
+ presets[presetk].file=new char [MAX_STRING_SIZE];
+ presets[presetk].name=new char [MAX_STRING_SIZE];
+ char tmpc=dirname[strlen(dirname)-1];
+ const char *tmps;
+ if ((tmpc=='/')||(tmpc=='\\')) tmps="";
+ else tmps="/";
+ snprintf(presets[presetk].file,MAX_STRING_SIZE,"%s%s%s",dirname,tmps,filename);
+ snprintf(presets[presetk].name,MAX_STRING_SIZE,"%s",filename);
+
+ char *tmp=strstr(presets[presetk].name,ftype);
+ if (tmp!=NULL) tmp[0]='\0';
+ presetk++;
+ if (presetk>=MAX_PRESETS) return;
+ };
+
+ closedir(dir);
};
//sort the presets
- for (int j=0;j<MAX_PRESETS-1;j++){
- for (int i=j+1;i<MAX_PRESETS;i++){
- if (Presets_compar(&presets[i],&presets[j])) {
- presetstruct tmp=presets[i];
- presets[i]=presets[j];
- presets[j]=tmp;
- };
- };
+ for (int j=0;j<MAX_PRESETS-1;j++) {
+ for (int i=j+1;i<MAX_PRESETS;i++) {
+ if (Presets_compar(&presets[i],&presets[j])) {
+ presetstruct tmp=presets[i];
+ presets[i]=presets[j];
+ presets[j]=tmp;
+ };
+ };
};
};
-void PresetsStore::copypreset(XMLwrapper *xml,char *type, const char *name){
+void PresetsStore::copypreset(XMLwrapper *xml,char *type, const char *name)
+{
char filename[MAX_STRING_SIZE],tmpfilename[MAX_STRING_SIZE];
-
+
if (config.cfg.presetsDirList[0]==NULL) return;
-
+
snprintf(tmpfilename,MAX_STRING_SIZE,"%s",name);
//make the filenames legal
for (int i=0;i<(int) strlen(tmpfilename);i++) {
- char c=tmpfilename[i];
- if ((c>='0')&&(c<='9')) continue;
- if ((c>='A')&&(c<='Z')) continue;
- if ((c>='a')&&(c<='z')) continue;
- if ((c=='-')||(c==' ')) continue;
- tmpfilename[i]='_';
+ char c=tmpfilename[i];
+ if ((c>='0')&&(c<='9')) continue;
+ if ((c>='A')&&(c<='Z')) continue;
+ if ((c>='a')&&(c<='z')) continue;
+ if ((c=='-')||(c==' ')) continue;
+ tmpfilename[i]='_';
};
-
+
const char *dirname=config.cfg.presetsDirList[0];
char tmpc=dirname[strlen(dirname)-1];
const char *tmps;
@@ -160,11 +170,12 @@ void PresetsStore::copypreset(XMLwrapper *xml,char *type, const char *name){
else tmps="/";
snprintf(filename,MAX_STRING_SIZE,"%s%s%s.%s.xpz",dirname,tmps,name,type);
-
+
xml->saveXMLfile(filename);
};
-bool PresetsStore::pastepreset(XMLwrapper *xml, int npreset){
+bool PresetsStore::pastepreset(XMLwrapper *xml, int npreset)
+{
npreset--;
if (npreset>=MAX_PRESETS) return(false);
char *filename=presets[npreset].file;
@@ -173,7 +184,8 @@ bool PresetsStore::pastepreset(XMLwrapper *xml, int npreset){
return(result);
};
-void PresetsStore::deletepreset(int npreset){
+void PresetsStore::deletepreset(int npreset)
+{
npreset--;
if (npreset>=MAX_PRESETS) return;
char *filename=presets[npreset].file;
diff --git a/src/Params/PresetsStore.h b/src/Params/PresetsStore.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
PresetsStore.C - Presets and Clipboard store
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,37 +26,38 @@
#define MAX_PRESETTYPE_SIZE 30
#define MAX_PRESETS 1000
-class PresetsStore{
- public:
- PresetsStore();
- ~PresetsStore();
-
- //Clipboard stuff
- void copyclipboard(XMLwrapper *xml,char *type);
- bool pasteclipboard(XMLwrapper *xml);
- bool checkclipboardtype(char *type);
-
- //presets stuff
- void copypreset(XMLwrapper *xml,char *type, const char *name);
- bool pastepreset(XMLwrapper *xml, int npreset);
- void deletepreset(int npreset);
-
- struct presetstruct{
- char *file;
- char *name;
- };
- presetstruct presets[MAX_PRESETS];
-
- void rescanforpresets(char *type);
-
- private:
- struct {
- char *data;
- char type[MAX_PRESETTYPE_SIZE];
- } clipboard;
-
- void clearpresets();
-
+class PresetsStore
+{
+public:
+ PresetsStore();
+ ~PresetsStore();
+
+ //Clipboard stuff
+ void copyclipboard(XMLwrapper *xml,char *type);
+ bool pasteclipboard(XMLwrapper *xml);
+ bool checkclipboardtype(char *type);
+
+ //presets stuff
+ void copypreset(XMLwrapper *xml,char *type, const char *name);
+ bool pastepreset(XMLwrapper *xml, int npreset);
+ void deletepreset(int npreset);
+
+ struct presetstruct {
+ char *file;
+ char *name;
+ };
+ presetstruct presets[MAX_PRESETS];
+
+ void rescanforpresets(char *type);
+
+private:
+ struct {
+ char *data;
+ char type[MAX_PRESETTYPE_SIZE];
+ } clipboard;
+
+ void clearpresets();
+
};
extern PresetsStore presetsstore;
diff --git a/src/Params/SUBnoteParameters.C b/src/Params/SUBnoteParameters.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
- SUBnoteParameters.C - Parameters for SUBnote (SUBsynth)
+
+ SUBnoteParameters.C - Parameters for SUBnote (SUBsynth)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,7 +24,8 @@
#include "SUBnoteParameters.h"
#include <stdio.h>
-SUBnoteParameters::SUBnoteParameters():Presets(){
+SUBnoteParameters::SUBnoteParameters():Presets()
+{
setpresettype("Psubsyth");
AmpEnvelope=new EnvelopeParams(64,1);
AmpEnvelope->ADSRinit_dB(0,40,127,25);
@@ -36,16 +37,17 @@ SUBnoteParameters::SUBnoteParameters():Presets(){
GlobalFilter=new FilterParams(2,80,40);
GlobalFilterEnvelope=new EnvelopeParams(0,1);
GlobalFilterEnvelope->ADSRinit_filter(64,40,64,70,60,64);
-
+
defaults();
};
-void SUBnoteParameters::defaults(){
- PVolume=96;
+void SUBnoteParameters::defaults()
+{
+ PVolume=96;
PPanning=64;
PAmpVelocityScaleFunction=90;
-
+
Pfixedfreq=0;
PfixedfreqET=0;
Pnumstages=2;
@@ -60,10 +62,10 @@ void SUBnoteParameters::defaults(){
PDetuneType=1;
PFreqEnvelopeEnabled=0;
PBandWidthEnvelopeEnabled=0;
-
- for (int n=0;n<MAX_SUB_HARMONICS;n++){
- Phmag[n]=0;
- Phrelbw[n]=64;
+
+ for (int n=0;n<MAX_SUB_HARMONICS;n++) {
+ Phmag[n]=0;
+ Phrelbw[n]=64;
};
Phmag[0]=127;
@@ -81,7 +83,8 @@ void SUBnoteParameters::defaults(){
-SUBnoteParameters::~SUBnoteParameters(){
+SUBnoteParameters::~SUBnoteParameters()
+{
delete (AmpEnvelope);
delete (FreqEnvelope);
delete (BandWidthEnvelope);
@@ -92,144 +95,146 @@ SUBnoteParameters::~SUBnoteParameters(){
-void SUBnoteParameters::add2XML(XMLwrapper *xml){
+void SUBnoteParameters::add2XML(XMLwrapper *xml)
+{
xml->addpar("num_stages",Pnumstages);
xml->addpar("harmonic_mag_type",Phmagtype);
xml->addpar("start",Pstart);
-
+
xml->beginbranch("HARMONICS");
- for (int i=0;i<MAX_SUB_HARMONICS;i++){
- if ((Phmag[i]==0)&&(xml->minimal)) continue;
- xml->beginbranch("HARMONIC",i);
- xml->addpar("mag",Phmag[i]);
- xml->addpar("relbw",Phrelbw[i]);
- xml->endbranch();
- };
+ for (int i=0;i<MAX_SUB_HARMONICS;i++) {
+ if ((Phmag[i]==0)&&(xml->minimal)) continue;
+ xml->beginbranch("HARMONIC",i);
+ xml->addpar("mag",Phmag[i]);
+ xml->addpar("relbw",Phrelbw[i]);
+ xml->endbranch();
+ };
xml->endbranch();
xml->beginbranch("AMPLITUDE_PARAMETERS");
- xml->addparbool("stereo",Pstereo);
- xml->addpar("volume",PVolume);
- xml->addpar("panning",PPanning);
- xml->addpar("velocity_sensing",PAmpVelocityScaleFunction);
- xml->beginbranch("AMPLITUDE_ENVELOPE");
- AmpEnvelope->add2XML(xml);
- xml->endbranch();
+ xml->addparbool("stereo",Pstereo);
+ xml->addpar("volume",PVolume);
+ xml->addpar("panning",PPanning);
+ xml->addpar("velocity_sensing",PAmpVelocityScaleFunction);
+ xml->beginbranch("AMPLITUDE_ENVELOPE");
+ AmpEnvelope->add2XML(xml);
+ xml->endbranch();
xml->endbranch();
xml->beginbranch("FREQUENCY_PARAMETERS");
- xml->addparbool("fixed_freq",Pfixedfreq);
- xml->addpar("fixed_freq_et",PfixedfreqET);
-
- xml->addpar("detune",PDetune);
- xml->addpar("coarse_detune",PCoarseDetune);
- xml->addpar("detune_type",PDetuneType);
-
- xml->addpar("bandwidth",Pbandwidth);
- xml->addpar("bandwidth_scale",Pbwscale);
-
- xml->addparbool("freq_envelope_enabled",PFreqEnvelopeEnabled);
- if ((PFreqEnvelopeEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("FREQUENCY_ENVELOPE");
- FreqEnvelope->add2XML(xml);
- xml->endbranch();
- };
-
- xml->addparbool("band_width_envelope_enabled",PBandWidthEnvelopeEnabled);
- if ((PBandWidthEnvelopeEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("BANDWIDTH_ENVELOPE");
- BandWidthEnvelope->add2XML(xml);
- xml->endbranch();
- };
+ xml->addparbool("fixed_freq",Pfixedfreq);
+ xml->addpar("fixed_freq_et",PfixedfreqET);
+
+ xml->addpar("detune",PDetune);
+ xml->addpar("coarse_detune",PCoarseDetune);
+ xml->addpar("detune_type",PDetuneType);
+
+ xml->addpar("bandwidth",Pbandwidth);
+ xml->addpar("bandwidth_scale",Pbwscale);
+
+ xml->addparbool("freq_envelope_enabled",PFreqEnvelopeEnabled);
+ if ((PFreqEnvelopeEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("FREQUENCY_ENVELOPE");
+ FreqEnvelope->add2XML(xml);
+ xml->endbranch();
+ };
+
+ xml->addparbool("band_width_envelope_enabled",PBandWidthEnvelopeEnabled);
+ if ((PBandWidthEnvelopeEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("BANDWIDTH_ENVELOPE");
+ BandWidthEnvelope->add2XML(xml);
+ xml->endbranch();
+ };
xml->endbranch();
xml->beginbranch("FILTER_PARAMETERS");
- xml->addparbool("enabled",PGlobalFilterEnabled);
- if ((PGlobalFilterEnabled!=0)||(!xml->minimal)){
- xml->beginbranch("FILTER");
- GlobalFilter->add2XML(xml);
- xml->endbranch();
-
- xml->addpar("filter_velocity_sensing",PGlobalFilterVelocityScaleFunction);
- xml->addpar("filter_velocity_sensing_amplitude",PGlobalFilterVelocityScale);
-
- xml->beginbranch("FILTER_ENVELOPE");
- GlobalFilterEnvelope->add2XML(xml);
- xml->endbranch();
- };
+ xml->addparbool("enabled",PGlobalFilterEnabled);
+ if ((PGlobalFilterEnabled!=0)||(!xml->minimal)) {
+ xml->beginbranch("FILTER");
+ GlobalFilter->add2XML(xml);
+ xml->endbranch();
+
+ xml->addpar("filter_velocity_sensing",PGlobalFilterVelocityScaleFunction);
+ xml->addpar("filter_velocity_sensing_amplitude",PGlobalFilterVelocityScale);
+
+ xml->beginbranch("FILTER_ENVELOPE");
+ GlobalFilterEnvelope->add2XML(xml);
+ xml->endbranch();
+ };
xml->endbranch();
};
-void SUBnoteParameters::getfromXML(XMLwrapper *xml){
+void SUBnoteParameters::getfromXML(XMLwrapper *xml)
+{
Pnumstages=xml->getpar127("num_stages",Pnumstages);
Phmagtype=xml->getpar127("harmonic_mag_type",Phmagtype);
Pstart=xml->getpar127("start",Pstart);
-
- if (xml->enterbranch("HARMONICS")){
- Phmag[0]=0;
- for (int i=0;i<MAX_SUB_HARMONICS;i++){
- if (xml->enterbranch("HARMONIC",i)==0) continue;
- Phmag[i]=xml->getpar127("mag",Phmag[i]);
- Phrelbw[i]=xml->getpar127("relbw",Phrelbw[i]);
- xml->exitbranch();
- };
- xml->exitbranch();
+
+ if (xml->enterbranch("HARMONICS")) {
+ Phmag[0]=0;
+ for (int i=0;i<MAX_SUB_HARMONICS;i++) {
+ if (xml->enterbranch("HARMONIC",i)==0) continue;
+ Phmag[i]=xml->getpar127("mag",Phmag[i]);
+ Phrelbw[i]=xml->getpar127("relbw",Phrelbw[i]);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
- if (xml->enterbranch("AMPLITUDE_PARAMETERS")){
- Pstereo=xml->getparbool("stereo",Pstereo);
- PVolume=xml->getpar127("volume",PVolume);
- PPanning=xml->getpar127("panning",PPanning);
- PAmpVelocityScaleFunction=xml->getpar127("velocity_sensing",PAmpVelocityScaleFunction);
- if (xml->enterbranch("AMPLITUDE_ENVELOPE")){
- AmpEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
- xml->exitbranch();
+ if (xml->enterbranch("AMPLITUDE_PARAMETERS")) {
+ Pstereo=xml->getparbool("stereo",Pstereo);
+ PVolume=xml->getpar127("volume",PVolume);
+ PPanning=xml->getpar127("panning",PPanning);
+ PAmpVelocityScaleFunction=xml->getpar127("velocity_sensing",PAmpVelocityScaleFunction);
+ if (xml->enterbranch("AMPLITUDE_ENVELOPE")) {
+ AmpEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
- 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);
- 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);
-
- PFreqEnvelopeEnabled=xml->getparbool("freq_envelope_enabled",PFreqEnvelopeEnabled);
- if (xml->enterbranch("FREQUENCY_ENVELOPE")){
- FreqEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
- PBandWidthEnvelopeEnabled=xml->getparbool("band_width_envelope_enabled",PBandWidthEnvelopeEnabled);
- if (xml->enterbranch("BANDWIDTH_ENVELOPE")){
- BandWidthEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
+ 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);
+ 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);
+
+ PFreqEnvelopeEnabled=xml->getparbool("freq_envelope_enabled",PFreqEnvelopeEnabled);
+ if (xml->enterbranch("FREQUENCY_ENVELOPE")) {
+ FreqEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ PBandWidthEnvelopeEnabled=xml->getparbool("band_width_envelope_enabled",PBandWidthEnvelopeEnabled);
+ if (xml->enterbranch("BANDWIDTH_ENVELOPE")) {
+ BandWidthEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
xml->exitbranch();
};
-
- if (xml->enterbranch("FILTER_PARAMETERS")){
- PGlobalFilterEnabled=xml->getparbool("enabled",PGlobalFilterEnabled);
- if (xml->enterbranch("FILTER")){
- GlobalFilter->getfromXML(xml);
- xml->exitbranch();
- };
-
- PGlobalFilterVelocityScaleFunction=xml->getpar127("filter_velocity_sensing",PGlobalFilterVelocityScaleFunction);
- PGlobalFilterVelocityScale=xml->getpar127("filter_velocity_sensing_amplitude",PGlobalFilterVelocityScale);
-
- if (xml->enterbranch("FILTER_ENVELOPE")){
- GlobalFilterEnvelope->getfromXML(xml);
- xml->exitbranch();
- };
-
- xml->exitbranch();
+
+ if (xml->enterbranch("FILTER_PARAMETERS")) {
+ PGlobalFilterEnabled=xml->getparbool("enabled",PGlobalFilterEnabled);
+ if (xml->enterbranch("FILTER")) {
+ GlobalFilter->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ PGlobalFilterVelocityScaleFunction=xml->getpar127("filter_velocity_sensing",PGlobalFilterVelocityScaleFunction);
+ PGlobalFilterVelocityScale=xml->getpar127("filter_velocity_sensing_amplitude",PGlobalFilterVelocityScale);
+
+ if (xml->enterbranch("FILTER_ENVELOPE")) {
+ GlobalFilterEnvelope->getfromXML(xml);
+ xml->exitbranch();
+ };
+
+ xml->exitbranch();
};
};
diff --git a/src/Params/SUBnoteParameters.h b/src/Params/SUBnoteParameters.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
- SUBnoteParameters.h - Parameters for SUBnote (SUBsynth)
+
+ SUBnoteParameters.h - Parameters for SUBnote (SUBsynth)
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,74 +29,75 @@
#include "FilterParams.h"
#include "Presets.h"
-class SUBnoteParameters:public Presets{
- public:
- SUBnoteParameters();
- ~SUBnoteParameters();
-
- void add2XML(XMLwrapper *xml);
- void defaults();
- void getfromXML(XMLwrapper *xml);
-
- //Parameters
- //AMPLITUDE PARAMETRERS
- unsigned char Pstereo;//0 for mono,1 for stereo
- unsigned char PVolume;
- unsigned char PPanning;
- unsigned char PAmpVelocityScaleFunction;
- EnvelopeParams *AmpEnvelope;
-
- //Frequency Parameters
- unsigned short int PDetune;
- unsigned short int PCoarseDetune;
- unsigned char PDetuneType;
- unsigned char PFreqEnvelopeEnabled;
- EnvelopeParams *FreqEnvelope;
- unsigned char PBandWidthEnvelopeEnabled;
- EnvelopeParams *BandWidthEnvelope;
-
- //Filter Parameters (Global)
- unsigned char PGlobalFilterEnabled;
- FilterParams *GlobalFilter;
- unsigned char PGlobalFilterVelocityScale;
- unsigned char PGlobalFilterVelocityScaleFunction;
- EnvelopeParams *GlobalFilterEnvelope;
-
-
- //Other Parameters
-
- //If the base frequency is fixed to 440 Hz
- unsigned char Pfixedfreq;
-
- /* 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;
-
-
- //how many times the filters are applied
- unsigned char Pnumstages;
-
- //bandwidth
- unsigned char Pbandwidth;
-
- //How the magnitudes are computed (0=linear,1=-60dB,2=-60dB)
- unsigned char Phmagtype;
-
- //Magnitudes
- unsigned char Phmag[MAX_SUB_HARMONICS];
-
- //Relative BandWidth ("64"=1.0)
- unsigned char Phrelbw[MAX_SUB_HARMONICS];
-
- //how much the bandwidth is increased according to lower/higher frequency; 64-default
- unsigned char Pbwscale;
-
- //how the harmonics start("0"=0,"1"=random,"2"=1)
- unsigned char Pstart;
-
-
- private:
+class SUBnoteParameters:public Presets
+{
+public:
+ SUBnoteParameters();
+ ~SUBnoteParameters();
+
+ void add2XML(XMLwrapper *xml);
+ void defaults();
+ void getfromXML(XMLwrapper *xml);
+
+ //Parameters
+ //AMPLITUDE PARAMETRERS
+ unsigned char Pstereo;//0 for mono,1 for stereo
+ unsigned char PVolume;
+ unsigned char PPanning;
+ unsigned char PAmpVelocityScaleFunction;
+ EnvelopeParams *AmpEnvelope;
+
+ //Frequency Parameters
+ unsigned short int PDetune;
+ unsigned short int PCoarseDetune;
+ unsigned char PDetuneType;
+ unsigned char PFreqEnvelopeEnabled;
+ EnvelopeParams *FreqEnvelope;
+ unsigned char PBandWidthEnvelopeEnabled;
+ EnvelopeParams *BandWidthEnvelope;
+
+ //Filter Parameters (Global)
+ unsigned char PGlobalFilterEnabled;
+ FilterParams *GlobalFilter;
+ unsigned char PGlobalFilterVelocityScale;
+ unsigned char PGlobalFilterVelocityScaleFunction;
+ EnvelopeParams *GlobalFilterEnvelope;
+
+
+ //Other Parameters
+
+ //If the base frequency is fixed to 440 Hz
+ unsigned char Pfixedfreq;
+
+ /* 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;
+
+
+ //how many times the filters are applied
+ unsigned char Pnumstages;
+
+ //bandwidth
+ unsigned char Pbandwidth;
+
+ //How the magnitudes are computed (0=linear,1=-60dB,2=-60dB)
+ unsigned char Phmagtype;
+
+ //Magnitudes
+ unsigned char Phmag[MAX_SUB_HARMONICS];
+
+ //Relative BandWidth ("64"=1.0)
+ unsigned char Phrelbw[MAX_SUB_HARMONICS];
+
+ //how much the bandwidth is increased according to lower/higher frequency; 64-default
+ unsigned char Pbwscale;
+
+ //how the harmonics start("0"=0,"1"=random,"2"=1)
+ unsigned char Pstart;
+
+
+private:
};
#endif
diff --git a/src/Samples/AuSample.C b/src/Samples/AuSample.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
AuSample.h - Object for storing information on audio samples (for one channel)
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -21,8 +21,8 @@
#include "AuSample.h"
AuSample::AuSample(const int &length)
- : Sample(length){}
+ : Sample(length) {}
AuSample::AuSample(float *input,const int &length)
- : Sample(input,length){}
+ : Sample(input,length) {}
diff --git a/src/Samples/AuSample.h b/src/Samples/AuSample.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
AuSample.h - Object for storing information on audio samples (for one channel)
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,10 +26,10 @@
class AuSample : public Sample
{
- public:
- AuSample(const int &length);
- AuSample(float *input,const int &length);
- FqSample getFqSample();/**\todo implement this*/
+public:
+ AuSample(const int &length);
+ AuSample(float *input,const int &length);
+ FqSample getFqSample();/**\todo implement this*/
};
#endif
diff --git a/src/Samples/FqSample.C b/src/Samples/FqSample.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
FqSample.C - Object for storing information on samples
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -21,11 +21,11 @@
#include "FqSample.h"
FqSample::FqSample(const int &length)
- :Sample(length)
+ :Sample(length)
{}
FqSample::FqSample(float *input,const int &length)
- : Sample(input,length)
+ : Sample(input,length)
{}
FqSample::~FqSample()
diff --git a/src/Samples/FqSample.h b/src/Samples/FqSample.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
FqSample.h - Object for storing information on samples
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,12 +26,12 @@
class FqSample : public Sample
{
- public:
- FqSample(const int &length);
- FqSample(float *input,const int &length);
- ~FqSample();
- //FqSample &operator=(const FqSample &smp);
- //float *dontuse(){return buffer;};
+public:
+ FqSample(const int &length);
+ FqSample(float *input,const int &length);
+ ~FqSample();
+ //FqSample &operator=(const FqSample &smp);
+ //float *dontuse(){return buffer;};
};
#endif
diff --git a/src/Samples/Sample.C b/src/Samples/Sample.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Sample.C - Object for storing information on samples
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -21,31 +21,30 @@
#include "Sample.h"
Sample::Sample(const Sample &smp)
- : bufferSize(smp.bufferSize)
+ : bufferSize(smp.bufferSize)
{
buffer=new float[bufferSize];
- for(int i=0;i<bufferSize;++i)
+ for (int i=0;i<bufferSize;++i)
*(i+buffer)=*(i+smp.buffer);
}
-
+
Sample::Sample(const int &length)
- : bufferSize(length)
+ : bufferSize(length)
{
- if(length<1)
- bufferSize=1;
+ if (length<1)
+ bufferSize=1;
buffer=new float[bufferSize];
clear();
}
Sample::Sample(float *input,const int &length)
- : bufferSize(length)
+ : bufferSize(length)
{
- if(length>0){
+ if (length>0) {
buffer=new float[length];
- for(int i=0;i<length;++i)
+ for (int i=0;i<length;++i)
*(buffer+i)=*(input+i);
- }
- else{
+ } else {
buffer=new float[1];
bufferSize=1;
*buffer=0;
@@ -54,29 +53,26 @@ Sample::Sample(float *input,const int &length)
Sample::~Sample()
{
- delete[] buffer;
+ delete[] buffer;
}
void Sample::clear()
{
- for(int i=0;i<bufferSize;++i)
+ for (int i=0;i<bufferSize;++i)
*(i+buffer)=0;
}
void Sample::operator=(const Sample &smp)
{
/**\todo rewrite to be less repetitive*/
- if(bufferSize==smp.bufferSize)
- {
- for(int i=0;i<bufferSize;++i)
+ if (bufferSize==smp.bufferSize) {
+ for (int i=0;i<bufferSize;++i)
*(i+buffer)=*(i+smp.buffer);
- }
- else
- {
+ } else {
delete[] buffer;
buffer=new float[smp.bufferSize];
bufferSize=smp.bufferSize;
- for(int i=0;i<bufferSize;++i)
+ for (int i=0;i<bufferSize;++i)
*(i+buffer)=*(i+smp.buffer);
}
}
diff --git a/src/Samples/Sample.h b/src/Samples/Sample.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Sample.h - Object for storing information on samples
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,31 +24,40 @@
/**
* Base Class for Samples
*/
-class Sample{
- public:
- Sample(const Sample &smp);
- Sample(const int &length);
- Sample(float *input,const int &length);
- ~Sample();
- /**Fills the buffer with zeros*/
- void clear();
- /**States the size of the buffer
- * @return the size of the buffer*/
- int size() const {return bufferSize;};
- /**Provides the indexing operator for non const Samples*/
- float &operator[](int index){return *(buffer+index%bufferSize);};
- /**Provides the indexing operator for const Samples*/
- const float &operator[](int index)const{return *(buffer+index%bufferSize);};
- /**Provides the assignment operator*/
- void operator=(const Sample &smp);
- /**Provides direct access to the buffer to allow for transition
- *
- * This method should be removed to ensure encapsulation once
- * it is integrated into the code*/
- float *dontuse(){return buffer;};
- private:
- int bufferSize;
- float *buffer;
+class Sample
+{
+public:
+ Sample(const Sample &smp);
+ Sample(const int &length);
+ Sample(float *input,const int &length);
+ ~Sample();
+ /**Fills the buffer with zeros*/
+ void clear();
+ /**States the size of the buffer
+ * @return the size of the buffer*/
+ int size() const {
+ return bufferSize;
+ };
+ /**Provides the indexing operator for non const Samples*/
+ float &operator[](int index) {
+ return *(buffer+index%bufferSize);
+ };
+ /**Provides the indexing operator for const Samples*/
+ const float &operator[](int index)const {
+ return *(buffer+index%bufferSize);
+ };
+ /**Provides the assignment operator*/
+ void operator=(const Sample &smp);
+ /**Provides direct access to the buffer to allow for transition
+ *
+ * This method should be removed to ensure encapsulation once
+ * it is integrated into the code*/
+ float *dontuse() {
+ return buffer;
+ };
+private:
+ int bufferSize;
+ float *buffer;
};
diff --git a/src/Seq/MIDIEvents.C b/src/Seq/MIDIEvents.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
MIDIEvents.C - It stores the midi events from midi file or sequencer
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,30 +24,34 @@
#include <stdlib.h>
#include <stdio.h>
-MIDIEvents::MIDIEvents(){
+MIDIEvents::MIDIEvents()
+{
};
-MIDIEvents::~MIDIEvents(){
+MIDIEvents::~MIDIEvents()
+{
};
/************** Track stuff ***************/
-void MIDIEvents::writeevent(list *l,event *ev){
+void MIDIEvents::writeevent(list *l,event *ev)
+{
listpos *tmp=new listpos;
tmp->next=NULL;
tmp->ev=*ev;
if (l->current!=NULL) l->current->next=tmp;
- else l->first=tmp;
+ else l->first=tmp;
l->current=tmp;
// printf("Wx%x ",(int) l->current);
// printf("-> %d \n",l->current->ev.deltatime);
l->size++;
};
-void MIDIEvents::readevent(list *l,event *ev){
+void MIDIEvents::readevent(list *l,event *ev)
+{
if (l->current==NULL) {
- ev->type=-1;
- return;
+ ev->type=-1;
+ return;
};
*ev=l->current->ev;
l->current=l->current->next;
@@ -63,22 +67,25 @@ void MIDIEvents::readevent(list *l,event *ev){
};
-void MIDIEvents::rewindlist(list *l){
+void MIDIEvents::rewindlist(list *l)
+{
l->current=l->first;
};
-void MIDIEvents::deletelist(list *l){
+void MIDIEvents::deletelist(list *l)
+{
l->current=l->first;
if (l->current==NULL) return;
- while (l->current->next!=NULL){
- listpos *tmp=l->current;
- l->current=l->current->next;
- delete(tmp);
+ while (l->current->next!=NULL) {
+ listpos *tmp=l->current;
+ l->current=l->current->next;
+ delete(tmp);
};
deletelistreference(l);
};
-void MIDIEvents::deletelistreference(list *l){
+void MIDIEvents::deletelistreference(list *l)
+{
l->current=l->first=NULL;
l->size=0;
l->length=0.0;
diff --git a/src/Seq/MIDIEvents.h b/src/Seq/MIDIEvents.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
MIDIEvents.h - It stores the midi events from midi file or sequencer
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,37 +28,38 @@
/**storage the midi events from midi file or sequencer
* \todo this looks quite like a remake of a linked list
* if it is, then it should be rewritten to use <list>*/
-class MIDIEvents{
+class MIDIEvents
+{
friend class MIDIFile;
- public:
- MIDIEvents();
- ~MIDIEvents();
+public:
+ MIDIEvents();
+ ~MIDIEvents();
- protected:
+protected:
/* Events */
- struct event{
+ struct event {
int deltatime;
- int channel;//on what midi channel is
- int type,par1,par2;//type=1 for note, type=2 for controller, type=255 for time messages
+ int channel;//on what midi channel is
+ int type,par1,par2;//type=1 for note, type=2 for controller, type=255 for time messages
} tmpevent;
- struct listpos{
- event ev;
+ struct listpos {
+ event ev;
struct listpos *next;
};
- struct list{
- listpos *first,*current;
- int size;//how many events are
- double length;//in seconds
+ struct list {
+ listpos *first,*current;
+ int size;//how many events are
+ double length;//in seconds
};
struct {
- list track;//the stored track
- list record;//the track being "recorded" from midi
+ list track;//the stored track
+ list record;//the track being "recorded" from midi
} miditrack[NUM_MIDI_TRACKS];
-
+
void writeevent(list *l,event *ev);
void readevent(list *l,event *ev);
-
+
void rewindlist(list *l);
void deletelist(list *l);
void deletelistreference(list *l);
diff --git a/src/Seq/MIDIFile.C b/src/Seq/MIDIFile.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
MIDIFile.C - MIDI file loader
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -25,7 +25,8 @@
#include "MIDIFile.h"
-MIDIFile::MIDIFile(){
+MIDIFile::MIDIFile()
+{
midifile=NULL;
midifilesize=0;
midifilek=0;
@@ -33,31 +34,33 @@ MIDIFile::MIDIFile(){
me=NULL;
};
-MIDIFile::~MIDIFile(){
+MIDIFile::~MIDIFile()
+{
clearmidifile();
};
-int MIDIFile::loadfile(const char *filename){
+int MIDIFile::loadfile(const char *filename)
+{
clearmidifile();
-
+
FILE *file=fopen(filename,"r");
if (file==NULL) return(-1);
-
+
char header[4];
ZERO(header,4);
fread(header,4,1,file);
//test to see if this a midi file
- if ((header[0]!='M')||(header[1]!='T')||(header[2]!='h')||(header[3]!='d')){
- fclose(file);
- return(-1);
+ if ((header[0]!='M')||(header[1]!='T')||(header[2]!='h')||(header[3]!='d')) {
+ fclose(file);
+ return(-1);
};
-
+
//get the filesize
fseek(file,0,SEEK_END);
midifilesize=ftell(file);
rewind(file);
-
+
midifile=new unsigned char[midifilesize];
ZERO(midifile,midifilesize);
fread(midifile,midifilesize,1,file);
@@ -65,12 +68,13 @@ int MIDIFile::loadfile(const char *filename){
// for (int i=0;i<midifilesize;i++) printf("%2x ",midifile[i]);
// printf("\n");
-
-
+
+
return(0);
};
-int MIDIFile::parsemidifile(MIDIEvents *me_){
+int MIDIFile::parsemidifile(MIDIEvents *me_)
+{
this->me=me_;
//read the header
@@ -88,19 +92,19 @@ int MIDIFile::parsemidifile(MIDIEvents *me_){
int division=getint16();
printf("division %d\n",division);
- if (division>=0){//delta time units in each a quater note
+ if (division>=0) {//delta time units in each a quater note
// tick=???;
} else {//SMPTE (frames/second and ticks/frame)
- printf("ERROR:in MIDIFile.C::parsemidifile() - SMPTE not implemented yet.");
- };
-
+ printf("ERROR:in MIDIFile.C::parsemidifile() - SMPTE not implemented yet.");
+ };
+
if (ntracks>=NUM_MIDI_TRACKS) ntracks=NUM_MIDI_TRACKS-1;
-
- for (int n=0;n<ntracks;n++){
- if (parsetrack(n)<0) {
- clearmidifile();
- return(-1);
- };
+
+ for (int n=0;n<ntracks;n++) {
+ if (parsetrack(n)<0) {
+ clearmidifile();
+ return(-1);
+ };
};
printf("\n\nCURRENT File position is = 0x%x\n",midifilek);
@@ -114,7 +118,8 @@ int MIDIFile::parsemidifile(MIDIEvents *me_){
//private members
-int MIDIFile::parsetrack(int ntrack){
+int MIDIFile::parsetrack(int ntrack)
+{
printf("\n--==*Reading track %d **==--\n",ntrack);
int chunk=getint32();//MTrk
@@ -127,82 +132,82 @@ int MIDIFile::parsetrack(int ntrack){
unsigned char lastmsg=0;
unsigned int dt=0;
-
- while(!midieof){
- unsigned int msgdeltatime=getvarint32();
-
+
+ while (!midieof) {
+ unsigned int msgdeltatime=getvarint32();
+
/// printf("MSGDELTATIME = %d\n",msgdeltatime);
-
+
// dt+=msgdeltatime;
- int msg=peekbyte();
+ int msg=peekbyte();
/// printf("raw msg=0x%x ",msg);
- if (msg<0x80) {
- msg=lastmsg;
- } else {
- lastmsg=msg;
- getbyte();
- };
+ if (msg<0x80) {
+ msg=lastmsg;
+ } else {
+ lastmsg=msg;
+ getbyte();
+ };
/// printf("msg=0x%x\n",msg);
// dt+=msgdeltatime;
- add_dt(ntrack, msgdeltatime);
-
- unsigned int mtype,mlength;
-
- switch(msg){
- case 0x80 ... 0x8f://note on off
- parsenoteoff(ntrack,msg & 0x0f,dt);
- dt=0;
- break;
- case 0x90 ... 0x9f://note on (or note off)
- parsenoteon(ntrack,msg & 0x0f,dt);
- dt=0;
- break;
- case 0xa0 ... 0xaf://aftertouch - ignored
- skipnbytes(2);
- break;
- case 0xb0 ... 0xbf://control change
- parsecontrolchange(ntrack,msg & 0x0f,dt);
- dt=0;
- break;
- case 0xc0 ... 0xcf://program change - ignored
- skipnbytes(1);
- break;
- case 0xd0 ... 0xdf://channel pressure - ignored
- skipnbytes(1);
- break;
- case 0xe0 ... 0xef://channel mode messages
- skipnbytes(2);
- break;
- case 0xf0://sysex - ignored
- while (getbyte()!=0xf7){
- if (midieof) break;
- };
- break;
- case 0xf7://sysex (another type) - ignored
- skipnbytes(getvarint32());
- break;
-
- case 0xff://meta-event
- mtype=getbyte();
- mlength=getbyte();
- parsemetaevent(mtype,mlength);
- break;
-
- default:
- getbyte();
- printf("UNKNOWN message! 0x%x\n",msg);
- return(-1);
- break;
- };
-
-
-
- if (midieof) return(-1);
-
- if ((midifilek-oldmidifilek)==size) break;
- else if((midifilek-oldmidifilek)>size) return(-1);
+ add_dt(ntrack, msgdeltatime);
+
+ unsigned int mtype,mlength;
+
+ switch (msg) {
+ case 0x80 ... 0x8f://note on off
+ parsenoteoff(ntrack,msg & 0x0f,dt);
+ dt=0;
+ break;
+ case 0x90 ... 0x9f://note on (or note off)
+ parsenoteon(ntrack,msg & 0x0f,dt);
+ dt=0;
+ break;
+ case 0xa0 ... 0xaf://aftertouch - ignored
+ skipnbytes(2);
+ break;
+ case 0xb0 ... 0xbf://control change
+ parsecontrolchange(ntrack,msg & 0x0f,dt);
+ dt=0;
+ break;
+ case 0xc0 ... 0xcf://program change - ignored
+ skipnbytes(1);
+ break;
+ case 0xd0 ... 0xdf://channel pressure - ignored
+ skipnbytes(1);
+ break;
+ case 0xe0 ... 0xef://channel mode messages
+ skipnbytes(2);
+ break;
+ case 0xf0://sysex - ignored
+ while (getbyte()!=0xf7) {
+ if (midieof) break;
+ };
+ break;
+ case 0xf7://sysex (another type) - ignored
+ skipnbytes(getvarint32());
+ break;
+
+ case 0xff://meta-event
+ mtype=getbyte();
+ mlength=getbyte();
+ parsemetaevent(mtype,mlength);
+ break;
+
+ default:
+ getbyte();
+ printf("UNKNOWN message! 0x%x\n",msg);
+ return(-1);
+ break;
+ };
+
+
+
+ if (midieof) return(-1);
+
+ if ((midifilek-oldmidifilek)==size) break;
+ else if ((midifilek-oldmidifilek)>size) return(-1);
// if (size!=6) return(-1);//header is always 6 bytes long
};
@@ -212,40 +217,42 @@ int MIDIFile::parsetrack(int ntrack){
};
-void MIDIFile::parsenoteoff(char ntrack,char chan,unsigned int dt){
+void MIDIFile::parsenoteoff(char ntrack,char chan,unsigned int dt)
+{
unsigned char note;
note=getbyte();
-
+
unsigned char noteoff_velocity=getbyte();//unused by zynaddsubfx
noteoff_velocity=0;
if (chan>=NUM_MIDI_CHANNELS) return;
-
+
me->tmpevent.deltatime=convertdt(dt);
me->tmpevent.type=1;
me->tmpevent.par1=note;
me->tmpevent.par2=0;
me->tmpevent.channel=chan;
-
+
printf("Note off:%d \n",note);
-
- ///test
+
+ ///test
// ntrack=0;
-
+
me->writeevent(&me->miditrack[(int)ntrack].record,&me->tmpevent);
-
+
};
-void MIDIFile::parsenoteon(char ntrack,char chan,unsigned int dt){
+void MIDIFile::parsenoteon(char ntrack,char chan,unsigned int dt)
+{
unsigned char note,vel;
note=getbyte();
vel=getbyte();
-
+
// printf("ntrack=%d\n",ntrack);
printf("[dt %d ] Note on:%d %d\n",dt,note,vel);
-
+
if (chan>=NUM_MIDI_CHANNELS) return;
-
+
me->tmpevent.deltatime=convertdt(dt);
me->tmpevent.type=1;
me->tmpevent.par1=note;
@@ -253,52 +260,56 @@ void MIDIFile::parsenoteon(char ntrack,char chan,unsigned int dt){
me->tmpevent.channel=chan;
me->writeevent(&me->miditrack[(int)ntrack].record,&me->tmpevent);
-
-
+
+
};
-void MIDIFile::parsecontrolchange(char ntrack,char chan,unsigned int dt){
+void MIDIFile::parsecontrolchange(char ntrack,char chan,unsigned int dt)
+{
unsigned char control,value;
control=getbyte();
value=getbyte();
-
+
if (chan>=NUM_MIDI_CHANNELS) return;
-
+
printf("[dt %d] Control change:%d %d\n",dt,control,value);
-
+
me->tmpevent.deltatime=convertdt(dt);
me->tmpevent.type=2;
me->tmpevent.par1=control;//???????????? ma uit la Sequencer::recordnote() din varianele vechi de zyn
me->tmpevent.par2=value;
me->tmpevent.channel=chan;
me->writeevent(&me->miditrack[(int)ntrack].record,&me->tmpevent);
-
+
};
-void MIDIFile::parsepitchwheel(char ntrack,char chan, unsigned int dt){
+void MIDIFile::parsepitchwheel(char ntrack,char chan, unsigned int dt)
+{
unsigned char valhi,vallo;
vallo=getbyte();
valhi=getbyte();
-
+
if (chan>=NUM_MIDI_CHANNELS) return;
-
+
int value=(int)valhi*128+vallo;
-
+
printf("[dt %d] Pitch wheel:%d\n",dt,value);
-
+
};
-void MIDIFile::parsemetaevent(unsigned char mtype,unsigned char mlength){
+void MIDIFile::parsemetaevent(unsigned char mtype,unsigned char mlength)
+{
int oldmidifilek=midifilek;
printf("meta-event type=0x%x length=%d\n",mtype,mlength);
-
+
midifilek=oldmidifilek+mlength;
-
+
};
-void MIDIFile::add_dt(char ntrack, unsigned int dt){
+void MIDIFile::add_dt(char ntrack, unsigned int dt)
+{
me->tmpevent.deltatime=convertdt(dt);
me->tmpevent.type=255;
me->tmpevent.par1=0;
@@ -308,7 +319,8 @@ void MIDIFile::add_dt(char ntrack, unsigned int dt){
};
-unsigned int MIDIFile::convertdt(unsigned int dt){
+unsigned int MIDIFile::convertdt(unsigned int dt)
+{
double result=dt;
printf("DT=%d\n",dt);
@@ -316,7 +328,8 @@ unsigned int MIDIFile::convertdt(unsigned int dt){
};
-void MIDIFile::clearmidifile(){
+void MIDIFile::clearmidifile()
+{
if (midifile!=NULL) delete(midifile);
midifile=NULL;
midifilesize=0;
@@ -325,43 +338,48 @@ void MIDIFile::clearmidifile(){
data.tick=0.05;
};
-unsigned char MIDIFile::getbyte(){
+unsigned char MIDIFile::getbyte()
+{
if (midifilek>=midifilesize) {
- midieof=true;
- return(0);
+ midieof=true;
+ return(0);
};
/// printf("(%d) ",midifile[midifilek]);
return(midifile[midifilek++]);
};
-unsigned char MIDIFile::peekbyte(){
+unsigned char MIDIFile::peekbyte()
+{
if (midifilek>=midifilesize) {
- midieof=true;
- return(0);
+ midieof=true;
+ return(0);
};
return(midifile[midifilek]);
};
-unsigned int MIDIFile::getint32(){
+unsigned int MIDIFile::getint32()
+{
unsigned int result=0;
for (int i=0;i<4;i++) {
- result=result*256+getbyte();
+ result=result*256+getbyte();
};
if (midieof) result=0;
return(result);
};
-unsigned short int MIDIFile::getint16(){
+unsigned short int MIDIFile::getint16()
+{
unsigned short int result=0;
for (int i=0;i<2;i++) {
- result=result*256+getbyte();
+ result=result*256+getbyte();
};
if (midieof) result=0;
return(result);
};
-unsigned int MIDIFile::getvarint32(){
+unsigned int MIDIFile::getvarint32()
+{
unsigned long result=0;
unsigned char b;
@@ -370,20 +388,21 @@ unsigned int MIDIFile::getvarint32(){
if ((result = getbyte()) & 0x80) {
result &= 0x7f;
do {
- b=getbyte();
+ b=getbyte();
result = (result << 7) + (b & 0x7f);
- }while (b & 0x80);
+ } while (b & 0x80);
}
/// printf("[end - result= %d]\n",result);
return result;
};
-void MIDIFile::skipnbytes(int n){
+void MIDIFile::skipnbytes(int n)
+{
midifilek+=n;
- if (midifilek>=midifilesize){
- midifilek=midifilesize-1;
- midieof=true;
+ if (midifilek>=midifilesize) {
+ midifilek=midifilesize-1;
+ midieof=true;
};
};
diff --git a/src/Seq/MIDIFile.h b/src/Seq/MIDIFile.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
MIDIFile.h - MIDI file loader
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,67 +27,68 @@
#include "MIDIEvents.h"
/**MIDI file loader*/
-class MIDIFile{
- public:
- MIDIFile();
- ~MIDIFile();
-
- /**Loads the given file
+class MIDIFile
+{
+public:
+ MIDIFile();
+ ~MIDIFile();
+
+ /**Loads the given file
* @param filename The name of the file to load
* @return -1 if there is an error, otherwise 0*/
- int loadfile(const char *filename);
-
- //returns -1 if there is an error, otherwise 0
- int parsemidifile(MIDIEvents *me_);
-
- private:
- MIDIEvents *me;
-
- unsigned char *midifile;
- int midifilesize,midifilek;
- bool midieof;
-
- //returns -1 if there is an error, otherwise 0
- int parsetrack(int ntrack);
-
- void parsenoteoff(char ntrack,char chan,unsigned int dt);
- void parsenoteon(char ntrack,char chan,unsigned int dt);
- void parsecontrolchange(char ntrack,char chan,unsigned int dt);
- void parsepitchwheel(char ntrack,char chan, unsigned int dt);
- void parsemetaevent(unsigned char mtype,unsigned char mlength);
-
- void add_dt(char ntrack, unsigned int dt);
-
- void clearmidifile();
-
- //convert the delta-time to internal format
- unsigned int convertdt(unsigned int dt);
-
- /* Low Level MIDIfile functions */
-
- //get a byte from the midifile
- unsigned char getbyte();
-
- //peek the current byte from the midifile
- unsigned char peekbyte();
-
- //get a set of 4 bytes from the midifile
- unsigned int getint32();
-
- //get a word of 2 bytes from the midifile
- unsigned short int getint16();
-
- //read a variable length quantity
- unsigned int getvarint32();
-
- //skip some bytes
- void skipnbytes(int n);
-
- struct {
- double tick;//how many seconds one tick has
-
- }data;
-
+ int loadfile(const char *filename);
+
+ //returns -1 if there is an error, otherwise 0
+ int parsemidifile(MIDIEvents *me_);
+
+private:
+ MIDIEvents *me;
+
+ unsigned char *midifile;
+ int midifilesize,midifilek;
+ bool midieof;
+
+ //returns -1 if there is an error, otherwise 0
+ int parsetrack(int ntrack);
+
+ void parsenoteoff(char ntrack,char chan,unsigned int dt);
+ void parsenoteon(char ntrack,char chan,unsigned int dt);
+ void parsecontrolchange(char ntrack,char chan,unsigned int dt);
+ void parsepitchwheel(char ntrack,char chan, unsigned int dt);
+ void parsemetaevent(unsigned char mtype,unsigned char mlength);
+
+ void add_dt(char ntrack, unsigned int dt);
+
+ void clearmidifile();
+
+ //convert the delta-time to internal format
+ unsigned int convertdt(unsigned int dt);
+
+ /* Low Level MIDIfile functions */
+
+ //get a byte from the midifile
+ unsigned char getbyte();
+
+ //peek the current byte from the midifile
+ unsigned char peekbyte();
+
+ //get a set of 4 bytes from the midifile
+ unsigned int getint32();
+
+ //get a word of 2 bytes from the midifile
+ unsigned short int getint16();
+
+ //read a variable length quantity
+ unsigned int getvarint32();
+
+ //skip some bytes
+ void skipnbytes(int n);
+
+ struct {
+ double tick;//how many seconds one tick has
+
+ }data;
+
};
#endif
diff --git a/src/Seq/Sequencer.C b/src/Seq/Sequencer.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Sequencer.C - The Sequencer
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -31,70 +31,76 @@
-Sequencer::Sequencer(){
+Sequencer::Sequencer()
+{
play=0;
- for (int i=0;i<NUM_MIDI_TRACKS;i++){
- miditrack[i].track.first=NULL;
- miditrack[i].track.current=NULL;
- miditrack[i].track.size=0;
- miditrack[i].track.length=0.0;
- miditrack[i].record.first=NULL;
- miditrack[i].record.current=NULL;
- miditrack[i].record.size=0;
- miditrack[i].record.length=0.0;
-
- nextevent[i].time=0.0;
- resettime(&playtime[i]);
+ for (int i=0;i<NUM_MIDI_TRACKS;i++) {
+ miditrack[i].track.first=NULL;
+ miditrack[i].track.current=NULL;
+ miditrack[i].track.size=0;
+ miditrack[i].track.length=0.0;
+ miditrack[i].record.first=NULL;
+ miditrack[i].record.current=NULL;
+ miditrack[i].record.size=0;
+ miditrack[i].record.length=0.0;
+
+ nextevent[i].time=0.0;
+ resettime(&playtime[i]);
};
-
+
setplayspeed(0);
};
-Sequencer::~Sequencer(){
- for (int i=0;i<NUM_MIDI_TRACKS;i++){
- deletelist(&miditrack[i].track);
- deletelist(&miditrack[i].record);
+Sequencer::~Sequencer()
+{
+ for (int i=0;i<NUM_MIDI_TRACKS;i++) {
+ deletelist(&miditrack[i].track);
+ deletelist(&miditrack[i].record);
};
};
-int Sequencer::importmidifile(const char *filename){
+int Sequencer::importmidifile(const char *filename)
+{
if (midifile.loadfile(filename)<0) return(-1);
- for (int i=0;i<NUM_MIDI_TRACKS;i++){
- deletelist(&miditrack[i].record);
+ for (int i=0;i<NUM_MIDI_TRACKS;i++) {
+ deletelist(&miditrack[i].record);
};
if (midifile.parsemidifile(this)<0) return(-1);
-
+
//copy the "record" track to the main track
- for (int i=0;i<NUM_MIDI_TRACKS;i++){
- deletelist(&miditrack[i].track);
- miditrack[i].track=miditrack[i].record;
- deletelistreference(&miditrack[i].record);
+ for (int i=0;i<NUM_MIDI_TRACKS;i++) {
+ deletelist(&miditrack[i].track);
+ miditrack[i].track=miditrack[i].record;
+ deletelistreference(&miditrack[i].record);
};
return(0);
};
-void Sequencer::startplay(){
+void Sequencer::startplay()
+{
if (play!=0) return;
for (int i=0;i<NUM_MIDI_TRACKS;i++) resettime(&playtime[i]);
-
- for (int i=0;i<NUM_MIDI_TRACKS;i++){
- rewindlist(&miditrack[i].track);
+
+ for (int i=0;i<NUM_MIDI_TRACKS;i++) {
+ rewindlist(&miditrack[i].track);
};
play=1;
-
+
};
-void Sequencer::stopplay(){
+void Sequencer::stopplay()
+{
if (play==0) return;
play=0;
};
// ************ Player stuff ***************
-int Sequencer::getevent(char ntrack,int *midich, int *type,int *par1, int *par2){
+int Sequencer::getevent(char ntrack,int *midich, int *type,int *par1, int *par2)
+{
*type=0;
if (play==0) return(-1);
@@ -106,9 +112,9 @@ int Sequencer::getevent(char ntrack,int *midich, int *type,int *par1, int *par2)
// printf("%g %g\n",nextevent[ntrack].time,playtime[ntrack].abs);
if (nextevent[(int)ntrack].time<playtime[(int)ntrack].abs) readevent(&miditrack[(int)ntrack].track,&nextevent[(int)ntrack].ev);
- else return(-1);
+ else return(-1);
if (nextevent[(int)ntrack].ev.type==-1) return(-1);
-// printf("********************************\n");
+// printf("********************************\n");
//sa pun aici o protectie. a.i. daca distanta dintre timpul curent si eveliment e prea mare (>1sec) sa elimin nota
@@ -119,7 +125,7 @@ int Sequencer::getevent(char ntrack,int *midich, int *type,int *par1, int *par2)
*par2=nextevent[(int)ntrack].ev.par2;
*midich=nextevent[(int)ntrack].ev.channel;
-
+
double dt=nextevent[(int)ntrack].ev.deltatime*0.0001*realplayspeed;
printf("zzzzzzzzzzzzzz[%d] %d\n",ntrack,nextevent[(int)ntrack].ev.deltatime);
nextevent[(int)ntrack].time+=dt;
@@ -130,36 +136,39 @@ int Sequencer::getevent(char ntrack,int *midich, int *type,int *par1, int *par2)
/************** Timer stuff ***************/
-void Sequencer::resettime(timestruct *t){
+void Sequencer::resettime(timestruct *t)
+{
t->abs=0.0;
t->rel=0.0;
-
+
timeval tval;
-
+
t->last=0.0;
- #ifndef OS_WINDOWS
- if (gettimeofday(&tval,NULL)==0)
- t->last=tval.tv_sec+tval.tv_usec*0.000001;
- #endif
-
+#ifndef OS_WINDOWS
+ if (gettimeofday(&tval,NULL)==0)
+ t->last=tval.tv_sec+tval.tv_usec*0.000001;
+#endif
+
};
-void Sequencer::updatecounter(timestruct *t){
+void Sequencer::updatecounter(timestruct *t)
+{
timeval tval;
double current=0.0;
- #ifndef OS_WINDOWS
- if (gettimeofday(&tval,NULL)==0)
- current=tval.tv_sec+tval.tv_usec*0.000001;
- #endif
-
+#ifndef OS_WINDOWS
+ if (gettimeofday(&tval,NULL)==0)
+ current=tval.tv_sec+tval.tv_usec*0.000001;
+#endif
+
t->rel=current - t->last;
t->abs+=t->rel;
t->last=current;
-
+
// printf("%f %f %f\n",t->last,t->abs,t->rel);
};
-void Sequencer::setplayspeed(int speed){
+void Sequencer::setplayspeed(int speed)
+{
playspeed=speed;
realplayspeed=pow(10.0,speed/128.0);
};
diff --git a/src/Seq/Sequencer.h b/src/Seq/Sequencer.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Sequencer.h - The Sequencer
Copyright (C) 2003-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,62 +28,63 @@
/**The Sequencer
* \todo restructure some of this code*/
-class Sequencer:public MIDIEvents{
- public:
- /**Constructor*/
- Sequencer();
- /**Destructor*/
- ~Sequencer();
-
- //these functions are called by the master and are ignored if the recorder/player are stopped
- void recordnote(char chan, char note, char vel);
- void recordcontroller(char chan,unsigned int type,int par);
-
- /**Gets an event \todo better description
- *
- * this is only for player
- * @return 1 if this must be called at least once more
- * 0 if there are no more notes for the current time
- * -1 if there are no notes*/
- int getevent(char ntrack, int *midich,int *type,int *par1, int *par2);
-
- /**Imports a given midifile
- * @return 0 if ok or -1 if there is a error loading file*/
- int importmidifile(const char *filename);
-
- void startplay();
- void stopplay();
-
-
- int play;
- int playspeed;//viteza de rulare (0.1x-10x), 0=1.0x, 128=10x
- void setplayspeed(int speed);
-
- private:
-
- MIDIFile midifile;
-
- /* Timer */
- struct timestruct{
- double abs;//the time from the begining of the track
- double rel;//the time difference between the last and the current event
- double last;//the time of the last event (absolute, since 1 Jan 1970)
- //these must be double, because the float's precision is too low
- //and all these represent the time in seconds
- } playtime[NUM_MIDI_TRACKS];
-
- void resettime(timestruct *t);
- void updatecounter(timestruct *t);//this updates the timer values
-
- /* Player only*/
-
- struct {
- event ev;
- double time;
- } nextevent[NUM_MIDI_TRACKS];
-
- double realplayspeed;
-
+class Sequencer:public MIDIEvents
+{
+public:
+ /**Constructor*/
+ Sequencer();
+ /**Destructor*/
+ ~Sequencer();
+
+ //these functions are called by the master and are ignored if the recorder/player are stopped
+ void recordnote(char chan, char note, char vel);
+ void recordcontroller(char chan,unsigned int type,int par);
+
+ /**Gets an event \todo better description
+ *
+ * this is only for player
+ * @return 1 if this must be called at least once more
+ * 0 if there are no more notes for the current time
+ * -1 if there are no notes*/
+ int getevent(char ntrack, int *midich,int *type,int *par1, int *par2);
+
+ /**Imports a given midifile
+ * @return 0 if ok or -1 if there is a error loading file*/
+ int importmidifile(const char *filename);
+
+ void startplay();
+ void stopplay();
+
+
+ int play;
+ int playspeed;//viteza de rulare (0.1x-10x), 0=1.0x, 128=10x
+ void setplayspeed(int speed);
+
+private:
+
+ MIDIFile midifile;
+
+ /* Timer */
+ struct timestruct {
+ double abs;//the time from the begining of the track
+ double rel;//the time difference between the last and the current event
+ double last;//the time of the last event (absolute, since 1 Jan 1970)
+ //these must be double, because the float's precision is too low
+ //and all these represent the time in seconds
+ } playtime[NUM_MIDI_TRACKS];
+
+ void resettime(timestruct *t);
+ void updatecounter(timestruct *t);//this updates the timer values
+
+ /* Player only*/
+
+ struct {
+ event ev;
+ double time;
+ } nextevent[NUM_MIDI_TRACKS];
+
+ double realplayspeed;
+
};
#endif
diff --git a/src/Synth/ADnote.C b/src/Synth/ADnote.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
ADnote.C - The "additive" synthesizer
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,7 +29,8 @@
#include "ADnote.h"
-ADnote::ADnote(ADnoteParameters *pars,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote_,bool besilent){
+ADnote::ADnote(ADnoteParameters *pars,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote_,bool besilent)
+{
ready=0;
tmpwave=new REALTYPE [SOUND_BUFFER_SIZE];
@@ -55,136 +56,152 @@ ADnote::ADnote(ADnoteParameters *pars,Controller *ctl_,REALTYPE freq,REALTYPE ve
NoteEnabled=ON;
basefreq=freq;
if (velocity>1.0) velocity=1.0;
- this->velocity=velocity;
+ this->velocity=velocity;
time=0.0;
stereo=pars->GlobalPar.PStereo;
NoteGlobalPar.Detune=getdetune(pars->GlobalPar.PDetuneType
- ,pars->GlobalPar.PCoarseDetune,pars->GlobalPar.PDetune);
+ ,pars->GlobalPar.PCoarseDetune,pars->GlobalPar.PDetune);
bandwidthDetuneMultiplier=pars->getBandwidthDetuneMultiplier();
-
+
if (pars->GlobalPar.PPanning==0) NoteGlobalPar.Panning=RND;
- else NoteGlobalPar.Panning=pars->GlobalPar.PPanning/128.0;
-
+ else NoteGlobalPar.Panning=pars->GlobalPar.PPanning/128.0;
+
NoteGlobalPar.FilterCenterPitch=pars->GlobalPar.GlobalFilter->getfreq()+//center freq
- pars->GlobalPar.PFilterVelocityScale/127.0*6.0* //velocity sensing
- (VelF(velocity,pars->GlobalPar.PFilterVelocityScaleFunction)-1);
+ pars->GlobalPar.PFilterVelocityScale/127.0*6.0* //velocity sensing
+ (VelF(velocity,pars->GlobalPar.PFilterVelocityScaleFunction)-1);
if (pars->GlobalPar.PPunchStrength!=0) {
NoteGlobalPar.Punch.Enabled=1;
- NoteGlobalPar.Punch.t=1.0;//start from 1.0 and to 0.0
- NoteGlobalPar.Punch.initialvalue=( (pow(10,1.5*pars->GlobalPar.PPunchStrength/127.0)-1.0)
- *VelF(velocity,pars->GlobalPar.PPunchVelocitySensing) );
- REALTYPE time=pow(10,3.0*pars->GlobalPar.PPunchTime/127.0)/10000.0;//0.1 .. 100 ms
- REALTYPE stretch=pow(440.0/freq,pars->GlobalPar.PPunchStretch/64.0);
- NoteGlobalPar.Punch.dt=1.0/(time*SAMPLE_RATE*stretch);
+ NoteGlobalPar.Punch.t=1.0;//start from 1.0 and to 0.0
+ NoteGlobalPar.Punch.initialvalue=( (pow(10,1.5*pars->GlobalPar.PPunchStrength/127.0)-1.0)
+ *VelF(velocity,pars->GlobalPar.PPunchVelocitySensing) );
+ REALTYPE time=pow(10,3.0*pars->GlobalPar.PPunchTime/127.0)/10000.0;//0.1 .. 100 ms
+ REALTYPE stretch=pow(440.0/freq,pars->GlobalPar.PPunchStretch/64.0);
+ NoteGlobalPar.Punch.dt=1.0/(time*SAMPLE_RATE*stretch);
} else NoteGlobalPar.Punch.Enabled=0;
- for (int nvoice=0;nvoice<NUM_VOICES;nvoice++){
- pars->VoicePar[nvoice].OscilSmp->newrandseed(rand());
- NoteVoicePar[nvoice].OscilSmp=NULL;
- NoteVoicePar[nvoice].FMSmp=NULL;
- NoteVoicePar[nvoice].VoiceOut=NULL;
-
- NoteVoicePar[nvoice].FMVoice=-1;
-
- if (pars->VoicePar[nvoice].Enabled==0) {
- NoteVoicePar[nvoice].Enabled=OFF;
- continue; //the voice is disabled
- };
+ for (int nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ pars->VoicePar[nvoice].OscilSmp->newrandseed(rand());
+ NoteVoicePar[nvoice].OscilSmp=NULL;
+ NoteVoicePar[nvoice].FMSmp=NULL;
+ NoteVoicePar[nvoice].VoiceOut=NULL;
- NoteVoicePar[nvoice].Enabled=ON;
- NoteVoicePar[nvoice].fixedfreq=pars->VoicePar[nvoice].Pfixedfreq;
- NoteVoicePar[nvoice].fixedfreqET=pars->VoicePar[nvoice].PfixedfreqET;
-
- //use the Globalpars.detunetype if the detunetype is 0
- if (pars->VoicePar[nvoice].PDetuneType!=0){
- NoteVoicePar[nvoice].Detune=getdetune(pars->VoicePar[nvoice].PDetuneType
- ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune
- NoteVoicePar[nvoice].FineDetune=getdetune(pars->VoicePar[nvoice].PDetuneType
- ,0,pars->VoicePar[nvoice].PDetune);//fine detune
- } else {
- NoteVoicePar[nvoice].Detune=getdetune(pars->GlobalPar.PDetuneType
- ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune
- NoteVoicePar[nvoice].FineDetune=getdetune(pars->GlobalPar.PDetuneType
- ,0,pars->VoicePar[nvoice].PDetune);//fine detune
- };
- if (pars->VoicePar[nvoice].PFMDetuneType!=0){
- NoteVoicePar[nvoice].FMDetune=getdetune(pars->VoicePar[nvoice].PFMDetuneType
- ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune);
- } else {
- NoteVoicePar[nvoice].FMDetune=getdetune(pars->GlobalPar.PDetuneType
- ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune);
- };
+ NoteVoicePar[nvoice].FMVoice=-1;
+
+ if (pars->VoicePar[nvoice].Enabled==0) {
+ NoteVoicePar[nvoice].Enabled=OFF;
+ continue; //the voice is disabled
+ };
+
+ NoteVoicePar[nvoice].Enabled=ON;
+ NoteVoicePar[nvoice].fixedfreq=pars->VoicePar[nvoice].Pfixedfreq;
+ NoteVoicePar[nvoice].fixedfreqET=pars->VoicePar[nvoice].PfixedfreqET;
+
+ //use the Globalpars.detunetype if the detunetype is 0
+ if (pars->VoicePar[nvoice].PDetuneType!=0) {
+ NoteVoicePar[nvoice].Detune=getdetune(pars->VoicePar[nvoice].PDetuneType
+ ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune
+ NoteVoicePar[nvoice].FineDetune=getdetune(pars->VoicePar[nvoice].PDetuneType
+ ,0,pars->VoicePar[nvoice].PDetune);//fine detune
+ } else {
+ NoteVoicePar[nvoice].Detune=getdetune(pars->GlobalPar.PDetuneType
+ ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune
+ NoteVoicePar[nvoice].FineDetune=getdetune(pars->GlobalPar.PDetuneType
+ ,0,pars->VoicePar[nvoice].PDetune);//fine detune
+ };
+ if (pars->VoicePar[nvoice].PFMDetuneType!=0) {
+ NoteVoicePar[nvoice].FMDetune=getdetune(pars->VoicePar[nvoice].PFMDetuneType
+ ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune);
+ } else {
+ NoteVoicePar[nvoice].FMDetune=getdetune(pars->GlobalPar.PDetuneType
+ ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune);
+ };
- oscposhi[nvoice]=0;oscposlo[nvoice]=0.0;
- oscposhiFM[nvoice]=0;oscposloFM[nvoice]=0.0;
-
- NoteVoicePar[nvoice].OscilSmp=new REALTYPE[OSCIL_SIZE+OSCIL_SMP_EXTRA_SAMPLES];//the extra points contains the first point
-
- //Get the voice's oscil or external's voice oscil
- int vc=nvoice;
- if (pars->VoicePar[nvoice].Pextoscil!=-1) vc=pars->VoicePar[nvoice].Pextoscil;
- if (!pars->GlobalPar.Hrandgrouping) pars->VoicePar[vc].OscilSmp->newrandseed(rand());
- oscposhi[nvoice]=pars->VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,getvoicebasefreq(nvoice),
- pars->VoicePar[nvoice].Presonance);
-
- //I store the first elments to the last position for speedups
- for (int i=0;i<OSCIL_SMP_EXTRA_SAMPLES;i++) NoteVoicePar[nvoice].OscilSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].OscilSmp[i];
-
- oscposhi[nvoice]+=(int)((pars->VoicePar[nvoice].Poscilphase-64.0)/128.0*OSCIL_SIZE+OSCIL_SIZE*4);
- oscposhi[nvoice]%=OSCIL_SIZE;
-
-
- NoteVoicePar[nvoice].FreqLfo=NULL;
- NoteVoicePar[nvoice].FreqEnvelope=NULL;
-
- NoteVoicePar[nvoice].AmpLfo=NULL;
- NoteVoicePar[nvoice].AmpEnvelope=NULL;
-
- NoteVoicePar[nvoice].VoiceFilter=NULL;
- NoteVoicePar[nvoice].FilterEnvelope=NULL;
- NoteVoicePar[nvoice].FilterLfo=NULL;
-
- NoteVoicePar[nvoice].FilterCenterPitch=pars->VoicePar[nvoice].VoiceFilter->getfreq();
+ oscposhi[nvoice]=0;
+ oscposlo[nvoice]=0.0;
+ oscposhiFM[nvoice]=0;
+ oscposloFM[nvoice]=0.0;
+
+ NoteVoicePar[nvoice].OscilSmp=new REALTYPE[OSCIL_SIZE+OSCIL_SMP_EXTRA_SAMPLES];//the extra points contains the first point
+
+ //Get the voice's oscil or external's voice oscil
+ int vc=nvoice;
+ if (pars->VoicePar[nvoice].Pextoscil!=-1) vc=pars->VoicePar[nvoice].Pextoscil;
+ if (!pars->GlobalPar.Hrandgrouping) pars->VoicePar[vc].OscilSmp->newrandseed(rand());
+ oscposhi[nvoice]=pars->VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,getvoicebasefreq(nvoice),
+ pars->VoicePar[nvoice].Presonance);
+
+ //I store the first elments to the last position for speedups
+ for (int i=0;i<OSCIL_SMP_EXTRA_SAMPLES;i++) NoteVoicePar[nvoice].OscilSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].OscilSmp[i];
+
+ oscposhi[nvoice]+=(int)((pars->VoicePar[nvoice].Poscilphase-64.0)/128.0*OSCIL_SIZE+OSCIL_SIZE*4);
+ oscposhi[nvoice]%=OSCIL_SIZE;
+
+
+ NoteVoicePar[nvoice].FreqLfo=NULL;
+ NoteVoicePar[nvoice].FreqEnvelope=NULL;
+
+ NoteVoicePar[nvoice].AmpLfo=NULL;
+ NoteVoicePar[nvoice].AmpEnvelope=NULL;
+
+ NoteVoicePar[nvoice].VoiceFilter=NULL;
+ NoteVoicePar[nvoice].FilterEnvelope=NULL;
+ NoteVoicePar[nvoice].FilterLfo=NULL;
+
+ NoteVoicePar[nvoice].FilterCenterPitch=pars->VoicePar[nvoice].VoiceFilter->getfreq();
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;
- };
- NoteVoicePar[nvoice].FMVoice=pars->VoicePar[nvoice].PFMVoice;
- NoteVoicePar[nvoice].FMFreqEnvelope=NULL;
- NoteVoicePar[nvoice].FMAmpEnvelope=NULL;
-
- //Compute the Voice's modulator volume (incl. damping)
- REALTYPE fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0-1.0);
- switch (NoteVoicePar[nvoice].FMEnabled){
- case PHASE_MOD:fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0);
- NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0;
- break;
- case FREQ_MOD:NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0;
- break;
- // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0*8.0)*fmvoldamp;//???????????
- // break;
- default:if (fmvoldamp>1.0) fmvoldamp=1.0;
- NoteVoicePar[nvoice].FMVolume=pars->VoicePar[nvoice].PFMVolume/127.0*fmvoldamp;
- };
+ 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;
+ };
+
+ NoteVoicePar[nvoice].FMVoice=pars->VoicePar[nvoice].PFMVoice;
+ NoteVoicePar[nvoice].FMFreqEnvelope=NULL;
+ NoteVoicePar[nvoice].FMAmpEnvelope=NULL;
+
+ //Compute the Voice's modulator volume (incl. damping)
+ REALTYPE fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0-1.0);
+ switch (NoteVoicePar[nvoice].FMEnabled) {
+ case PHASE_MOD:
+ fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0);
+ NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0;
+ break;
+ case FREQ_MOD:
+ NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0;
+ break;
+ // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0*8.0)*fmvoldamp;//???????????
+ // break;
+ default:
+ if (fmvoldamp>1.0) fmvoldamp=1.0;
+ NoteVoicePar[nvoice].FMVolume=pars->VoicePar[nvoice].PFMVolume/127.0*fmvoldamp;
+ };
- //Voice's modulator velocity sensing
- NoteVoicePar[nvoice].FMVolume*=VelF(velocity,partparams->VoicePar[nvoice].PFMVelocityScaleFunction);
+ //Voice's modulator velocity sensing
+ NoteVoicePar[nvoice].FMVolume*=VelF(velocity,partparams->VoicePar[nvoice].PFMVelocityScaleFunction);
- FMoldsmp[nvoice]=0.0;//this is for FM (integration)
+ FMoldsmp[nvoice]=0.0;//this is for FM (integration)
- firsttick[nvoice]=1;
- NoteVoicePar[nvoice].DelayTicks=(int)((exp(pars->VoicePar[nvoice].PDelay/127.0*log(50.0))-1.0)/SOUND_BUFFER_SIZE/10.0*SAMPLE_RATE);
- };
+ firsttick[nvoice]=1;
+ NoteVoicePar[nvoice].DelayTicks=(int)((exp(pars->VoicePar[nvoice].PDelay/127.0*log(50.0))-1.0)/SOUND_BUFFER_SIZE/10.0*SAMPLE_RATE);
+ };
initparameters();
ready=1;
@@ -195,29 +212,30 @@ ADnote::ADnote(ADnoteParameters *pars,Controller *ctl_,REALTYPE freq,REALTYPE ve
// initparameters() stuck together with some lines removed so that it
// only alter the already playing note (to perform legato). It is
// possible I left stuff that is not required for this.
-void ADnote::ADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote_, bool externcall){
+void ADnote::ADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote_, bool externcall)
+{
ADnoteParameters *pars=partparams;
//Controller *ctl_=ctl;
// Manage legato stuff
if (externcall) Legato.msg=LM_Norm;
- if (Legato.msg!=LM_CatchUp){
- Legato.lastfreq=Legato.param.freq;
- Legato.param.freq=freq;
- Legato.param.vel=velocity;
- Legato.param.portamento=portamento_;
- Legato.param.midinote=midinote_;
- if (Legato.msg==LM_Norm){
- if (Legato.silent){
- Legato.fade.m=0.0;
- Legato.msg=LM_FadeIn;
- } else {
- Legato.fade.m=1.0;
- Legato.msg=LM_FadeOut;
- return;
- }
- }
- if (Legato.msg==LM_ToNorm) Legato.msg=LM_Norm;
+ if (Legato.msg!=LM_CatchUp) {
+ Legato.lastfreq=Legato.param.freq;
+ Legato.param.freq=freq;
+ Legato.param.vel=velocity;
+ Legato.param.portamento=portamento_;
+ Legato.param.midinote=midinote_;
+ if (Legato.msg==LM_Norm) {
+ if (Legato.silent) {
+ Legato.fade.m=0.0;
+ Legato.msg=LM_FadeIn;
+ } else {
+ Legato.fade.m=1.0;
+ Legato.msg=LM_FadeOut;
+ return;
+ }
+ }
+ if (Legato.msg==LM_ToNorm) Legato.msg=LM_Norm;
}
portamento=portamento_;
@@ -225,10 +243,10 @@ void ADnote::ADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int
basefreq=freq;
if (velocity>1.0) velocity=1.0;
- this->velocity=velocity;
+ this->velocity=velocity;
NoteGlobalPar.Detune=getdetune(pars->GlobalPar.PDetuneType
- ,pars->GlobalPar.PCoarseDetune,pars->GlobalPar.PDetune);
+ ,pars->GlobalPar.PCoarseDetune,pars->GlobalPar.PDetune);
bandwidthDetuneMultiplier=pars->getBandwidthDetuneMultiplier();
if (pars->GlobalPar.PPanning==0) NoteGlobalPar.Panning=RND;
@@ -236,85 +254,88 @@ void ADnote::ADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int
NoteGlobalPar.FilterCenterPitch=pars->GlobalPar.GlobalFilter->getfreq()+//center freq
- pars->GlobalPar.PFilterVelocityScale/127.0*6.0* //velocity sensing
- (VelF(velocity,pars->GlobalPar.PFilterVelocityScaleFunction)-1);
-
-
- for (int nvoice=0;nvoice<NUM_VOICES;nvoice++){
- if (NoteVoicePar[nvoice].Enabled==OFF)
- continue; //(gf) Stay the same as first note in legato.
-
- NoteVoicePar[nvoice].fixedfreq=pars->VoicePar[nvoice].Pfixedfreq;
- NoteVoicePar[nvoice].fixedfreqET=pars->VoicePar[nvoice].PfixedfreqET;
-
- //use the Globalpars.detunetype if the detunetype is 0
- if (pars->VoicePar[nvoice].PDetuneType!=0){
- NoteVoicePar[nvoice].Detune=getdetune(pars->VoicePar[nvoice].PDetuneType
- ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune
- NoteVoicePar[nvoice].FineDetune=getdetune(pars->VoicePar[nvoice].PDetuneType
- ,0,pars->VoicePar[nvoice].PDetune);//fine detune
- } else {
- NoteVoicePar[nvoice].Detune=getdetune(pars->GlobalPar.PDetuneType
- ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune
- NoteVoicePar[nvoice].FineDetune=getdetune(pars->GlobalPar.PDetuneType
- ,0,pars->VoicePar[nvoice].PDetune);//fine detune
- };
- if (pars->VoicePar[nvoice].PFMDetuneType!=0){
- NoteVoicePar[nvoice].FMDetune=getdetune(pars->VoicePar[nvoice].PFMDetuneType
- ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune);
- } else {
- NoteVoicePar[nvoice].FMDetune=getdetune(pars->GlobalPar.PDetuneType
- ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune);
- };
-
- //Get the voice's oscil or external's voice oscil
- int vc=nvoice;
- if (pars->VoicePar[nvoice].Pextoscil!=-1) vc=pars->VoicePar[nvoice].Pextoscil;
- if (!pars->GlobalPar.Hrandgrouping) pars->VoicePar[vc].OscilSmp->newrandseed(rand());
-
- ///oscposhi[nvoice]=pars->VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,getvoicebasefreq(nvoice),pars->VoicePar[nvoice].Presonance);
- pars->VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,getvoicebasefreq(nvoice),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++) NoteVoicePar[nvoice].OscilSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].OscilSmp[i];
-
-
- NoteVoicePar[nvoice].FilterCenterPitch=pars->VoicePar[nvoice].VoiceFilter->getfreq();
- NoteVoicePar[nvoice].filterbypass=pars->VoicePar[nvoice].Pfilterbypass;
-
-
- NoteVoicePar[nvoice].FMVoice=pars->VoicePar[nvoice].PFMVoice;
-
- //Compute the Voice's modulator volume (incl. damping)
- REALTYPE fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0-1.0);
-
- switch (NoteVoicePar[nvoice].FMEnabled){
- case PHASE_MOD:fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0);
- NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0;
- break;
- case FREQ_MOD:NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0;
- break;
- // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0*8.0)*fmvoldamp;//???????????
- // break;
- default:if (fmvoldamp>1.0) fmvoldamp=1.0;
- NoteVoicePar[nvoice].FMVolume=pars->VoicePar[nvoice].PFMVolume/127.0*fmvoldamp;
- };
-
- //Voice's modulator velocity sensing
- NoteVoicePar[nvoice].FMVolume*=VelF(velocity,partparams->VoicePar[nvoice].PFMVelocityScaleFunction);
-
- NoteVoicePar[nvoice].DelayTicks=(int)((exp(pars->VoicePar[nvoice].PDelay/127.0*log(50.0))-1.0)/SOUND_BUFFER_SIZE/10.0*SAMPLE_RATE);
+ pars->GlobalPar.PFilterVelocityScale/127.0*6.0* //velocity sensing
+ (VelF(velocity,pars->GlobalPar.PFilterVelocityScaleFunction)-1);
+
+
+ for (int nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ if (NoteVoicePar[nvoice].Enabled==OFF)
+ continue; //(gf) Stay the same as first note in legato.
+
+ NoteVoicePar[nvoice].fixedfreq=pars->VoicePar[nvoice].Pfixedfreq;
+ NoteVoicePar[nvoice].fixedfreqET=pars->VoicePar[nvoice].PfixedfreqET;
+
+ //use the Globalpars.detunetype if the detunetype is 0
+ if (pars->VoicePar[nvoice].PDetuneType!=0) {
+ NoteVoicePar[nvoice].Detune=getdetune(pars->VoicePar[nvoice].PDetuneType
+ ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune
+ NoteVoicePar[nvoice].FineDetune=getdetune(pars->VoicePar[nvoice].PDetuneType
+ ,0,pars->VoicePar[nvoice].PDetune);//fine detune
+ } else {
+ NoteVoicePar[nvoice].Detune=getdetune(pars->GlobalPar.PDetuneType
+ ,pars->VoicePar[nvoice].PCoarseDetune,8192);//coarse detune
+ NoteVoicePar[nvoice].FineDetune=getdetune(pars->GlobalPar.PDetuneType
+ ,0,pars->VoicePar[nvoice].PDetune);//fine detune
+ };
+ if (pars->VoicePar[nvoice].PFMDetuneType!=0) {
+ NoteVoicePar[nvoice].FMDetune=getdetune(pars->VoicePar[nvoice].PFMDetuneType
+ ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune);
+ } else {
+ NoteVoicePar[nvoice].FMDetune=getdetune(pars->GlobalPar.PDetuneType
+ ,pars->VoicePar[nvoice].PFMCoarseDetune,pars->VoicePar[nvoice].PFMDetune);
+ };
+
+ //Get the voice's oscil or external's voice oscil
+ int vc=nvoice;
+ if (pars->VoicePar[nvoice].Pextoscil!=-1) vc=pars->VoicePar[nvoice].Pextoscil;
+ if (!pars->GlobalPar.Hrandgrouping) pars->VoicePar[vc].OscilSmp->newrandseed(rand());
+
+ ///oscposhi[nvoice]=pars->VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,getvoicebasefreq(nvoice),pars->VoicePar[nvoice].Presonance);
+ pars->VoicePar[vc].OscilSmp->get(NoteVoicePar[nvoice].OscilSmp,getvoicebasefreq(nvoice),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++) NoteVoicePar[nvoice].OscilSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].OscilSmp[i];
+
+
+ NoteVoicePar[nvoice].FilterCenterPitch=pars->VoicePar[nvoice].VoiceFilter->getfreq();
+ NoteVoicePar[nvoice].filterbypass=pars->VoicePar[nvoice].Pfilterbypass;
+
+
+ NoteVoicePar[nvoice].FMVoice=pars->VoicePar[nvoice].PFMVoice;
+
+ //Compute the Voice's modulator volume (incl. damping)
+ REALTYPE fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0-1.0);
+
+ switch (NoteVoicePar[nvoice].FMEnabled) {
+ case PHASE_MOD:
+ fmvoldamp=pow(440.0/getvoicebasefreq(nvoice),pars->VoicePar[nvoice].PFMVolumeDamp/64.0);
+ NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0;
+ break;
+ case FREQ_MOD:
+ NoteVoicePar[nvoice].FMVolume=(exp(pars->VoicePar[nvoice].PFMVolume/127.0*FM_AMP_MULTIPLIER)-1.0)*fmvoldamp*4.0;
+ break;
+ // case PITCH_MOD:NoteVoicePar[nvoice].FMVolume=(pars->VoicePar[nvoice].PFMVolume/127.0*8.0)*fmvoldamp;//???????????
+ // break;
+ default:
+ if (fmvoldamp>1.0) fmvoldamp=1.0;
+ NoteVoicePar[nvoice].FMVolume=pars->VoicePar[nvoice].PFMVolume/127.0*fmvoldamp;
+ };
+
+ //Voice's modulator velocity sensing
+ NoteVoicePar[nvoice].FMVolume*=VelF(velocity,partparams->VoicePar[nvoice].PFMVelocityScaleFunction);
+
+ NoteVoicePar[nvoice].DelayTicks=(int)((exp(pars->VoicePar[nvoice].PDelay/127.0*log(50.0))-1.0)/SOUND_BUFFER_SIZE/10.0*SAMPLE_RATE);
};
/// initparameters();
- ///////////////
- // Altered content of initparameters():
+ ///////////////
+ // Altered content of initparameters():
int nvoice,i,tmp[NUM_VOICES];
NoteGlobalPar.Volume=4.0*pow(0.1,3.0*(1.0-partparams->GlobalPar.PVolume/96.0))//-60 dB .. 0 dB
- *VelF(velocity,partparams->GlobalPar.PAmpVelocityScaleFunction);//velocity sensing
+ *VelF(velocity,partparams->GlobalPar.PAmpVelocityScaleFunction);//velocity sensing
globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout();
@@ -325,96 +346,97 @@ void ADnote::ADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int
for (i=0;i<NUM_VOICES;i++) if (NoteVoicePar[i].FMVoice>=i) NoteVoicePar[i].FMVoice=-1;
// Voice Parameter init
- for (nvoice=0;nvoice<NUM_VOICES;nvoice++){
- if (NoteVoicePar[nvoice].Enabled==0) continue;
+ for (nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ if (NoteVoicePar[nvoice].Enabled==0) continue;
- NoteVoicePar[nvoice].noisetype=partparams->VoicePar[nvoice].Type;
- /* Voice Amplitude Parameters Init */
- NoteVoicePar[nvoice].Volume=pow(0.1,3.0*(1.0-partparams->VoicePar[nvoice].PVolume/127.0)) // -60 dB .. 0 dB
- *VelF(velocity,partparams->VoicePar[nvoice].PAmpVelocityScaleFunction);//velocity
+ NoteVoicePar[nvoice].noisetype=partparams->VoicePar[nvoice].Type;
+ /* Voice Amplitude Parameters Init */
+ NoteVoicePar[nvoice].Volume=pow(0.1,3.0*(1.0-partparams->VoicePar[nvoice].PVolume/127.0)) // -60 dB .. 0 dB
+ *VelF(velocity,partparams->VoicePar[nvoice].PAmpVelocityScaleFunction);//velocity
- if (partparams->VoicePar[nvoice].PVolumeminus!=0) NoteVoicePar[nvoice].Volume=-NoteVoicePar[nvoice].Volume;
+ if (partparams->VoicePar[nvoice].PVolumeminus!=0) NoteVoicePar[nvoice].Volume=-NoteVoicePar[nvoice].Volume;
- if (partparams->VoicePar[nvoice].PPanning==0)
- NoteVoicePar[nvoice].Panning=RND;// random panning
- else NoteVoicePar[nvoice].Panning=partparams->VoicePar[nvoice].PPanning/128.0;
+ if (partparams->VoicePar[nvoice].PPanning==0)
+ NoteVoicePar[nvoice].Panning=RND;// random panning
+ else NoteVoicePar[nvoice].Panning=partparams->VoicePar[nvoice].PPanning/128.0;
- newamplitude[nvoice]=1.0;
- if ((partparams->VoicePar[nvoice].PAmpEnvelopeEnabled!=0)
- && (NoteVoicePar[nvoice].AmpEnvelope!=NULL)){
- newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpEnvelope->envout_dB();
- };
+ newamplitude[nvoice]=1.0;
+ if ((partparams->VoicePar[nvoice].PAmpEnvelopeEnabled!=0)
+ && (NoteVoicePar[nvoice].AmpEnvelope!=NULL)) {
+ newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpEnvelope->envout_dB();
+ };
- if ((partparams->VoicePar[nvoice].PAmpLfoEnabled!=0)
- && (NoteVoicePar[nvoice].AmpLfo!=NULL)){
- newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpLfo->amplfoout();
- };
+ if ((partparams->VoicePar[nvoice].PAmpLfoEnabled!=0)
+ && (NoteVoicePar[nvoice].AmpLfo!=NULL)) {
+ newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpLfo->amplfoout();
+ };
- NoteVoicePar[nvoice].FilterFreqTracking=partparams->VoicePar[nvoice].VoiceFilter->getfreqtracking(basefreq);
+ NoteVoicePar[nvoice].FilterFreqTracking=partparams->VoicePar[nvoice].VoiceFilter->getfreqtracking(basefreq);
- /* Voice Modulation Parameters Init */
- if ((NoteVoicePar[nvoice].FMEnabled!=NONE)&&(NoteVoicePar[nvoice].FMVoice<0)){
- partparams->VoicePar[nvoice].FMSmp->newrandseed(rand());
+ /* Voice Modulation Parameters Init */
+ if ((NoteVoicePar[nvoice].FMEnabled!=NONE)&&(NoteVoicePar[nvoice].FMVoice<0)) {
+ partparams->VoicePar[nvoice].FMSmp->newrandseed(rand());
- //Perform Anti-aliasing only on MORPH or RING MODULATION
+ //Perform Anti-aliasing only on MORPH or RING MODULATION
- int vc=nvoice;
- if (partparams->VoicePar[nvoice].PextFMoscil!=-1) vc=partparams->VoicePar[nvoice].PextFMoscil;
+ int vc=nvoice;
+ if (partparams->VoicePar[nvoice].PextFMoscil!=-1) vc=partparams->VoicePar[nvoice].PextFMoscil;
- REALTYPE tmp=1.0;
- if ((partparams->VoicePar[vc].FMSmp->Padaptiveharmonics!=0)||
- (NoteVoicePar[nvoice].FMEnabled==MORPH)||
- (NoteVoicePar[nvoice].FMEnabled==RING_MOD)){
- tmp=getFMvoicebasefreq(nvoice);
- };
- if (!partparams->GlobalPar.Hrandgrouping) partparams->VoicePar[vc].FMSmp->newrandseed(rand());
+ REALTYPE tmp=1.0;
+ if ((partparams->VoicePar[vc].FMSmp->Padaptiveharmonics!=0)||
+ (NoteVoicePar[nvoice].FMEnabled==MORPH)||
+ (NoteVoicePar[nvoice].FMEnabled==RING_MOD)) {
+ tmp=getFMvoicebasefreq(nvoice);
+ };
+ if (!partparams->GlobalPar.Hrandgrouping) partparams->VoicePar[vc].FMSmp->newrandseed(rand());
- ///oscposhiFM[nvoice]=(oscposhi[nvoice]+partparams->VoicePar[vc].FMSmp->get(NoteVoicePar[nvoice].FMSmp,tmp)) % OSCIL_SIZE;
- // / oscposhi[nvoice]+partparams->VoicePar[vc].FMSmp->get(NoteVoicePar[nvoice].FMSmp,tmp); //(gf) Modif of the above line.
- for (int i=0;i<OSCIL_SMP_EXTRA_SAMPLES;i++) NoteVoicePar[nvoice].FMSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].FMSmp[i];
- ///oscposhiFM[nvoice]+=(int)((partparams->VoicePar[nvoice].PFMoscilphase-64.0)/128.0*OSCIL_SIZE+OSCIL_SIZE*4);
- ///oscposhiFM[nvoice]%=OSCIL_SIZE;
- };
+ ///oscposhiFM[nvoice]=(oscposhi[nvoice]+partparams->VoicePar[vc].FMSmp->get(NoteVoicePar[nvoice].FMSmp,tmp)) % OSCIL_SIZE;
+ // / oscposhi[nvoice]+partparams->VoicePar[vc].FMSmp->get(NoteVoicePar[nvoice].FMSmp,tmp); //(gf) Modif of the above line.
+ for (int i=0;i<OSCIL_SMP_EXTRA_SAMPLES;i++) NoteVoicePar[nvoice].FMSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].FMSmp[i];
+ ///oscposhiFM[nvoice]+=(int)((partparams->VoicePar[nvoice].PFMoscilphase-64.0)/128.0*OSCIL_SIZE+OSCIL_SIZE*4);
+ ///oscposhiFM[nvoice]%=OSCIL_SIZE;
+ };
- FMnewamplitude[nvoice]=NoteVoicePar[nvoice].FMVolume*ctl->fmamp.relamp;
+ FMnewamplitude[nvoice]=NoteVoicePar[nvoice].FMVolume*ctl->fmamp.relamp;
- if ((partparams->VoicePar[nvoice].PFMAmpEnvelopeEnabled!=0)
- && (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL)){
- FMnewamplitude[nvoice]*=NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();
- };
+ if ((partparams->VoicePar[nvoice].PFMAmpEnvelopeEnabled!=0)
+ && (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL)) {
+ FMnewamplitude[nvoice]*=NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();
+ };
};
- for (nvoice=0;nvoice<NUM_VOICES;nvoice++){
- for (i=nvoice+1;i<NUM_VOICES;i++) tmp[i]=0;
- for (i=nvoice+1;i<NUM_VOICES;i++)
- if ((NoteVoicePar[i].FMVoice==nvoice)&&(tmp[i]==0)){
- tmp[i]=1;
- };
- /// if (NoteVoicePar[nvoice].VoiceOut!=NULL) for (i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=0.0;
+ for (nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ for (i=nvoice+1;i<NUM_VOICES;i++) tmp[i]=0;
+ for (i=nvoice+1;i<NUM_VOICES;i++)
+ if ((NoteVoicePar[i].FMVoice==nvoice)&&(tmp[i]==0)) {
+ tmp[i]=1;
+ };
+ /// if (NoteVoicePar[nvoice].VoiceOut!=NULL) for (i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=0.0;
};
///////////////
- // End of the ADlegatonote function.
+ // End of the ADlegatonote function.
};
/*
* Kill a voice of ADnote
*/
-void ADnote::KillVoice(int nvoice){
-
+void ADnote::KillVoice(int nvoice)
+{
+
delete []NoteVoicePar[nvoice].OscilSmp;
if (NoteVoicePar[nvoice].FreqEnvelope!=NULL) delete(NoteVoicePar[nvoice].FreqEnvelope);
NoteVoicePar[nvoice].FreqEnvelope=NULL;
-
+
if (NoteVoicePar[nvoice].FreqLfo!=NULL) delete(NoteVoicePar[nvoice].FreqLfo);
NoteVoicePar[nvoice].FreqLfo=NULL;
if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) delete (NoteVoicePar[nvoice].AmpEnvelope);
NoteVoicePar[nvoice].AmpEnvelope=NULL;
-
+
if (NoteVoicePar[nvoice].AmpLfo!=NULL) delete (NoteVoicePar[nvoice].AmpLfo);
NoteVoicePar[nvoice].AmpLfo=NULL;
@@ -429,14 +451,14 @@ void ADnote::KillVoice(int nvoice){
if (NoteVoicePar[nvoice].FMFreqEnvelope!=NULL) delete (NoteVoicePar[nvoice].FMFreqEnvelope);
NoteVoicePar[nvoice].FMFreqEnvelope=NULL;
-
+
if (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL) delete (NoteVoicePar[nvoice].FMAmpEnvelope);
NoteVoicePar[nvoice].FMAmpEnvelope=NULL;
-
+
if ((NoteVoicePar[nvoice].FMEnabled!=NONE)&&(NoteVoicePar[nvoice].FMVoice<0)) delete []NoteVoicePar[nvoice].FMSmp;
-
- if (NoteVoicePar[nvoice].VoiceOut!=NULL)
- for (int i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=0.0;//do not delete, yet: perhaps is used by another voice
+
+ if (NoteVoicePar[nvoice].VoiceOut!=NULL)
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=0.0;//do not delete, yet: perhaps is used by another voice
NoteVoicePar[nvoice].Enabled=OFF;
};
@@ -444,16 +466,17 @@ void ADnote::KillVoice(int nvoice){
/*
* Kill the note
*/
-void ADnote::KillNote(){
+void ADnote::KillNote()
+{
int nvoice;
- for (nvoice=0;nvoice<NUM_VOICES;nvoice++){
- if (NoteVoicePar[nvoice].Enabled==ON) KillVoice(nvoice);
+ for (nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ if (NoteVoicePar[nvoice].Enabled==ON) KillVoice(nvoice);
- //delete VoiceOut
- if (NoteVoicePar[nvoice].VoiceOut!=NULL) delete(NoteVoicePar[nvoice].VoiceOut);
- NoteVoicePar[nvoice].VoiceOut=NULL;
+ //delete VoiceOut
+ if (NoteVoicePar[nvoice].VoiceOut!=NULL) delete(NoteVoicePar[nvoice].VoiceOut);
+ NoteVoicePar[nvoice].VoiceOut=NULL;
};
-
+
delete (NoteGlobalPar.FreqEnvelope);
delete (NoteGlobalPar.FreqLfo);
delete (NoteGlobalPar.AmpEnvelope);
@@ -462,11 +485,12 @@ void ADnote::KillNote(){
if (stereo!=0) delete (NoteGlobalPar.GlobalFilterR);
delete (NoteGlobalPar.FilterEnvelope);
delete (NoteGlobalPar.FilterLfo);
-
+
NoteEnabled=OFF;
};
-ADnote::~ADnote(){
+ADnote::~ADnote()
+{
if (NoteEnabled==ON) KillNote();
delete [] tmpwave;
delete [] bypassl;
@@ -477,122 +501,123 @@ ADnote::~ADnote(){
/*
* Init the parameters
*/
-void ADnote::initparameters(){
+void ADnote::initparameters()
+{
int nvoice,i,tmp[NUM_VOICES];
// Global Parameters
NoteGlobalPar.FreqEnvelope=new Envelope(partparams->GlobalPar.FreqEnvelope,basefreq);
NoteGlobalPar.FreqLfo=new LFO(partparams->GlobalPar.FreqLfo,basefreq);
-
+
NoteGlobalPar.AmpEnvelope=new Envelope(partparams->GlobalPar.AmpEnvelope,basefreq);
NoteGlobalPar.AmpLfo=new LFO(partparams->GlobalPar.AmpLfo,basefreq);
NoteGlobalPar.Volume=4.0*pow(0.1,3.0*(1.0-partparams->GlobalPar.PVolume/96.0))//-60 dB .. 0 dB
- *VelF(velocity,partparams->GlobalPar.PAmpVelocityScaleFunction);//velocity sensing
+ *VelF(velocity,partparams->GlobalPar.PAmpVelocityScaleFunction);//velocity sensing
NoteGlobalPar.AmpEnvelope->envout_dB();//discard the first envelope output
globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout();
NoteGlobalPar.GlobalFilterL=new Filter(partparams->GlobalPar.GlobalFilter);
if (stereo!=0) NoteGlobalPar.GlobalFilterR=new Filter(partparams->GlobalPar.GlobalFilter);
-
+
NoteGlobalPar.FilterEnvelope=new Envelope(partparams->GlobalPar.FilterEnvelope,basefreq);
NoteGlobalPar.FilterLfo=new LFO(partparams->GlobalPar.FilterLfo,basefreq);
NoteGlobalPar.FilterQ=partparams->GlobalPar.GlobalFilter->getq();
NoteGlobalPar.FilterFreqTracking=partparams->GlobalPar.GlobalFilter->getfreqtracking(basefreq);
-
+
// Forbids the Modulation Voice to be greater or equal than voice
for (i=0;i<NUM_VOICES;i++) if (NoteVoicePar[i].FMVoice>=i) NoteVoicePar[i].FMVoice=-1;
// Voice Parameter init
- for (nvoice=0;nvoice<NUM_VOICES;nvoice++){
+ for (nvoice=0;nvoice<NUM_VOICES;nvoice++) {
if (NoteVoicePar[nvoice].Enabled==0) continue;
- NoteVoicePar[nvoice].noisetype=partparams->VoicePar[nvoice].Type;
- /* Voice Amplitude Parameters Init */
- NoteVoicePar[nvoice].Volume=pow(0.1,3.0*(1.0-partparams->VoicePar[nvoice].PVolume/127.0)) // -60 dB .. 0 dB
- *VelF(velocity,partparams->VoicePar[nvoice].PAmpVelocityScaleFunction);//velocity
-
- if (partparams->VoicePar[nvoice].PVolumeminus!=0) NoteVoicePar[nvoice].Volume=-NoteVoicePar[nvoice].Volume;
-
- if (partparams->VoicePar[nvoice].PPanning==0)
- NoteVoicePar[nvoice].Panning=RND;// random panning
- else NoteVoicePar[nvoice].Panning=partparams->VoicePar[nvoice].PPanning/128.0;
-
- newamplitude[nvoice]=1.0;
- if (partparams->VoicePar[nvoice].PAmpEnvelopeEnabled!=0) {
- NoteVoicePar[nvoice].AmpEnvelope=new Envelope(partparams->VoicePar[nvoice].AmpEnvelope,basefreq);
- NoteVoicePar[nvoice].AmpEnvelope->envout_dB();//discard the first envelope sample
- newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpEnvelope->envout_dB();
- };
-
- if (partparams->VoicePar[nvoice].PAmpLfoEnabled!=0){
- NoteVoicePar[nvoice].AmpLfo=new LFO(partparams->VoicePar[nvoice].AmpLfo,basefreq);
- newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpLfo->amplfoout();
- };
+ NoteVoicePar[nvoice].noisetype=partparams->VoicePar[nvoice].Type;
+ /* Voice Amplitude Parameters Init */
+ NoteVoicePar[nvoice].Volume=pow(0.1,3.0*(1.0-partparams->VoicePar[nvoice].PVolume/127.0)) // -60 dB .. 0 dB
+ *VelF(velocity,partparams->VoicePar[nvoice].PAmpVelocityScaleFunction);//velocity
- /* Voice Frequency Parameters Init */
- if (partparams->VoicePar[nvoice].PFreqEnvelopeEnabled!=0)
- NoteVoicePar[nvoice].FreqEnvelope=new Envelope(partparams->VoicePar[nvoice].FreqEnvelope,basefreq);
+ if (partparams->VoicePar[nvoice].PVolumeminus!=0) NoteVoicePar[nvoice].Volume=-NoteVoicePar[nvoice].Volume;
- if (partparams->VoicePar[nvoice].PFreqLfoEnabled!=0) NoteVoicePar[nvoice].FreqLfo=new LFO(partparams->VoicePar[nvoice].FreqLfo,basefreq);
+ if (partparams->VoicePar[nvoice].PPanning==0)
+ NoteVoicePar[nvoice].Panning=RND;// random panning
+ else NoteVoicePar[nvoice].Panning=partparams->VoicePar[nvoice].PPanning/128.0;
- /* Voice Filter Parameters Init */
- if (partparams->VoicePar[nvoice].PFilterEnabled!=0){
- NoteVoicePar[nvoice].VoiceFilter=new Filter(partparams->VoicePar[nvoice].VoiceFilter);
- };
+ newamplitude[nvoice]=1.0;
+ if (partparams->VoicePar[nvoice].PAmpEnvelopeEnabled!=0) {
+ NoteVoicePar[nvoice].AmpEnvelope=new Envelope(partparams->VoicePar[nvoice].AmpEnvelope,basefreq);
+ NoteVoicePar[nvoice].AmpEnvelope->envout_dB();//discard the first envelope sample
+ newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpEnvelope->envout_dB();
+ };
- if (partparams->VoicePar[nvoice].PFilterEnvelopeEnabled!=0)
- NoteVoicePar[nvoice].FilterEnvelope=new Envelope(partparams->VoicePar[nvoice].FilterEnvelope,basefreq);
-
- if (partparams->VoicePar[nvoice].PFilterLfoEnabled!=0)
- NoteVoicePar[nvoice].FilterLfo=new LFO(partparams->VoicePar[nvoice].FilterLfo,basefreq);
-
- NoteVoicePar[nvoice].FilterFreqTracking=partparams->VoicePar[nvoice].VoiceFilter->getfreqtracking(basefreq);
-
- /* Voice Modulation Parameters Init */
- if ((NoteVoicePar[nvoice].FMEnabled!=NONE)&&(NoteVoicePar[nvoice].FMVoice<0)){
- partparams->VoicePar[nvoice].FMSmp->newrandseed(rand());
- NoteVoicePar[nvoice].FMSmp=new REALTYPE[OSCIL_SIZE+OSCIL_SMP_EXTRA_SAMPLES];
-
- //Perform Anti-aliasing only on MORPH or RING MODULATION
-
- int vc=nvoice;
- if (partparams->VoicePar[nvoice].PextFMoscil!=-1) vc=partparams->VoicePar[nvoice].PextFMoscil;
-
- REALTYPE tmp=1.0;
- if ((partparams->VoicePar[vc].FMSmp->Padaptiveharmonics!=0)||
- (NoteVoicePar[nvoice].FMEnabled==MORPH)||
- (NoteVoicePar[nvoice].FMEnabled==RING_MOD)){
- tmp=getFMvoicebasefreq(nvoice);
- };
- if (!partparams->GlobalPar.Hrandgrouping) partparams->VoicePar[vc].FMSmp->newrandseed(rand());
-
- oscposhiFM[nvoice]=(oscposhi[nvoice]+partparams->VoicePar[vc].FMSmp->get(NoteVoicePar[nvoice].FMSmp,tmp)) % OSCIL_SIZE;
- for (int i=0;i<OSCIL_SMP_EXTRA_SAMPLES;i++) NoteVoicePar[nvoice].FMSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].FMSmp[i];
- oscposhiFM[nvoice]+=(int)((partparams->VoicePar[nvoice].PFMoscilphase-64.0)/128.0*OSCIL_SIZE+OSCIL_SIZE*4);
- oscposhiFM[nvoice]%=OSCIL_SIZE;
- };
+ if (partparams->VoicePar[nvoice].PAmpLfoEnabled!=0) {
+ NoteVoicePar[nvoice].AmpLfo=new LFO(partparams->VoicePar[nvoice].AmpLfo,basefreq);
+ newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpLfo->amplfoout();
+ };
- if (partparams->VoicePar[nvoice].PFMFreqEnvelopeEnabled!=0)
- NoteVoicePar[nvoice].FMFreqEnvelope=new Envelope(partparams->VoicePar[nvoice].FMFreqEnvelope,basefreq);
+ /* Voice Frequency Parameters Init */
+ if (partparams->VoicePar[nvoice].PFreqEnvelopeEnabled!=0)
+ NoteVoicePar[nvoice].FreqEnvelope=new Envelope(partparams->VoicePar[nvoice].FreqEnvelope,basefreq);
- FMnewamplitude[nvoice]=NoteVoicePar[nvoice].FMVolume*ctl->fmamp.relamp;
+ if (partparams->VoicePar[nvoice].PFreqLfoEnabled!=0) NoteVoicePar[nvoice].FreqLfo=new LFO(partparams->VoicePar[nvoice].FreqLfo,basefreq);
- if (partparams->VoicePar[nvoice].PFMAmpEnvelopeEnabled!=0){
- NoteVoicePar[nvoice].FMAmpEnvelope=new Envelope(partparams->VoicePar[nvoice].FMAmpEnvelope,basefreq);
- FMnewamplitude[nvoice]*=NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();
- };
+ /* Voice Filter Parameters Init */
+ if (partparams->VoicePar[nvoice].PFilterEnabled!=0) {
+ NoteVoicePar[nvoice].VoiceFilter=new Filter(partparams->VoicePar[nvoice].VoiceFilter);
+ };
+
+ if (partparams->VoicePar[nvoice].PFilterEnvelopeEnabled!=0)
+ NoteVoicePar[nvoice].FilterEnvelope=new Envelope(partparams->VoicePar[nvoice].FilterEnvelope,basefreq);
+
+ if (partparams->VoicePar[nvoice].PFilterLfoEnabled!=0)
+ NoteVoicePar[nvoice].FilterLfo=new LFO(partparams->VoicePar[nvoice].FilterLfo,basefreq);
+
+ NoteVoicePar[nvoice].FilterFreqTracking=partparams->VoicePar[nvoice].VoiceFilter->getfreqtracking(basefreq);
+
+ /* Voice Modulation Parameters Init */
+ if ((NoteVoicePar[nvoice].FMEnabled!=NONE)&&(NoteVoicePar[nvoice].FMVoice<0)) {
+ partparams->VoicePar[nvoice].FMSmp->newrandseed(rand());
+ NoteVoicePar[nvoice].FMSmp=new REALTYPE[OSCIL_SIZE+OSCIL_SMP_EXTRA_SAMPLES];
+
+ //Perform Anti-aliasing only on MORPH or RING MODULATION
+
+ int vc=nvoice;
+ if (partparams->VoicePar[nvoice].PextFMoscil!=-1) vc=partparams->VoicePar[nvoice].PextFMoscil;
+
+ REALTYPE tmp=1.0;
+ if ((partparams->VoicePar[vc].FMSmp->Padaptiveharmonics!=0)||
+ (NoteVoicePar[nvoice].FMEnabled==MORPH)||
+ (NoteVoicePar[nvoice].FMEnabled==RING_MOD)) {
+ tmp=getFMvoicebasefreq(nvoice);
+ };
+ if (!partparams->GlobalPar.Hrandgrouping) partparams->VoicePar[vc].FMSmp->newrandseed(rand());
+
+ oscposhiFM[nvoice]=(oscposhi[nvoice]+partparams->VoicePar[vc].FMSmp->get(NoteVoicePar[nvoice].FMSmp,tmp)) % OSCIL_SIZE;
+ for (int i=0;i<OSCIL_SMP_EXTRA_SAMPLES;i++) NoteVoicePar[nvoice].FMSmp[OSCIL_SIZE+i]=NoteVoicePar[nvoice].FMSmp[i];
+ oscposhiFM[nvoice]+=(int)((partparams->VoicePar[nvoice].PFMoscilphase-64.0)/128.0*OSCIL_SIZE+OSCIL_SIZE*4);
+ oscposhiFM[nvoice]%=OSCIL_SIZE;
+ };
+
+ if (partparams->VoicePar[nvoice].PFMFreqEnvelopeEnabled!=0)
+ NoteVoicePar[nvoice].FMFreqEnvelope=new Envelope(partparams->VoicePar[nvoice].FMFreqEnvelope,basefreq);
+
+ FMnewamplitude[nvoice]=NoteVoicePar[nvoice].FMVolume*ctl->fmamp.relamp;
+
+ if (partparams->VoicePar[nvoice].PFMAmpEnvelopeEnabled!=0) {
+ NoteVoicePar[nvoice].FMAmpEnvelope=new Envelope(partparams->VoicePar[nvoice].FMAmpEnvelope,basefreq);
+ FMnewamplitude[nvoice]*=NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();
+ };
};
- for (nvoice=0;nvoice<NUM_VOICES;nvoice++){
- for (i=nvoice+1;i<NUM_VOICES;i++) tmp[i]=0;
- for (i=nvoice+1;i<NUM_VOICES;i++)
- if ((NoteVoicePar[i].FMVoice==nvoice)&&(tmp[i]==0)){
- NoteVoicePar[nvoice].VoiceOut=new REALTYPE[SOUND_BUFFER_SIZE];
- tmp[i]=1;
- };
- if (NoteVoicePar[nvoice].VoiceOut!=NULL) for (i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=0.0;
+ for (nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ for (i=nvoice+1;i<NUM_VOICES;i++) tmp[i]=0;
+ for (i=nvoice+1;i<NUM_VOICES;i++)
+ if ((NoteVoicePar[i].FMVoice==nvoice)&&(tmp[i]==0)) {
+ NoteVoicePar[nvoice].VoiceOut=new REALTYPE[SOUND_BUFFER_SIZE];
+ tmp[i]=1;
+ };
+ if (NoteVoicePar[nvoice].VoiceOut!=NULL) for (i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=0.0;
};
};
@@ -601,7 +626,8 @@ void ADnote::initparameters(){
/*
* Computes the frequency of an oscillator
*/
-void ADnote::setfreq(int nvoice,REALTYPE freq){
+void ADnote::setfreq(int nvoice,REALTYPE freq)
+{
REALTYPE speed;
freq=fabs(freq);
speed=freq*REALTYPE(OSCIL_SIZE)/(REALTYPE) SAMPLE_RATE;
@@ -609,12 +635,13 @@ void ADnote::setfreq(int nvoice,REALTYPE freq){
F2I(speed,oscfreqhi[nvoice]);
oscfreqlo[nvoice]=speed-floor(speed);
-};
+};
/*
* Computes the frequency of an modullator oscillator
*/
-void ADnote::setfreqFM(int nvoice,REALTYPE freq){
+void ADnote::setfreqFM(int nvoice,REALTYPE freq)
+{
REALTYPE speed;
freq=fabs(freq);
speed=freq*REALTYPE(OSCIL_SIZE)/(REALTYPE) SAMPLE_RATE;
@@ -627,28 +654,30 @@ void ADnote::setfreqFM(int nvoice,REALTYPE freq){
/*
* Get Voice base frequency
*/
-REALTYPE ADnote::getvoicebasefreq(int nvoice){
+REALTYPE ADnote::getvoicebasefreq(int nvoice)
+{
REALTYPE detune=NoteVoicePar[nvoice].Detune/100.0+
- NoteVoicePar[nvoice].FineDetune/100.0*ctl->bandwidth.relbw*bandwidthDetuneMultiplier+
- NoteGlobalPar.Detune/100.0;
+ NoteVoicePar[nvoice].FineDetune/100.0*ctl->bandwidth.relbw*bandwidthDetuneMultiplier+
+ NoteGlobalPar.Detune/100.0;
if (NoteVoicePar[nvoice].fixedfreq==0) return(this->basefreq*pow(2,detune/12.0));
- else {//the fixed freq is enabled
- REALTYPE fixedfreq=440.0;
- int fixedfreqET=NoteVoicePar[nvoice].fixedfreqET;
- if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
- REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
- if (fixedfreqET<=64) fixedfreq*=pow(2.0,tmp);
- else fixedfreq*=pow(3.0,tmp);
- };
- return(fixedfreq*pow(2.0,detune/12.0));
- };
+ else {//the fixed freq is enabled
+ REALTYPE fixedfreq=440.0;
+ int fixedfreqET=NoteVoicePar[nvoice].fixedfreqET;
+ if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
+ REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
+ if (fixedfreqET<=64) fixedfreq*=pow(2.0,tmp);
+ else fixedfreq*=pow(3.0,tmp);
+ };
+ return(fixedfreq*pow(2.0,detune/12.0));
+ };
};
/*
* Get Voice's Modullator base frequency
*/
-REALTYPE ADnote::getFMvoicebasefreq(int nvoice){
+REALTYPE ADnote::getFMvoicebasefreq(int nvoice)
+{
REALTYPE detune=NoteVoicePar[nvoice].FMDetune/100.0;
return(getvoicebasefreq(nvoice)*pow(2,detune/12.0));
};
@@ -656,102 +685,103 @@ REALTYPE ADnote::getFMvoicebasefreq(int nvoice){
/*
* Computes all the parameters for each tick
*/
-void ADnote::computecurrentparameters(){
+void ADnote::computecurrentparameters()
+{
int nvoice;
REALTYPE voicefreq,voicepitch,filterpitch,filterfreq,FMfreq,FMrelativepitch,globalpitch,globalfilterpitch;
globalpitch=0.01*(NoteGlobalPar.FreqEnvelope->envout()+
- NoteGlobalPar.FreqLfo->lfoout()*ctl->modwheel.relmod);
+ NoteGlobalPar.FreqLfo->lfoout()*ctl->modwheel.relmod);
globaloldamplitude=globalnewamplitude;
globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout();
-
+
globalfilterpitch=NoteGlobalPar.FilterEnvelope->envout()+NoteGlobalPar.FilterLfo->lfoout()
+NoteGlobalPar.FilterCenterPitch;
-
+
REALTYPE tmpfilterfreq=globalfilterpitch+ctl->filtercutoff.relfreq
- +NoteGlobalPar.FilterFreqTracking;
-
+ +NoteGlobalPar.FilterFreqTracking;
+
tmpfilterfreq=NoteGlobalPar.GlobalFilterL->getrealfreq(tmpfilterfreq);
-
+
REALTYPE globalfilterq=NoteGlobalPar.FilterQ*ctl->filterq.relq;
NoteGlobalPar.GlobalFilterL->setfreq_and_q(tmpfilterfreq,globalfilterq);
if (stereo!=0) NoteGlobalPar.GlobalFilterR->setfreq_and_q(tmpfilterfreq,globalfilterq);
//compute the portamento, if it is used by this note
- REALTYPE portamentofreqrap=1.0;
- if (portamento!=0){//this voice use portamento
- portamentofreqrap=ctl->portamento.freqrap;
- if (ctl->portamento.used==0){//the portamento has finished
- portamento=0;//this note is no longer "portamented"
- };
+ REALTYPE portamentofreqrap=1.0;
+ if (portamento!=0) {//this voice use portamento
+ 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
- for (nvoice=0;nvoice<NUM_VOICES;nvoice++){
- if (NoteVoicePar[nvoice].Enabled!=ON) continue;
- NoteVoicePar[nvoice].DelayTicks-=1;
- if (NoteVoicePar[nvoice].DelayTicks>0) continue;
-
- /*******************/
- /* Voice Amplitude */
- /*******************/
+ for (nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ if (NoteVoicePar[nvoice].Enabled!=ON) continue;
+ NoteVoicePar[nvoice].DelayTicks-=1;
+ if (NoteVoicePar[nvoice].DelayTicks>0) continue;
+
+ /*******************/
+ /* Voice Amplitude */
+ /*******************/
oldamplitude[nvoice]=newamplitude[nvoice];
- newamplitude[nvoice]=1.0;
-
- if (NoteVoicePar[nvoice].AmpEnvelope!=NULL)
- newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpEnvelope->envout_dB();
-
- if (NoteVoicePar[nvoice].AmpLfo!=NULL)
- newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpLfo->amplfoout();
-
- /****************/
- /* Voice Filter */
- /****************/
- if (NoteVoicePar[nvoice].VoiceFilter!=NULL){
- filterpitch=NoteVoicePar[nvoice].FilterCenterPitch;
-
- if (NoteVoicePar[nvoice].FilterEnvelope!=NULL)
- filterpitch+=NoteVoicePar[nvoice].FilterEnvelope->envout();
-
- if (NoteVoicePar[nvoice].FilterLfo!=NULL)
- filterpitch+=NoteVoicePar[nvoice].FilterLfo->lfoout();
-
- filterfreq=filterpitch+NoteVoicePar[nvoice].FilterFreqTracking;
- filterfreq=NoteVoicePar[nvoice].VoiceFilter->getrealfreq(filterfreq);
-
- NoteVoicePar[nvoice].VoiceFilter->setfreq(filterfreq);
- };
+ newamplitude[nvoice]=1.0;
+
+ if (NoteVoicePar[nvoice].AmpEnvelope!=NULL)
+ newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpEnvelope->envout_dB();
+
+ if (NoteVoicePar[nvoice].AmpLfo!=NULL)
+ newamplitude[nvoice]*=NoteVoicePar[nvoice].AmpLfo->amplfoout();
+
+ /****************/
+ /* Voice Filter */
+ /****************/
+ if (NoteVoicePar[nvoice].VoiceFilter!=NULL) {
+ filterpitch=NoteVoicePar[nvoice].FilterCenterPitch;
+
+ if (NoteVoicePar[nvoice].FilterEnvelope!=NULL)
+ filterpitch+=NoteVoicePar[nvoice].FilterEnvelope->envout();
+
+ if (NoteVoicePar[nvoice].FilterLfo!=NULL)
+ filterpitch+=NoteVoicePar[nvoice].FilterLfo->lfoout();
+
+ filterfreq=filterpitch+NoteVoicePar[nvoice].FilterFreqTracking;
+ filterfreq=NoteVoicePar[nvoice].VoiceFilter->getrealfreq(filterfreq);
+
+ NoteVoicePar[nvoice].VoiceFilter->setfreq(filterfreq);
+ };
+
+ if (NoteVoicePar[nvoice].noisetype==0) {//compute only if the voice isn't noise
+
+ /*******************/
+ /* Voice Frequency */
+ /*******************/
+ voicepitch=0.0;
+ if (NoteVoicePar[nvoice].FreqLfo!=NULL)
+ voicepitch+=NoteVoicePar[nvoice].FreqLfo->lfoout()/100.0
+ *ctl->bandwidth.relbw;
+
+ if (NoteVoicePar[nvoice].FreqEnvelope!=NULL) voicepitch+=NoteVoicePar[nvoice].FreqEnvelope->envout()/100.0;
+ voicefreq=getvoicebasefreq(nvoice)*pow(2,(voicepitch+globalpitch)/12.0);//Hz frequency
+ voicefreq*=ctl->pitchwheel.relfreq;//change the frequency by the controller
+ setfreq(nvoice,voicefreq*portamentofreqrap);
+
+ /***************/
+ /* Modulator */
+ /***************/
+ if (NoteVoicePar[nvoice].FMEnabled!=NONE) {
+ FMrelativepitch=NoteVoicePar[nvoice].FMDetune/100.0;
+ if (NoteVoicePar[nvoice].FMFreqEnvelope!=NULL) FMrelativepitch+=NoteVoicePar[nvoice].FMFreqEnvelope->envout()/100;
+ FMfreq=pow(2.0,FMrelativepitch/12.0)*voicefreq*portamentofreqrap;
+ setfreqFM(nvoice,FMfreq);
+
+ FMoldamplitude[nvoice]=FMnewamplitude[nvoice];
+ FMnewamplitude[nvoice]=NoteVoicePar[nvoice].FMVolume*ctl->fmamp.relamp;
+ if (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL)
+ FMnewamplitude[nvoice]*=NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();
+ };
+ };
- if (NoteVoicePar[nvoice].noisetype==0){//compute only if the voice isn't noise
-
- /*******************/
- /* Voice Frequency */
- /*******************/
- voicepitch=0.0;
- if (NoteVoicePar[nvoice].FreqLfo!=NULL)
- voicepitch+=NoteVoicePar[nvoice].FreqLfo->lfoout()/100.0
- *ctl->bandwidth.relbw;
-
- if (NoteVoicePar[nvoice].FreqEnvelope!=NULL) voicepitch+=NoteVoicePar[nvoice].FreqEnvelope->envout()/100.0;
- voicefreq=getvoicebasefreq(nvoice)*pow(2,(voicepitch+globalpitch)/12.0);//Hz frequency
- voicefreq*=ctl->pitchwheel.relfreq;//change the frequency by the controller
- setfreq(nvoice,voicefreq*portamentofreqrap);
-
- /***************/
- /* Modulator */
- /***************/
- if (NoteVoicePar[nvoice].FMEnabled!=NONE){
- FMrelativepitch=NoteVoicePar[nvoice].FMDetune/100.0;
- if (NoteVoicePar[nvoice].FMFreqEnvelope!=NULL) FMrelativepitch+=NoteVoicePar[nvoice].FMFreqEnvelope->envout()/100;
- FMfreq=pow(2.0,FMrelativepitch/12.0)*voicefreq*portamentofreqrap;
- setfreqFM(nvoice,FMfreq);
-
- FMoldamplitude[nvoice]=FMnewamplitude[nvoice];
- FMnewamplitude[nvoice]=NoteVoicePar[nvoice].FMVolume*ctl->fmamp.relamp;
- if (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL)
- FMnewamplitude[nvoice]*=NoteVoicePar[nvoice].FMAmpEnvelope->envout_dB();
- };
- };
-
};
time+=(REALTYPE)SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE;
};
@@ -760,41 +790,43 @@ void ADnote::computecurrentparameters(){
/*
* Fadein in a way that removes clicks but keep sound "punchy"
*/
-inline void ADnote::fadein(REALTYPE *smps){
+inline void ADnote::fadein(REALTYPE *smps)
+{
int zerocrossings=0;
for (int i=1;i<SOUND_BUFFER_SIZE;i++)
- if ((smps[i-1]<0.0) && (smps[i]>0.0)) zerocrossings++;//this is only the possitive crossings
+ if ((smps[i-1]<0.0) && (smps[i]>0.0)) zerocrossings++;//this is only the possitive crossings
REALTYPE tmp=(SOUND_BUFFER_SIZE-1.0)/(zerocrossings+1)/3.0;
if (tmp<8.0) tmp=8.0;
int n;
- F2I(tmp,n);//how many samples is the fade-in
+ F2I(tmp,n);//how many samples is the fade-in
if (n>SOUND_BUFFER_SIZE) n=SOUND_BUFFER_SIZE;
for (int i=0;i<n;i++) {//fade-in
- REALTYPE tmp=0.5-cos((REALTYPE)i/(REALTYPE) n*PI)*0.5;
- smps[i]*=tmp;
- };
+ REALTYPE tmp=0.5-cos((REALTYPE)i/(REALTYPE) n*PI)*0.5;
+ smps[i]*=tmp;
+ };
};
/*
* Computes the Oscillator (Without Modulation) - LinearInterpolation
*/
-inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice){
+inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice)
+{
int i,poshi;
REALTYPE poslo;
poshi=oscposhi[nvoice];
poslo=oscposlo[nvoice];
REALTYPE *smps=NoteVoicePar[nvoice].OscilSmp;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- tmpwave[i]=smps[poshi]*(1.0-poslo)+smps[poshi+1]*poslo;
- poslo+=oscfreqlo[nvoice];
- if (poslo>=1.0) {
- poslo-=1.0;
- poshi++;
- };
- poshi+=oscfreqhi[nvoice];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpwave[i]=smps[poshi]*(1.0-poslo)+smps[poshi+1]*poslo;
+ poslo+=oscfreqlo[nvoice];
+ if (poslo>=1.0) {
+ poslo-=1.0;
+ poshi++;
+ };
+ poshi+=oscfreqhi[nvoice];
poshi&=OSCIL_SIZE-1;
};
oscposhi[nvoice]=poshi;
@@ -807,7 +839,7 @@ inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice){
* Computes the Oscillator (Without Modulation) - CubicInterpolation
*
The differences from the Linear are to little to deserve to be used. This is because I am using a large OSCIL_SIZE (>512)
-inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){
+inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){
int i,poshi;
REALTYPE poslo;
@@ -828,10 +860,10 @@ inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){
//tmpwave[i]=smps[poshi]*(1.0-poslo)+smps[poshi+1]*poslo;
poslo+=oscfreqlo[nvoice];
if (poslo>=1.0) {
- poslo-=1.0;
+ poslo-=1.0;
poshi++;
};
- poshi+=oscfreqhi[nvoice];
+ poshi+=oscfreqhi[nvoice];
poshi&=OSCIL_SIZE-1;
};
oscposhi[nvoice]=poshi;
@@ -841,82 +873,84 @@ inline void ADnote::ComputeVoiceOscillator_CubicInterpolation(int nvoice){
/*
* Computes the Oscillator (Morphing)
*/
-inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice){
+inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)
+{
int i;
REALTYPE amp;
- ComputeVoiceOscillator_LinearInterpolation(nvoice);
+ ComputeVoiceOscillator_LinearInterpolation(nvoice);
if (FMnewamplitude[nvoice]>1.0) FMnewamplitude[nvoice]=1.0;
if (FMoldamplitude[nvoice]>1.0) FMoldamplitude[nvoice]=1.0;
-
- if (NoteVoicePar[nvoice].FMVoice>=0){
- //if I use VoiceOut[] as modullator
- int FMVoice=NoteVoicePar[nvoice].FMVoice;
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
- ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
- tmpwave[i]=tmpwave[i]*(1.0-amp)+amp*NoteVoicePar[FMVoice].VoiceOut[i];
- };
+
+ if (NoteVoicePar[nvoice].FMVoice>=0) {
+ //if I use VoiceOut[] as modullator
+ int FMVoice=NoteVoicePar[nvoice].FMVoice;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
+ ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
+ tmpwave[i]=tmpwave[i]*(1.0-amp)+amp*NoteVoicePar[FMVoice].VoiceOut[i];
+ };
} else {
- int poshiFM=oscposhiFM[nvoice];
- REALTYPE posloFM=oscposloFM[nvoice];
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
- ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
- tmpwave[i]=tmpwave[i]*(1.0-amp)+amp
- *(NoteVoicePar[nvoice].FMSmp[poshiFM]*(1-posloFM)
- +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM);
- posloFM+=oscfreqloFM[nvoice];
- if (posloFM>=1.0) {
- posloFM-=1.0;
- poshiFM++;
- };
- poshiFM+=oscfreqhiFM[nvoice];
- poshiFM&=OSCIL_SIZE-1;
- };
- oscposhiFM[nvoice]=poshiFM;
- oscposloFM[nvoice]=posloFM;
+ int poshiFM=oscposhiFM[nvoice];
+ REALTYPE posloFM=oscposloFM[nvoice];
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
+ ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
+ tmpwave[i]=tmpwave[i]*(1.0-amp)+amp
+ *(NoteVoicePar[nvoice].FMSmp[poshiFM]*(1-posloFM)
+ +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM);
+ posloFM+=oscfreqloFM[nvoice];
+ if (posloFM>=1.0) {
+ posloFM-=1.0;
+ poshiFM++;
+ };
+ poshiFM+=oscfreqhiFM[nvoice];
+ poshiFM&=OSCIL_SIZE-1;
+ };
+ oscposhiFM[nvoice]=poshiFM;
+ oscposloFM[nvoice]=posloFM;
};
};
/*
* Computes the Oscillator (Ring Modulation)
*/
-inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice){
+inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)
+{
int i;
REALTYPE amp;
- ComputeVoiceOscillator_LinearInterpolation(nvoice);
+ ComputeVoiceOscillator_LinearInterpolation(nvoice);
if (FMnewamplitude[nvoice]>1.0) FMnewamplitude[nvoice]=1.0;
if (FMoldamplitude[nvoice]>1.0) FMoldamplitude[nvoice]=1.0;
- if (NoteVoicePar[nvoice].FMVoice>=0){
- // if I use VoiceOut[] as modullator
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
- ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
- int FMVoice=NoteVoicePar[nvoice].FMVoice;
- for (i=0;i<SOUND_BUFFER_SIZE;i++)
- tmpwave[i]*=(1.0-amp)+amp*NoteVoicePar[FMVoice].VoiceOut[i];
- };
+ if (NoteVoicePar[nvoice].FMVoice>=0) {
+ // if I use VoiceOut[] as modullator
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
+ ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
+ int FMVoice=NoteVoicePar[nvoice].FMVoice;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++)
+ tmpwave[i]*=(1.0-amp)+amp*NoteVoicePar[FMVoice].VoiceOut[i];
+ };
} else {
- int poshiFM=oscposhiFM[nvoice];
- REALTYPE posloFM=oscposloFM[nvoice];
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
- ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
- tmpwave[i]*=( NoteVoicePar[nvoice].FMSmp[poshiFM]*(1.0-posloFM)
- +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM)*amp
- +(1.0-amp);
- posloFM+=oscfreqloFM[nvoice];
- if (posloFM>=1.0) {
- posloFM-=1.0;
- poshiFM++;
- };
- poshiFM+=oscfreqhiFM[nvoice];
- poshiFM&=OSCIL_SIZE-1;
- };
- oscposhiFM[nvoice]=poshiFM;
- oscposloFM[nvoice]=posloFM;
+ int poshiFM=oscposhiFM[nvoice];
+ REALTYPE posloFM=oscposloFM[nvoice];
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ amp=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
+ ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
+ tmpwave[i]*=( NoteVoicePar[nvoice].FMSmp[poshiFM]*(1.0-posloFM)
+ +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM)*amp
+ +(1.0-amp);
+ posloFM+=oscfreqloFM[nvoice];
+ if (posloFM>=1.0) {
+ posloFM-=1.0;
+ poshiFM++;
+ };
+ poshiFM+=oscfreqhiFM[nvoice];
+ poshiFM&=OSCIL_SIZE-1;
+ };
+ oscposhiFM[nvoice]=poshiFM;
+ oscposloFM[nvoice]=posloFM;
};
};
@@ -925,346 +959,362 @@ inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice){
/*
* Computes the Oscillator (Phase Modulation or Frequency Modulation)
*/
-inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,int FMmode){
+inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,int FMmode)
+{
int carposhi;
int i,FMmodfreqhi;
REALTYPE FMmodfreqlo,carposlo;
-
- if (NoteVoicePar[nvoice].FMVoice>=0){
- //if I use VoiceOut[] as modulator
- for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]=NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut[i];
+
+ if (NoteVoicePar[nvoice].FMVoice>=0) {
+ //if I use VoiceOut[] as modulator
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]=NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut[i];
} else {
- //Compute the modulator and store it in tmpwave[]
- int poshiFM=oscposhiFM[nvoice];
- REALTYPE posloFM=oscposloFM[nvoice];
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- tmpwave[i]=(NoteVoicePar[nvoice].FMSmp[poshiFM]*(1.0-posloFM)
- +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM);
- posloFM+=oscfreqloFM[nvoice];
- if (posloFM>=1.0) {
- posloFM=fmod(posloFM,1.0);
- poshiFM++;
- };
- poshiFM+=oscfreqhiFM[nvoice];
- poshiFM&=OSCIL_SIZE-1;
- };
- oscposhiFM[nvoice]=poshiFM;
- oscposloFM[nvoice]=posloFM;
+ //Compute the modulator and store it in tmpwave[]
+ int poshiFM=oscposhiFM[nvoice];
+ REALTYPE posloFM=oscposloFM[nvoice];
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpwave[i]=(NoteVoicePar[nvoice].FMSmp[poshiFM]*(1.0-posloFM)
+ +NoteVoicePar[nvoice].FMSmp[poshiFM+1]*posloFM);
+ posloFM+=oscfreqloFM[nvoice];
+ if (posloFM>=1.0) {
+ posloFM=fmod(posloFM,1.0);
+ poshiFM++;
+ };
+ poshiFM+=oscfreqhiFM[nvoice];
+ poshiFM&=OSCIL_SIZE-1;
+ };
+ oscposhiFM[nvoice]=poshiFM;
+ oscposloFM[nvoice]=posloFM;
};
// Amplitude interpolation
- if (ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],FMnewamplitude[nvoice])){
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- tmpwave[i]*=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
- ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
- };
- } else for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=FMnewamplitude[nvoice];
-
-
- //normalize makes all sample-rates, oscil_sizes toproduce same sound
- if (FMmode!=0){//Frequency modulation
- REALTYPE normalize=OSCIL_SIZE/262144.0*44100.0/(REALTYPE)SAMPLE_RATE;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- FMoldsmp[nvoice]=fmod(FMoldsmp[nvoice]+tmpwave[i]*normalize,OSCIL_SIZE);
- tmpwave[i]=FMoldsmp[nvoice];
- };
- } else {//Phase modulation
- REALTYPE normalize=OSCIL_SIZE/262144.0;
- for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=normalize;
+ if (ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],FMnewamplitude[nvoice])) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ tmpwave[i]*=INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice]
+ ,FMnewamplitude[nvoice],i,SOUND_BUFFER_SIZE);
+ };
+ } else for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=FMnewamplitude[nvoice];
+
+
+ //normalize makes all sample-rates, oscil_sizes toproduce same sound
+ if (FMmode!=0) {//Frequency modulation
+ REALTYPE normalize=OSCIL_SIZE/262144.0*44100.0/(REALTYPE)SAMPLE_RATE;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ FMoldsmp[nvoice]=fmod(FMoldsmp[nvoice]+tmpwave[i]*normalize,OSCIL_SIZE);
+ tmpwave[i]=FMoldsmp[nvoice];
+ };
+ } else {//Phase modulation
+ REALTYPE normalize=OSCIL_SIZE/262144.0;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=normalize;
};
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- F2I(tmpwave[i],FMmodfreqhi);
- FMmodfreqlo=fmod(tmpwave[i]+0.0000000001,1.0);
- if (FMmodfreqhi<0) FMmodfreqlo++;
-
- //carrier
- carposhi=oscposhi[nvoice]+FMmodfreqhi;
- carposlo=oscposlo[nvoice]+FMmodfreqlo;
-
- if (carposlo>=1.0) {
- carposhi++;
- carposlo=fmod(carposlo,1.0);
- };
- carposhi&=(OSCIL_SIZE-1);
-
- tmpwave[i]=NoteVoicePar[nvoice].OscilSmp[carposhi]*(1.0-carposlo)
- +NoteVoicePar[nvoice].OscilSmp[carposhi+1]*carposlo;
-
- oscposlo[nvoice]+=oscfreqlo[nvoice];
- if (oscposlo[nvoice]>=1.0) {
- oscposlo[nvoice]=fmod(oscposlo[nvoice],1.0);
- oscposhi[nvoice]++;
- };
-
- oscposhi[nvoice]+=oscfreqhi[nvoice];
- oscposhi[nvoice]&=OSCIL_SIZE-1;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ F2I(tmpwave[i],FMmodfreqhi);
+ FMmodfreqlo=fmod(tmpwave[i]+0.0000000001,1.0);
+ if (FMmodfreqhi<0) FMmodfreqlo++;
+
+ //carrier
+ carposhi=oscposhi[nvoice]+FMmodfreqhi;
+ carposlo=oscposlo[nvoice]+FMmodfreqlo;
+
+ if (carposlo>=1.0) {
+ carposhi++;
+ carposlo=fmod(carposlo,1.0);
+ };
+ carposhi&=(OSCIL_SIZE-1);
+
+ tmpwave[i]=NoteVoicePar[nvoice].OscilSmp[carposhi]*(1.0-carposlo)
+ +NoteVoicePar[nvoice].OscilSmp[carposhi+1]*carposlo;
+
+ oscposlo[nvoice]+=oscfreqlo[nvoice];
+ if (oscposlo[nvoice]>=1.0) {
+ oscposlo[nvoice]=fmod(oscposlo[nvoice],1.0);
+ oscposhi[nvoice]++;
+ };
+
+ oscposhi[nvoice]+=oscfreqhi[nvoice];
+ oscposhi[nvoice]&=OSCIL_SIZE-1;
};
};
/*Calculeaza Oscilatorul cu PITCH MODULATION*/
-inline void ADnote::ComputeVoiceOscillatorPitchModulation(int nvoice){
+inline void ADnote::ComputeVoiceOscillatorPitchModulation(int nvoice)
+{
//TODO
};
/*
- * Computes the Noise
+ * Computes the Noise
*/
-inline void ADnote::ComputeVoiceNoise(int nvoice){
+inline void ADnote::ComputeVoiceNoise(int nvoice)
+{
for (int i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]=RND*2.0-1.0;
};
/*
- * Compute the ADnote samples
+ * Compute the ADnote samples
* Returns 0 if the note is finished
*/
-int ADnote::noteout(REALTYPE *outl,REALTYPE *outr){
- int i,nvoice;
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- outl[i]=denormalkillbuf[i];
- outr[i]=denormalkillbuf[i];
- };
-
- if (NoteEnabled==OFF) return(0);
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- bypassl[i]=0.0;
- bypassr[i]=0.0;
- };
-
- computecurrentparameters();
-
- for (nvoice=0;nvoice<NUM_VOICES;nvoice++){
- if ((NoteVoicePar[nvoice].Enabled!=ON) || (NoteVoicePar[nvoice].DelayTicks>0)) 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);
-
- };
- } else ComputeVoiceNoise(nvoice);
- // Voice Processing
-
- // Amplitude
- if (ABOVE_AMPLITUDE_THRESHOLD(oldamplitude[nvoice],newamplitude[nvoice])){
- int rest=SOUND_BUFFER_SIZE;
- //test if the amplitude if raising and the difference is high
- if ((newamplitude[nvoice]>oldamplitude[nvoice])&&((newamplitude[nvoice]-oldamplitude[nvoice])>0.25)){
- rest=10;
- if (rest>SOUND_BUFFER_SIZE) rest=SOUND_BUFFER_SIZE;
- for (int i=0;i<SOUND_BUFFER_SIZE-rest;i++) tmpwave[i]*=oldamplitude[nvoice];
- };
- // Amplitude interpolation
- for (i=0;i<rest;i++){
- tmpwave[i+(SOUND_BUFFER_SIZE-rest)]*=INTERPOLATE_AMPLITUDE(oldamplitude[nvoice]
- ,newamplitude[nvoice],i,rest);
- };
- } else for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=newamplitude[nvoice];
-
- // Fade in
- if (firsttick[nvoice]!=0){
- fadein(&tmpwave[0]);
- firsttick[nvoice]=0;
- };
-
-
- // Filter
- if (NoteVoicePar[nvoice].VoiceFilter!=NULL) NoteVoicePar[nvoice].VoiceFilter->filterout(&tmpwave[0]);
-
- //check if the amplitude envelope is finished, if yes, the voice will be fadeout
- if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) {
- if (NoteVoicePar[nvoice].AmpEnvelope->finished()!=0)
- for (i=0;i<SOUND_BUFFER_SIZE;i++)
- tmpwave[i]*=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
- //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)
- if (NoteVoicePar[nvoice].VoiceOut!=NULL)
- for (i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=tmpwave[i];
-
-
- // Add the voice that do not bypass the filter to out
- if (NoteVoicePar[nvoice].filterbypass==0){//no bypass
- if (stereo==0) for (i=0;i<SOUND_BUFFER_SIZE;i++) outl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume;//mono
- else for (i=0;i<SOUND_BUFFER_SIZE;i++) {//stereo
- outl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*NoteVoicePar[nvoice].Panning*2.0;
- outr[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*(1.0-NoteVoicePar[nvoice].Panning)*2.0;
- };
- } else {//bypass the filter
- if (stereo==0) for (i=0;i<SOUND_BUFFER_SIZE;i++) bypassl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume;//mono
- else for (i=0;i<SOUND_BUFFER_SIZE;i++) {//stereo
- bypassl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*NoteVoicePar[nvoice].Panning*2.0;
- bypassr[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*(1.0-NoteVoicePar[nvoice].Panning)*2.0;
- };
- };
- // 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);
- };
- };
-
-
- //Processing Global parameters
- NoteGlobalPar.GlobalFilterL->filterout(&outl[0]);
-
- if (stereo==0) {
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {//set the right channel=left channel
- outr[i]=outl[i];
- bypassr[i]=bypassl[i];
- }
- } else NoteGlobalPar.GlobalFilterR->filterout(&outr[0]);
-
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- outl[i]+=bypassl[i];
- outr[i]+=bypassr[i];
- };
-
- if (ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude,globalnewamplitude)){
- // Amplitude Interpolation
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(globaloldamplitude
- ,globalnewamplitude,i,SOUND_BUFFER_SIZE);
- outl[i]*=tmpvol*NoteGlobalPar.Panning;
- outr[i]*=tmpvol*(1.0-NoteGlobalPar.Panning);
+int ADnote::noteout(REALTYPE *outl,REALTYPE *outr)
+{
+ int i,nvoice;
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]=denormalkillbuf[i];
+ outr[i]=denormalkillbuf[i];
+ };
+
+ if (NoteEnabled==OFF) return(0);
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ bypassl[i]=0.0;
+ bypassr[i]=0.0;
+ };
+
+ computecurrentparameters();
+
+ for (nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ if ((NoteVoicePar[nvoice].Enabled!=ON) || (NoteVoicePar[nvoice].DelayTicks>0)) 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);
+
+ };
+ } else ComputeVoiceNoise(nvoice);
+ // Voice Processing
+
+ // Amplitude
+ if (ABOVE_AMPLITUDE_THRESHOLD(oldamplitude[nvoice],newamplitude[nvoice])) {
+ int rest=SOUND_BUFFER_SIZE;
+ //test if the amplitude if raising and the difference is high
+ if ((newamplitude[nvoice]>oldamplitude[nvoice])&&((newamplitude[nvoice]-oldamplitude[nvoice])>0.25)) {
+ rest=10;
+ if (rest>SOUND_BUFFER_SIZE) rest=SOUND_BUFFER_SIZE;
+ for (int i=0;i<SOUND_BUFFER_SIZE-rest;i++) tmpwave[i]*=oldamplitude[nvoice];
+ };
+ // Amplitude interpolation
+ for (i=0;i<rest;i++) {
+ tmpwave[i+(SOUND_BUFFER_SIZE-rest)]*=INTERPOLATE_AMPLITUDE(oldamplitude[nvoice]
+ ,newamplitude[nvoice],i,rest);
+ };
+ } else for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpwave[i]*=newamplitude[nvoice];
+
+ // Fade in
+ if (firsttick[nvoice]!=0) {
+ fadein(&tmpwave[0]);
+ firsttick[nvoice]=0;
};
- } else {
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- outl[i]*=globalnewamplitude*NoteGlobalPar.Panning;
- outr[i]*=globalnewamplitude*(1.0-NoteGlobalPar.Panning);
- };
- };
- //Apply the punch
- if (NoteGlobalPar.Punch.Enabled!=0){
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- REALTYPE punchamp=NoteGlobalPar.Punch.initialvalue*NoteGlobalPar.Punch.t+1.0;
- outl[i]*=punchamp;
- outr[i]*=punchamp;
- NoteGlobalPar.Punch.t-=NoteGlobalPar.Punch.dt;
- if (NoteGlobalPar.Punch.t<0.0) {
- NoteGlobalPar.Punch.Enabled=0;
- break;
- };
- };
- };
+ // Filter
+ if (NoteVoicePar[nvoice].VoiceFilter!=NULL) NoteVoicePar[nvoice].VoiceFilter->filterout(&tmpwave[0]);
- // Apply legato-specific sound signal modifications
- if (Legato.silent){ // Silencer
- if (Legato.msg!=LM_FadeIn){
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- outl[i]=0.0;
- outr[i]=0.0;
- }
- }
- }
- switch (Legato.msg){
- case LM_CatchUp : // Continue the catch-up...
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){//Yea, could be done without the loop...
- Legato.decounter--;
- if (Legato.decounter<1){
- // Catching-up done, we can finally set
- // the note to the actual parameters.
- Legato.decounter=-10;
- Legato.msg=LM_ToNorm;
- ADlegatonote(Legato.param.freq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
- break;
- }
- }
- break;
- case LM_FadeIn : // Fade-in
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- Legato.silent=false;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- Legato.decounter--;
- if (Legato.decounter<1){
- Legato.decounter=-10;
- Legato.msg=LM_Norm;
- break;
- }
- Legato.fade.m+=Legato.fade.step;
- outl[i]*=Legato.fade.m;
- outr[i]*=Legato.fade.m;
+ //check if the amplitude envelope is finished, if yes, the voice will be fadeout
+ if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) {
+ if (NoteVoicePar[nvoice].AmpEnvelope->finished()!=0)
+ for (i=0;i<SOUND_BUFFER_SIZE;i++)
+ tmpwave[i]*=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
+ //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)
+ if (NoteVoicePar[nvoice].VoiceOut!=NULL)
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) NoteVoicePar[nvoice].VoiceOut[i]=tmpwave[i];
+
+
+ // Add the voice that do not bypass the filter to out
+ if (NoteVoicePar[nvoice].filterbypass==0) {//no bypass
+ if (stereo==0) for (i=0;i<SOUND_BUFFER_SIZE;i++) outl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume;//mono
+ else for (i=0;i<SOUND_BUFFER_SIZE;i++) {//stereo
+ outl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*NoteVoicePar[nvoice].Panning*2.0;
+ outr[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*(1.0-NoteVoicePar[nvoice].Panning)*2.0;
+ };
+ } else {//bypass the filter
+ if (stereo==0) for (i=0;i<SOUND_BUFFER_SIZE;i++) bypassl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume;//mono
+ else for (i=0;i<SOUND_BUFFER_SIZE;i++) {//stereo
+ bypassl[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*NoteVoicePar[nvoice].Panning*2.0;
+ bypassr[i]+=tmpwave[i]*NoteVoicePar[nvoice].Volume*(1.0-NoteVoicePar[nvoice].Panning)*2.0;
+ };
+ };
+ // 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);
+ };
+ };
+
+
+ //Processing Global parameters
+ NoteGlobalPar.GlobalFilterL->filterout(&outl[0]);
+
+ if (stereo==0) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {//set the right channel=left channel
+ outr[i]=outl[i];
+ bypassr[i]=bypassl[i];
+ }
+ } else NoteGlobalPar.GlobalFilterR->filterout(&outr[0]);
+
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]+=bypassl[i];
+ outr[i]+=bypassr[i];
+ };
+
+ if (ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude,globalnewamplitude)) {
+ // Amplitude Interpolation
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(globaloldamplitude
+ ,globalnewamplitude,i,SOUND_BUFFER_SIZE);
+ outl[i]*=tmpvol*NoteGlobalPar.Panning;
+ outr[i]*=tmpvol*(1.0-NoteGlobalPar.Panning);
+ };
+ } else {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]*=globalnewamplitude*NoteGlobalPar.Panning;
+ outr[i]*=globalnewamplitude*(1.0-NoteGlobalPar.Panning);
+ };
+ };
+
+//Apply the punch
+ if (NoteGlobalPar.Punch.Enabled!=0) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE punchamp=NoteGlobalPar.Punch.initialvalue*NoteGlobalPar.Punch.t+1.0;
+ outl[i]*=punchamp;
+ outr[i]*=punchamp;
+ NoteGlobalPar.Punch.t-=NoteGlobalPar.Punch.dt;
+ if (NoteGlobalPar.Punch.t<0.0) {
+ NoteGlobalPar.Punch.Enabled=0;
+ break;
+ };
+ };
+ };
+
+
+ // Apply legato-specific sound signal modifications
+ if (Legato.silent) { // Silencer
+ if (Legato.msg!=LM_FadeIn) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]=0.0;
+ outr[i]=0.0;
+ }
+ }
}
- break;
- case LM_FadeOut : // Fade-out, then set the catch-up
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- Legato.decounter--;
- if (Legato.decounter<1){
- for (int j=i;j<SOUND_BUFFER_SIZE;j++){
- outl[j]=0.0;
- outr[j]=0.0;
- }
- Legato.decounter=-10;
- Legato.silent=true;
- // Fading-out done, now set the catch-up :
- Legato.decounter=Legato.fade.length;
- Legato.msg=LM_CatchUp;
- REALTYPE catchupfreq=Legato.param.freq*(Legato.param.freq/Legato.lastfreq);//This freq should make this now silent note to catch-up (or should I say resync ?) with the heard note for the same length it stayed at the previous freq during the fadeout.
- ADlegatonote(catchupfreq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
- break;
- }
- Legato.fade.m-=Legato.fade.step;
- outl[i]*=Legato.fade.m;
- outr[i]*=Legato.fade.m;
+ switch (Legato.msg) {
+ case LM_CatchUp : // Continue the catch-up...
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {//Yea, could be done without the loop...
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ // Catching-up done, we can finally set
+ // the note to the actual parameters.
+ Legato.decounter=-10;
+ Legato.msg=LM_ToNorm;
+ ADlegatonote(Legato.param.freq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
+ break;
+ }
+ }
+ break;
+ case LM_FadeIn : // Fade-in
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ Legato.silent=false;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ Legato.decounter=-10;
+ Legato.msg=LM_Norm;
+ break;
+ }
+ Legato.fade.m+=Legato.fade.step;
+ outl[i]*=Legato.fade.m;
+ outr[i]*=Legato.fade.m;
+ }
+ break;
+ case LM_FadeOut : // Fade-out, then set the catch-up
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ for (int j=i;j<SOUND_BUFFER_SIZE;j++) {
+ outl[j]=0.0;
+ outr[j]=0.0;
+ }
+ Legato.decounter=-10;
+ Legato.silent=true;
+ // Fading-out done, now set the catch-up :
+ Legato.decounter=Legato.fade.length;
+ Legato.msg=LM_CatchUp;
+ REALTYPE catchupfreq=Legato.param.freq*(Legato.param.freq/Legato.lastfreq);//This freq should make this now silent note to catch-up (or should I say resync ?) with the heard note for the same length it stayed at the previous freq during the fadeout.
+ ADlegatonote(catchupfreq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
+ break;
+ }
+ Legato.fade.m-=Legato.fade.step;
+ outl[i]*=Legato.fade.m;
+ outr[i]*=Legato.fade.m;
+ }
+ break;
+ default :
+ break;
}
- break;
- default : break;
- }
-
-
- // Check if the global amplitude is finished.
- // If it does, disable the note
- if (NoteGlobalPar.AmpEnvelope->finished()!=0) {
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out
- REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
- outl[i]*=tmp;
- outr[i]*=tmp;
- };
- KillNote();
- };
- return(1);
+
+
+// Check if the global amplitude is finished.
+// If it does, disable the note
+ if (NoteGlobalPar.AmpEnvelope->finished()!=0) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out
+ REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
+ outl[i]*=tmp;
+ outr[i]*=tmp;
+ };
+ KillNote();
+ };
+ return(1);
};
/*
* Relase the key (NoteOff)
*/
-void ADnote::relasekey(){
-int nvoice;
- for (nvoice=0;nvoice<NUM_VOICES;nvoice++){
- if (NoteVoicePar[nvoice].Enabled==0) continue;
- if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) NoteVoicePar[nvoice].AmpEnvelope->relasekey();
- if (NoteVoicePar[nvoice].FreqEnvelope!=NULL) NoteVoicePar[nvoice].FreqEnvelope->relasekey();
- if (NoteVoicePar[nvoice].FilterEnvelope!=NULL) NoteVoicePar[nvoice].FilterEnvelope->relasekey();
- if (NoteVoicePar[nvoice].FMFreqEnvelope!=NULL) NoteVoicePar[nvoice].FMFreqEnvelope->relasekey();
- if (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL) NoteVoicePar[nvoice].FMAmpEnvelope->relasekey();
- };
- NoteGlobalPar.FreqEnvelope->relasekey();
- NoteGlobalPar.FilterEnvelope->relasekey();
- NoteGlobalPar.AmpEnvelope->relasekey();
+void ADnote::relasekey()
+{
+ int nvoice;
+ for (nvoice=0;nvoice<NUM_VOICES;nvoice++) {
+ if (NoteVoicePar[nvoice].Enabled==0) continue;
+ if (NoteVoicePar[nvoice].AmpEnvelope!=NULL) NoteVoicePar[nvoice].AmpEnvelope->relasekey();
+ if (NoteVoicePar[nvoice].FreqEnvelope!=NULL) NoteVoicePar[nvoice].FreqEnvelope->relasekey();
+ if (NoteVoicePar[nvoice].FilterEnvelope!=NULL) NoteVoicePar[nvoice].FilterEnvelope->relasekey();
+ if (NoteVoicePar[nvoice].FMFreqEnvelope!=NULL) NoteVoicePar[nvoice].FMFreqEnvelope->relasekey();
+ if (NoteVoicePar[nvoice].FMAmpEnvelope!=NULL) NoteVoicePar[nvoice].FMAmpEnvelope->relasekey();
+ };
+ NoteGlobalPar.FreqEnvelope->relasekey();
+ NoteGlobalPar.FilterEnvelope->relasekey();
+ NoteGlobalPar.AmpEnvelope->relasekey();
};
/*
* Check if the note is finished
*/
-int ADnote::finished(){
+int ADnote::finished()
+{
if (NoteEnabled==ON) return(0);
- else return(1);
+ else return(1);
};
diff --git a/src/Synth/ADnote.h b/src/Synth/ADnote.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
ADnote.h - The "additive" synthesizer
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -38,14 +38,15 @@
#define OSCIL_SMP_EXTRA_SAMPLES 5
/**The "additive" synthesizer*/
-class ADnote{ //ADDitive note
- public:
+class ADnote //ADDitive note
+{
+public:
ADnote(ADnoteParameters *pars,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote_,bool besilent);//(gf)Added the besilent parameter to tell it to start silent (if true).
~ADnote();
void ADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote_, bool externcall);
- int noteout(REALTYPE *outl,REALTYPE *outr);
+ int noteout(REALTYPE *outl,REALTYPE *outr);
void relasekey();
int finished();
@@ -54,30 +55,30 @@ class ADnote{ //ADDitive note
or other value if the parameters has been computed and if it is ready to output*/
char ready;
- private:
-
- void setfreq(int nvoice,REALTYPE freq);
- void setfreqFM(int nvoice,REALTYPE freq);
- void computecurrentparameters();
- void initparameters();
- void KillVoice(int nvoice);
- void KillNote();
- inline REALTYPE getvoicebasefreq(int nvoice);
- inline REALTYPE getFMvoicebasefreq(int nvoice);
- inline void ComputeVoiceOscillator_LinearInterpolation(int nvoice);
- inline void ComputeVoiceOscillator_CubicInterpolation(int nvoice);
- inline void ComputeVoiceOscillatorMorph(int nvoice);
- inline void ComputeVoiceOscillatorRingModulation(int nvoice);
- inline void ComputeVoiceOscillatorFrequencyModulation(int nvoice,int FMmode);//FMmode=0 for phase modulation, 1 for Frequency modulation
- // inline void ComputeVoiceOscillatorFrequencyModulation(int nvoice);
- inline void ComputeVoiceOscillatorPitchModulation(int nvoice);
-
- inline void ComputeVoiceNoise(int nvoice);
-
- inline void fadein(REALTYPE *smps);
-
-
- //GLOBALS
+private:
+
+ void setfreq(int nvoice,REALTYPE freq);
+ void setfreqFM(int nvoice,REALTYPE freq);
+ void computecurrentparameters();
+ void initparameters();
+ void KillVoice(int nvoice);
+ void KillNote();
+ inline REALTYPE getvoicebasefreq(int nvoice);
+ inline REALTYPE getFMvoicebasefreq(int nvoice);
+ inline void ComputeVoiceOscillator_LinearInterpolation(int nvoice);
+ inline void ComputeVoiceOscillator_CubicInterpolation(int nvoice);
+ inline void ComputeVoiceOscillatorMorph(int nvoice);
+ inline void ComputeVoiceOscillatorRingModulation(int nvoice);
+ inline void ComputeVoiceOscillatorFrequencyModulation(int nvoice,int FMmode);//FMmode=0 for phase modulation, 1 for Frequency modulation
+ // inline void ComputeVoiceOscillatorFrequencyModulation(int nvoice);
+ inline void ComputeVoiceOscillatorPitchModulation(int nvoice);
+
+ inline void ComputeVoiceNoise(int nvoice);
+
+ inline void fadein(REALTYPE *smps);
+
+
+ //GLOBALS
ADnoteParameters *partparams;
unsigned char stereo;//if the note is stereo (allows note Panning)
int midinote;
@@ -86,188 +87,188 @@ class ADnote{ //ADDitive note
ONOFFTYPE NoteEnabled;
Controller *ctl;
- /*****************************************************************/
- /* GLOBAL PARAMETERS */
- /*****************************************************************/
-
- struct ADnoteGlobal{
- /******************************************
- * FREQUENCY GLOBAL PARAMETERS *
- ******************************************/
- REALTYPE Detune;//cents
-
- Envelope *FreqEnvelope;
- LFO *FreqLfo;
-
- /********************************************
- * AMPLITUDE GLOBAL PARAMETERS *
- ********************************************/
- REALTYPE Volume;// [ 0 .. 1 ]
-
- REALTYPE Panning;// [ 0 .. 1 ]
-
- Envelope *AmpEnvelope;
- LFO *AmpLfo;
-
- struct {
- int Enabled;
- REALTYPE initialvalue,dt,t;
- } Punch;
-
- /******************************************
- * FILTER GLOBAL PARAMETERS *
- ******************************************/
- Filter *GlobalFilterL,*GlobalFilterR;
-
- REALTYPE FilterCenterPitch;//octaves
- REALTYPE FilterQ;
- REALTYPE FilterFreqTracking;
-
- Envelope *FilterEnvelope;
-
- LFO *FilterLfo;
- } NoteGlobalPar;
-
-
-
- /***********************************************************/
- /* VOICE PARAMETERS */
- /***********************************************************/
- struct ADnoteVoice{
+ /*****************************************************************/
+ /* GLOBAL PARAMETERS */
+ /*****************************************************************/
+
+ struct ADnoteGlobal {
+ /******************************************
+ * FREQUENCY GLOBAL PARAMETERS *
+ ******************************************/
+ REALTYPE Detune;//cents
+
+ Envelope *FreqEnvelope;
+ LFO *FreqLfo;
+
+ /********************************************
+ * AMPLITUDE GLOBAL PARAMETERS *
+ ********************************************/
+ REALTYPE Volume;// [ 0 .. 1 ]
+
+ REALTYPE Panning;// [ 0 .. 1 ]
+
+ Envelope *AmpEnvelope;
+ LFO *AmpLfo;
+
+ struct {
+ int Enabled;
+ REALTYPE initialvalue,dt,t;
+ } Punch;
+
+ /******************************************
+ * FILTER GLOBAL PARAMETERS *
+ ******************************************/
+ Filter *GlobalFilterL,*GlobalFilterR;
+
+ REALTYPE FilterCenterPitch;//octaves
+ REALTYPE FilterQ;
+ REALTYPE FilterFreqTracking;
+
+ Envelope *FilterEnvelope;
+
+ LFO *FilterLfo;
+ } NoteGlobalPar;
+
+
+
+ /***********************************************************/
+ /* VOICE PARAMETERS */
+ /***********************************************************/
+ struct ADnoteVoice {
/* If the voice is enabled */
- ONOFFTYPE Enabled;
+ ONOFFTYPE Enabled;
+
+ /* Voice Type (sound/noise)*/
+ int noisetype;
- /* Voice Type (sound/noise)*/
- int noisetype;
+ /* Filter Bypass */
+ int filterbypass;
- /* Filter Bypass */
- int filterbypass;
-
- /* Delay (ticks) */
- int DelayTicks;
-
- /* Waveform of the Voice */
- REALTYPE *OscilSmp;
+ /* Delay (ticks) */
+ int DelayTicks;
+
+ /* Waveform of the Voice */
+ REALTYPE *OscilSmp;
/************************************
- * FREQUENCY PARAMETERS *
- ************************************/
- int fixedfreq;//if the frequency is fixed to 440 Hz
- int fixedfreqET;//if the "fixed" frequency varies according to the note (ET)
+ * FREQUENCY PARAMETERS *
+ ************************************/
+ int fixedfreq;//if the frequency is fixed to 440 Hz
+ int fixedfreqET;//if the "fixed" frequency varies according to the note (ET)
+
+ // cents = basefreq*VoiceDetune
+ REALTYPE Detune,FineDetune;
+
+ Envelope *FreqEnvelope;
+ LFO *FreqLfo;
+
- // cents = basefreq*VoiceDetune
- REALTYPE Detune,FineDetune;
-
- Envelope *FreqEnvelope;
- LFO *FreqLfo;
-
+ /***************************
+ * AMPLITUDE PARAMETERS *
+ ***************************/
- /***************************
- * AMPLITUDE PARAMETERS *
- ***************************/
+ /* Panning 0.0=left, 0.5 - center, 1.0 = right */
+ REALTYPE Panning;
+ REALTYPE Volume;// [-1.0 .. 1.0]
- /* Panning 0.0=left, 0.5 - center, 1.0 = right */
- REALTYPE Panning;
- REALTYPE Volume;// [-1.0 .. 1.0]
+ Envelope *AmpEnvelope;
+ LFO *AmpLfo;
- Envelope *AmpEnvelope;
- LFO *AmpLfo;
+ /*************************
+ * FILTER PARAMETERS *
+ *************************/
- /*************************
- * FILTER PARAMETERS *
- *************************/
-
- Filter *VoiceFilter;
-
- REALTYPE FilterCenterPitch;/* Filter center Pitch*/
- REALTYPE FilterFreqTracking;
+ Filter *VoiceFilter;
- Envelope *FilterEnvelope;
- LFO *FilterLfo;
+ REALTYPE FilterCenterPitch;/* Filter center Pitch*/
+ REALTYPE FilterFreqTracking;
+ Envelope *FilterEnvelope;
+ LFO *FilterLfo;
- /****************************
- * MODULLATOR PARAMETERS *
- ****************************/
- FMTYPE FMEnabled;
+ /****************************
+ * MODULLATOR PARAMETERS *
+ ****************************/
- int FMVoice;
+ FMTYPE FMEnabled;
- // Voice Output used by other voices if use this as modullator
- REALTYPE *VoiceOut;
+ int FMVoice;
- /* Wave of the Voice */
- REALTYPE *FMSmp;
+ // Voice Output used by other voices if use this as modullator
+ REALTYPE *VoiceOut;
- REALTYPE FMVolume;
- REALTYPE FMDetune; //in cents
-
- Envelope *FMFreqEnvelope;
- Envelope *FMAmpEnvelope;
- } NoteVoicePar[NUM_VOICES];
+ /* Wave of the Voice */
+ REALTYPE *FMSmp;
+ REALTYPE FMVolume;
+ REALTYPE FMDetune; //in cents
- /********************************************************/
- /* INTERNAL VALUES OF THE NOTE AND OF THE VOICES */
- /********************************************************/
+ Envelope *FMFreqEnvelope;
+ Envelope *FMAmpEnvelope;
+ } NoteVoicePar[NUM_VOICES];
+
+
+ /********************************************************/
+ /* INTERNAL VALUES OF THE NOTE AND OF THE VOICES */
+ /********************************************************/
//time from the start of the note
REALTYPE time;
//fractional part (skip)
- REALTYPE oscposlo[NUM_VOICES],oscfreqlo[NUM_VOICES];
+ REALTYPE oscposlo[NUM_VOICES],oscfreqlo[NUM_VOICES];
//integer part (skip)
int oscposhi[NUM_VOICES],oscfreqhi[NUM_VOICES];
//fractional part (skip) of the Modullator
- REALTYPE oscposloFM[NUM_VOICES],oscfreqloFM[NUM_VOICES];
+ REALTYPE oscposloFM[NUM_VOICES],oscfreqloFM[NUM_VOICES];
//integer part (skip) of the Modullator
unsigned short int oscposhiFM[NUM_VOICES],oscfreqhiFM[NUM_VOICES];
//used to compute and interpolate the amplitudes of voices and modullators
REALTYPE 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)
REALTYPE FMoldsmp[NUM_VOICES];
-
+
//temporary buffer
REALTYPE *tmpwave;
-
+
//Filter bypass samples
REALTYPE *bypassl,*bypassr;
- //interpolate the amplitudes
+ //interpolate the amplitudes
REALTYPE globaloldamplitude,globalnewamplitude;
-
+
//1 - if it is the fitst tick (used to fade in the sound)
char firsttick[NUM_VOICES];
-
- //1 if the note has portamento
+
+ //1 if the note has portamento
int portamento;
-
+
//how the fine detunes are made bigger or smaller
REALTYPE bandwidthDetuneMultiplier;
// Legato vars
struct {
- bool silent;
- REALTYPE lastfreq;
- LegatoMsg msg;
- int decounter;
- struct { // Fade In/Out vars
- int length;
- REALTYPE m, step;
- } fade;
- struct { // Note parameters
- REALTYPE freq, vel;
- int portamento, midinote;
- } param;
+ bool silent;
+ REALTYPE lastfreq;
+ LegatoMsg msg;
+ int decounter;
+ struct { // Fade In/Out vars
+ int length;
+ REALTYPE m, step;
+ } fade;
+ struct { // Note parameters
+ REALTYPE freq, vel;
+ int portamento, midinote;
+ } param;
} Legato;
};
diff --git a/src/Synth/Envelope.C b/src/Synth/Envelope.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Envelope.C - Envelope implementation
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -23,7 +23,8 @@
#include <stdio.h>
#include "Envelope.h"
-Envelope::Envelope(EnvelopeParams *envpars,REALTYPE basefreq){
+Envelope::Envelope(EnvelopeParams *envpars,REALTYPE basefreq)
+{
int i;
envpoints=envpars->Penvpoints;
if (envpoints>MAX_ENVELOPE_POINTS) envpoints=MAX_ENVELOPE_POINTS;
@@ -33,7 +34,7 @@ Envelope::Envelope(EnvelopeParams *envpars,REALTYPE basefreq){
linearenvelope=envpars->Plinearenvelope;
if (envpars->Pfreemode==0) envpars->converttofree();
-
+
REALTYPE bufferdt=SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE;
int mode=envpars->Envmode;
@@ -44,22 +45,27 @@ Envelope::Envelope(EnvelopeParams *envpars,REALTYPE basefreq){
for (i=0;i<MAX_ENVELOPE_POINTS;i++) {
REALTYPE tmp=envpars->getdt(i)/1000.0*envstretch;
- if (tmp>bufferdt) envdt[i]=bufferdt/tmp;
- else envdt[i]=2.0;//any value larger than 1
-
- switch (mode){
- case 2:envval[i]=(1.0-envpars->Penvval[i]/127.0)*MIN_ENVELOPE_DB;
- break;
- case 3:envval[i]=(pow(2,6.0*fabs(envpars->Penvval[i]-64.0)/64.0)-1.0)*100.0;
- if (envpars->Penvval[i]<64) envval[i]=-envval[i];
- break;
- case 4:envval[i]=(envpars->Penvval[i]-64.0)/64.0*6.0;//6 octaves (filtru)
- break;
- case 5:envval[i]=(envpars->Penvval[i]-64.0)/64.0*10;
- break;
- default:envval[i]=envpars->Penvval[i]/127.0;
- };
-
+ if (tmp>bufferdt) envdt[i]=bufferdt/tmp;
+ else envdt[i]=2.0;//any value larger than 1
+
+ switch (mode) {
+ case 2:
+ envval[i]=(1.0-envpars->Penvval[i]/127.0)*MIN_ENVELOPE_DB;
+ break;
+ case 3:
+ envval[i]=(pow(2,6.0*fabs(envpars->Penvval[i]-64.0)/64.0)-1.0)*100.0;
+ if (envpars->Penvval[i]<64) envval[i]=-envval[i];
+ break;
+ case 4:
+ envval[i]=(envpars->Penvval[i]-64.0)/64.0*6.0;//6 octaves (filtru)
+ break;
+ case 5:
+ envval[i]=(envpars->Penvval[i]-64.0)/64.0*10;
+ break;
+ default:
+ envval[i]=envpars->Penvval[i]/127.0;
+ };
+
};
envdt[0]=1.0;
@@ -72,14 +78,16 @@ Envelope::Envelope(EnvelopeParams *envpars,REALTYPE basefreq){
envoutval=0.0;
};
-Envelope::~Envelope(){
+Envelope::~Envelope()
+{
};
/*
* Relase the key (note envelope)
*/
-void Envelope::relasekey(){
+void Envelope::relasekey()
+{
if (keyreleased==1) return;
keyreleased=1;
if (forcedrelase!=0) t=0.0;
@@ -88,78 +96,81 @@ void Envelope::relasekey(){
/*
* Envelope Output
*/
-REALTYPE Envelope::envout(){
+REALTYPE Envelope::envout()
+{
REALTYPE out;
if (envfinish!=0) {//if the envelope is finished
- envoutval=envval[envpoints-1];
- return(envoutval);
+ envoutval=envval[envpoints-1];
+ return(envoutval);
};
if ((currentpoint==envsustain+1)&&(keyreleased==0)) {//if it is sustaining now
- envoutval=envval[envsustain];
- return(envoutval);
+ envoutval=envval[envsustain];
+ return(envoutval);
};
- if ((keyreleased!=0) && (forcedrelase!=0)){//do the forced release
-
+ if ((keyreleased!=0) && (forcedrelase!=0)) {//do the forced release
+
int tmp=(envsustain<0) ? (envpoints-1):(envsustain+1);//if there is no sustain point, use the last point for release
- if (envdt[tmp]<0.00000001) out=envval[tmp];
- else out=envoutval+(envval[tmp]-envoutval)*t;
- t+=envdt[tmp]*envstretch;
-
- if (t>=1.0) {
- currentpoint=envsustain+2;
- forcedrelase=0;
- t=0.0;
- inct=envdt[currentpoint];
- if ((currentpoint>=envpoints)||(envsustain<0)) envfinish=1;
- };
- return(out);
+ if (envdt[tmp]<0.00000001) out=envval[tmp];
+ else out=envoutval+(envval[tmp]-envoutval)*t;
+ t+=envdt[tmp]*envstretch;
+
+ if (t>=1.0) {
+ currentpoint=envsustain+2;
+ forcedrelase=0;
+ t=0.0;
+ inct=envdt[currentpoint];
+ if ((currentpoint>=envpoints)||(envsustain<0)) envfinish=1;
+ };
+ return(out);
};
if (inct>=1.0) out=envval[currentpoint];
- else out=envval[currentpoint-1]+(envval[currentpoint]-envval[currentpoint-1])*t;
+ else out=envval[currentpoint-1]+(envval[currentpoint]-envval[currentpoint-1])*t;
t+=inct;
- if (t>=1.0){
- if (currentpoint>=envpoints-1) envfinish=1;
- else currentpoint++;
- t=0.0;
- inct=envdt[currentpoint];
+ if (t>=1.0) {
+ if (currentpoint>=envpoints-1) envfinish=1;
+ else currentpoint++;
+ t=0.0;
+ inct=envdt[currentpoint];
};
- envoutval=out;
+ envoutval=out;
return (out);
};
/*
* Envelope Output (dB)
*/
-REALTYPE Envelope::envout_dB(){
+REALTYPE Envelope::envout_dB()
+{
REALTYPE out;
if (linearenvelope!=0) return (envout());
-
+
if ((currentpoint==1)&&((keyreleased==0)||(forcedrelase==0))) {//first point is always lineary interpolated
- REALTYPE v1=dB2rap(envval[0]);
- REALTYPE v2=dB2rap(envval[1]);
- out=v1+(v2-v1)*t;
-
- t+=inct;
- if (t>=1.0) {
- t=0.0;
- inct=envdt[2];
- currentpoint++;
- out=v2;
- };
-
- if (out>0.001) envoutval=rap2dB(out);
- else envoutval=-40.0;
+ REALTYPE v1=dB2rap(envval[0]);
+ REALTYPE v2=dB2rap(envval[1]);
+ out=v1+(v2-v1)*t;
+
+ t+=inct;
+ if (t>=1.0) {
+ t=0.0;
+ inct=envdt[2];
+ currentpoint++;
+ out=v2;
+ };
+
+ if (out>0.001) envoutval=rap2dB(out);
+ else envoutval=-40.0;
} else out=dB2rap(envout());
return(out);
};
-int Envelope::finished(){
+int Envelope::finished()
+{
return(envfinish);
};
diff --git a/src/Synth/Envelope.h b/src/Synth/Envelope.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Envelope.h - Envelope implementation
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,7 +28,8 @@
#include "../Params/EnvelopeParams.h"
/**Implementation of a general Envelope*/
-class Envelope{
+class Envelope
+{
public:
/**Constructor*/
@@ -48,15 +49,15 @@ private:
int envsustain;//"-1" means disabled
REALTYPE envdt[MAX_ENVELOPE_POINTS];//millisecons
REALTYPE envval[MAX_ENVELOPE_POINTS];// [0.0 .. 1.0]
- REALTYPE envstretch;
+ REALTYPE envstretch;
int linearenvelope;
int currentpoint; //current envelope point (starts from 1)
int forcedrelase;
char keyreleased; //if the key was released /** \todo figure out WHY IS THIS A CHAR*/
- char envfinish; /** \todo figure out WHY IS THIS A CHAR*/
+ char envfinish; /** \todo figure out WHY IS THIS A CHAR*/
REALTYPE t; // the time from the last point
- REALTYPE inct;// the time increment
+ REALTYPE inct;// the time increment
REALTYPE envoutval;//used to do the forced release
};
diff --git a/src/Synth/LFO.C b/src/Synth/LFO.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
LFO.C - LFO implementation
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,37 +27,44 @@
#include "LFO.h"
-LFO::LFO(LFOParams *lfopars,REALTYPE basefreq){
+LFO::LFO(LFOParams *lfopars,REALTYPE basefreq)
+{
if (lfopars->Pstretch==0) lfopars->Pstretch=1;
REALTYPE lfostretch=pow(basefreq/440.0,(lfopars->Pstretch-64.0)/63.0);//max 2x/octave
REALTYPE lfofreq=(pow(2,lfopars->Pfreq*10.0)-1.0)/12.0*lfostretch;
incx=fabs(lfofreq)*(REALTYPE)SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE;
- if (lfopars->Pcontinous==0){
- if (lfopars->Pstartphase==0) x=RND;
- else x=fmod((lfopars->Pstartphase-64.0)/127.0+1.0,1.0);
+ if (lfopars->Pcontinous==0) {
+ if (lfopars->Pstartphase==0) x=RND;
+ else x=fmod((lfopars->Pstartphase-64.0)/127.0+1.0,1.0);
} else {
- REALTYPE tmp=fmod(lfopars->time*incx,1.0);
- x=fmod((lfopars->Pstartphase-64.0)/127.0+1.0+tmp,1.0);
+ REALTYPE tmp=fmod(lfopars->time*incx,1.0);
+ x=fmod((lfopars->Pstartphase-64.0)/127.0+1.0+tmp,1.0);
};
//Limit the Frequency(or else...)
if (incx>0.49999999) incx=0.499999999;
-
+
lfornd=lfopars->Prandomness/127.0;
- if (lfornd<0.0) lfornd=0.0; else if (lfornd>1.0) lfornd=1.0;
+ if (lfornd<0.0) lfornd=0.0;
+ else if (lfornd>1.0) lfornd=1.0;
// lfofreqrnd=pow(lfopars->Pfreqrand/127.0,2.0)*2.0*4.0;
lfofreqrnd=pow(lfopars->Pfreqrand/127.0,2.0)*4.0;
- switch (lfopars->fel){
- case 1:lfointensity=lfopars->Pintensity/127.0;break;
- case 2:lfointensity=lfopars->Pintensity/127.0*4.0;break;//in octave
- default:lfointensity=pow(2,lfopars->Pintensity/127.0*11.0)-1.0;//in centi
- x-=0.25;//chance the starting phase
- break;
+ switch (lfopars->fel) {
+ case 1:
+ lfointensity=lfopars->Pintensity/127.0;
+ break;
+ case 2:
+ lfointensity=lfopars->Pintensity/127.0*4.0;
+ break;//in octave
+ default:
+ lfointensity=pow(2,lfopars->Pintensity/127.0*11.0)-1.0;//in centi
+ x-=0.25;//chance the starting phase
+ break;
};
amp1=(1-lfornd)+lfornd*RND;
@@ -70,57 +77,60 @@ LFO::LFO(LFOParams *lfopars,REALTYPE basefreq){
computenextincrnd();//twice because I want incrnd & nextincrnd to be random
};
-LFO::~LFO(){
+LFO::~LFO()
+{
};
/*
* LFO out
*/
-REALTYPE LFO::lfoout(){
- REALTYPE out;
- switch (lfotype){
- case 1: //LFO_TRIANGLE
- if ((x>=0.0)&&(x<0.25)) out=4.0*x;
- else if ((x>0.25)&&(x<0.75)) out=2-4*x;
- else out=4.0*x-4.0;
- break;
- case 2: //LFO_SQUARE
- if (x<0.5) out=-1;
- else out=1;
- break;
- case 3: //LFO_RAMPUP
- out=(x-0.5)*2.0;
- break;
- case 4: //LFO_RAMPDOWN
- out=(0.5-x)*2.0;
- break;
- case 5: //LFO_EXP_DOWN 1
- out=pow(0.05,x)*2.0-1.0;
- break;
- case 6: //LFO_EXP_DOWN 2
- out=pow(0.001,x)*2.0-1.0;
- break;
- default:out=cos(x*2.0*PI);//LFO_SINE
+REALTYPE LFO::lfoout()
+{
+ REALTYPE out;
+ switch (lfotype) {
+ case 1: //LFO_TRIANGLE
+ if ((x>=0.0)&&(x<0.25)) out=4.0*x;
+ else if ((x>0.25)&&(x<0.75)) out=2-4*x;
+ else out=4.0*x-4.0;
+ break;
+ case 2: //LFO_SQUARE
+ if (x<0.5) out=-1;
+ else out=1;
+ break;
+ case 3: //LFO_RAMPUP
+ out=(x-0.5)*2.0;
+ break;
+ case 4: //LFO_RAMPDOWN
+ out=(0.5-x)*2.0;
+ break;
+ case 5: //LFO_EXP_DOWN 1
+ out=pow(0.05,x)*2.0-1.0;
+ break;
+ case 6: //LFO_EXP_DOWN 2
+ out=pow(0.001,x)*2.0-1.0;
+ break;
+ default:
+ out=cos(x*2.0*PI);//LFO_SINE
};
if ((lfotype==0)||(lfotype==1)) out*=lfointensity*(amp1+x*(amp2-amp1));
- else out*=lfointensity*amp2;
+ else out*=lfointensity*amp2;
if (lfodelay<0.00001) {
- if (freqrndenabled==0) x+=incx;
- else {
- float tmp=(incrnd*(1.0-x)+nextincrnd*x);
- if (tmp>1.0) tmp=1.0;
- else if (tmp<0.0) tmp=0.0;
- x+=incx*tmp;
- };
- if (x>=1) {
- x=fmod(x,1.0);
- amp1=amp2;
- amp2=(1-lfornd)+lfornd*RND;
-
- computenextincrnd();
- };
+ if (freqrndenabled==0) x+=incx;
+ else {
+ float tmp=(incrnd*(1.0-x)+nextincrnd*x);
+ if (tmp>1.0) tmp=1.0;
+ else if (tmp<0.0) tmp=0.0;
+ x+=incx*tmp;
+ };
+ if (x>=1) {
+ x=fmod(x,1.0);
+ amp1=amp2;
+ amp2=(1-lfornd)+lfornd*RND;
+
+ computenextincrnd();
+ };
} else lfodelay-=(REALTYPE)SOUND_BUFFER_SIZE/(REALTYPE)SAMPLE_RATE;
return(out);
};
@@ -128,18 +138,20 @@ REALTYPE LFO::lfoout(){
/*
* LFO out (for amplitude)
*/
-REALTYPE LFO::amplfoout(){
+REALTYPE LFO::amplfoout()
+{
REALTYPE out;
out=1.0-lfointensity+lfoout();
if (out<-1.0) out=-1.0;
- else if (out>1.0) out=1.0;
+ else if (out>1.0) out=1.0;
return(out);
};
-void LFO::computenextincrnd(){
- if (freqrndenabled==0) return;
- incrnd=nextincrnd;
- nextincrnd=pow(0.5,lfofreqrnd)+RND*(pow(2.0,lfofreqrnd)-1.0);
+void LFO::computenextincrnd()
+{
+ if (freqrndenabled==0) return;
+ incrnd=nextincrnd;
+ nextincrnd=pow(0.5,lfofreqrnd)+RND*(pow(2.0,lfofreqrnd)-1.0);
};
diff --git a/src/Synth/LFO.h b/src/Synth/LFO.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
LFO.h - LFO implementation
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,8 +27,9 @@
#include "../Params/LFOParams.h"
/**Class for creating Low Frequency Ocillators*/
-class LFO{
- public:
+class LFO
+{
+public:
/**Constructor
*
* @param lfopars pointer to a LFOParams object
@@ -39,7 +40,7 @@ class LFO{
~LFO();
REALTYPE lfoout();
REALTYPE amplfoout();
- private:
+private:
REALTYPE x;
REALTYPE incx,incrnd,nextincrnd;
REALTYPE amp1,amp2;// used for randomness
@@ -49,10 +50,10 @@ class LFO{
/**\todo see if an enum would be better here*/
char lfotype;
int freqrndenabled;
-
-
+
+
void computenextincrnd();
-
+
};
#endif
diff --git a/src/Synth/OscilGen.C b/src/Synth/OscilGen.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
OscilGen.C - Waveform generator for ADnote
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -31,7 +31,8 @@ REALTYPE *OscilGen::tmpsmps;//this array stores some termporary data and it has
FFTFREQS OscilGen::outoscilFFTfreqs;
-OscilGen::OscilGen(FFTwrapper *fft_,Resonance *res_):Presets(){
+OscilGen::OscilGen(FFTwrapper *fft_,Resonance *res_):Presets()
+{
setpresettype("Poscilgen");
fft=fft_;
res=res_;
@@ -44,28 +45,41 @@ OscilGen::OscilGen(FFTwrapper *fft_,Resonance *res_):Presets(){
defaults();
};
-OscilGen::~OscilGen(){
+OscilGen::~OscilGen()
+{
deleteFFTFREQS(&basefuncFFTfreqs);
deleteFFTFREQS(&oscilFFTfreqs);
};
-void OscilGen::defaults(){
-
- oldbasefunc=0;oldbasepar=64;oldhmagtype=0;oldwaveshapingfunction=0;oldwaveshaping=64;
- oldbasefuncmodulation=0;oldharmonicshift=0;oldbasefuncmodulationpar1=0;oldbasefuncmodulationpar2=0;oldbasefuncmodulationpar3=0;
- oldmodulation=0;oldmodulationpar1=0;oldmodulationpar2=0;oldmodulationpar3=0;
-
- for (int i=0;i<MAX_AD_HARMONICS;i++){
- hmag[i]=0.0;
- hphase[i]=0.0;
- Phmag[i]=64;
- Phphase[i]=64;
+void OscilGen::defaults()
+{
+
+ oldbasefunc=0;
+ oldbasepar=64;
+ oldhmagtype=0;
+ oldwaveshapingfunction=0;
+ oldwaveshaping=64;
+ oldbasefuncmodulation=0;
+ oldharmonicshift=0;
+ oldbasefuncmodulationpar1=0;
+ oldbasefuncmodulationpar2=0;
+ oldbasefuncmodulationpar3=0;
+ oldmodulation=0;
+ oldmodulationpar1=0;
+ oldmodulationpar2=0;
+ oldmodulationpar3=0;
+
+ for (int i=0;i<MAX_AD_HARMONICS;i++) {
+ hmag[i]=0.0;
+ hphase[i]=0.0;
+ Phmag[i]=64;
+ Phphase[i]=64;
};
Phmag[0]=127;
Phmagtype=0;
if (ADvsPAD) Prand=127;//max phase randomness (usefull if the oscil will be imported to a ADsynth from a PADsynth
- else Prand=64;//no randomness
+ else Prand=64;//no randomness
Pcurrentbasefunc=0;
Pbasefuncpar=64;
@@ -91,7 +105,7 @@ void OscilGen::defaults(){
Pamprandpower=64;
Pamprandtype=0;
-
+
Pharmonicshift=0;
Pharmonicshiftfirst=0;
@@ -99,20 +113,22 @@ void OscilGen::defaults(){
Padaptiveharmonicspower=100;
Padaptiveharmonicsbasefreq=128;
Padaptiveharmonicspar=50;
-
+
for (int i=0;i<OSCIL_SIZE/2;i++) {
- oscilFFTfreqs.s[i]=0.0;
- oscilFFTfreqs.c[i]=0.0;
- basefuncFFTfreqs.s[i]=0.0;
- basefuncFFTfreqs.c[i]=0.0;
+ oscilFFTfreqs.s[i]=0.0;
+ oscilFFTfreqs.c[i]=0.0;
+ basefuncFFTfreqs.s[i]=0.0;
+ basefuncFFTfreqs.c[i]=0.0;
};
oscilprepared=0;
- oldfilterpars=0;oldsapars=0;
+ oldfilterpars=0;
+ oldsapars=0;
prepare();
};
-void OscilGen::convert2sine(int magtype){
- REALTYPE mag[MAX_AD_HARMONICS],phase[MAX_AD_HARMONICS];
+void OscilGen::convert2sine(int magtype)
+{
+ REALTYPE mag[MAX_AD_HARMONICS],phase[MAX_AD_HARMONICS];
REALTYPE oscil[OSCIL_SIZE];
FFTFREQS freqs;
newFFTFREQS(&freqs,OSCIL_SIZE/2);
@@ -123,115 +139,128 @@ void OscilGen::convert2sine(int magtype){
delete(fft);
REALTYPE max=0.0;
-
+
mag[0]=0;
phase[0]=0;
- for (int i=0;i<MAX_AD_HARMONICS;i++){
+ for (int i=0;i<MAX_AD_HARMONICS;i++) {
mag[i]=sqrt(pow(freqs.s[i+1],2)+pow(freqs.c[i+1],2.0));
- phase[i]=atan2(freqs.c[i+1],freqs.s[i+1]);
- if (max<mag[i]) max=mag[i];
+ phase[i]=atan2(freqs.c[i+1],freqs.s[i+1]);
+ if (max<mag[i]) max=mag[i];
};
if (max<0.00001) max=1.0;
-
+
defaults();
-
- for (int i=0;i<MAX_AD_HARMONICS-1;i++){
- REALTYPE newmag=mag[i]/max;
- REALTYPE newphase=phase[i];
-
- Phmag[i]=(int) ((newmag)*64.0)+64;
-
- Phphase[i]=64-(int) (64.0*newphase/PI);
- if (Phphase[i]>127) Phphase[i]=127;
-
- if (Phmag[i]==64) Phphase[i]=64;
+
+ for (int i=0;i<MAX_AD_HARMONICS-1;i++) {
+ REALTYPE newmag=mag[i]/max;
+ REALTYPE newphase=phase[i];
+
+ Phmag[i]=(int) ((newmag)*64.0)+64;
+
+ Phphase[i]=64-(int) (64.0*newphase/PI);
+ if (Phphase[i]>127) Phphase[i]=127;
+
+ if (Phmag[i]==64) Phphase[i]=64;
};
deleteFFTFREQS(&freqs);
prepare();
};
-/*
- * Base Functions - START
+/*
+ * Base Functions - START
*/
-REALTYPE OscilGen::basefunc_pulse(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_pulse(REALTYPE x,REALTYPE a)
+{
return((fmod(x,1.0)<a)?-1.0:1.0);
};
-REALTYPE OscilGen::basefunc_saw(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_saw(REALTYPE x,REALTYPE a)
+{
if (a<0.00001) a=0.00001;
- else if (a>0.99999) a=0.99999;
+ else if (a>0.99999) a=0.99999;
x=fmod(x,1);
if (x<a) return(x/a*2.0-1.0);
- else return((1.0-x)/(1.0-a)*2.0-1.0);
+ else return((1.0-x)/(1.0-a)*2.0-1.0);
};
-REALTYPE OscilGen::basefunc_triangle(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_triangle(REALTYPE x,REALTYPE a)
+{
x=fmod(x+0.25,1);
a=1-a;
if (a<0.00001) a=0.00001;
if (x<0.5) x=x*4-1.0;
- else x=(1.0-x)*4-1.0;
+ else x=(1.0-x)*4-1.0;
x/=-a;
if (x<-1.0) x=-1.0;
if (x>1.0) x=1.0;
return(x);
};
-REALTYPE OscilGen::basefunc_power(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_power(REALTYPE x,REALTYPE a)
+{
x=fmod(x,1);
if (a<0.00001) a=0.00001;
- else if (a>0.99999) a=0.99999;
+ else if (a>0.99999) a=0.99999;
return(pow(x,exp((a-0.5)*10.0))*2.0-1.0);
};
-REALTYPE OscilGen::basefunc_gauss(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_gauss(REALTYPE x,REALTYPE a)
+{
x=fmod(x,1)*2.0-1.0;
if (a<0.00001) a=0.00001;
return(exp(-x*x*(exp(a*8)+5.0))*2.0-1.0);
};
-REALTYPE OscilGen::basefunc_diode(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_diode(REALTYPE x,REALTYPE a)
+{
if (a<0.00001) a=0.00001;
- else if (a>0.99999) a=0.99999;
+ else if (a>0.99999) a=0.99999;
a=a*2.0-1.0;
x=cos((x+0.5)*2.0*PI)-a;
if (x<0.0) x=0.0;
return(x/(1.0-a)*2-1.0);
};
-REALTYPE OscilGen::basefunc_abssine(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_abssine(REALTYPE x,REALTYPE a)
+{
x=fmod(x,1);
if (a<0.00001) a=0.00001;
- else if (a>0.99999) a=0.99999;
+ else if (a>0.99999) a=0.99999;
return(sin(pow(x,exp((a-0.5)*5.0))*PI)*2.0-1.0);
};
-REALTYPE OscilGen::basefunc_pulsesine(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_pulsesine(REALTYPE x,REALTYPE a)
+{
if (a<0.00001) a=0.00001;
x=(fmod(x,1)-0.5)*exp((a-0.5)*log(128));
if (x<-0.5) x=-0.5;
- else if (x>0.5) x=0.5;
+ else if (x>0.5) x=0.5;
x=sin(x*PI*2.0);
return(x);
};
-REALTYPE OscilGen::basefunc_stretchsine(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_stretchsine(REALTYPE x,REALTYPE a)
+{
x=fmod(x+0.5,1)*2.0-1.0;
- a=(a-0.5)*4;if (a>0.0) a*=2;
+ a=(a-0.5)*4;
+ if (a>0.0) a*=2;
a=pow(3.0,a);
REALTYPE b=pow(fabs(x),a);
if (x<0) b=-b;
return(-sin(b*PI));
};
-REALTYPE OscilGen::basefunc_chirp(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_chirp(REALTYPE x,REALTYPE a)
+{
x=fmod(x,1.0)*2.0*PI;
- a=(a-0.5)*4;if (a<0.0) a*=2.0;
+ a=(a-0.5)*4;
+ if (a<0.0) a*=2.0;
a=pow(3.0,a);
return(sin(x/2.0)*sin(a*x*x));
};
-REALTYPE OscilGen::basefunc_absstretchsine(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_absstretchsine(REALTYPE x,REALTYPE a)
+{
x=fmod(x+0.5,1)*2.0-1.0;
a=(a-0.5)*9;
a=pow(3.0,a);
@@ -240,197 +269,235 @@ REALTYPE OscilGen::basefunc_absstretchsine(REALTYPE x,REALTYPE a){
return(-pow(sin(b*PI),2));
};
-REALTYPE OscilGen::basefunc_chebyshev(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_chebyshev(REALTYPE x,REALTYPE a)
+{
a=a*a*a*30.0+1.0;
return(cos(acos(x*2.0-1.0)*a));
};
-REALTYPE OscilGen::basefunc_sqr(REALTYPE x,REALTYPE a){
+REALTYPE OscilGen::basefunc_sqr(REALTYPE x,REALTYPE a)
+{
a=a*a*a*a*160.0+0.001;
return(-atan(sin(x*2.0*PI)*a));
};
-/*
+/*
* Base Functions - END
*/
-/*
+/*
* Get the base function
*/
-void OscilGen::getbasefunction(REALTYPE *smps){
- int i;
+void OscilGen::getbasefunction(REALTYPE *smps)
+{
+ int i;
REALTYPE par=(Pbasefuncpar+0.5)/128.0;
if (Pbasefuncpar==64) par=0.5;
-
+
REALTYPE basefuncmodulationpar1=Pbasefuncmodulationpar1/127.0,
- basefuncmodulationpar2=Pbasefuncmodulationpar2/127.0,
- basefuncmodulationpar3=Pbasefuncmodulationpar3/127.0;
-
- switch(Pbasefuncmodulation){
- case 1:basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*5.0)-1.0)/10.0;
- basefuncmodulationpar3=floor((pow(2,basefuncmodulationpar3*5.0)-1.0));
- if (basefuncmodulationpar3<0.9999) basefuncmodulationpar3=-1.0;
- break;
- case 2:basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*5.0)-1.0)/10.0;
- basefuncmodulationpar3=1.0+floor((pow(2,basefuncmodulationpar3*5.0)-1.0));
- break;
- case 3:basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*7.0)-1.0)/10.0;
- basefuncmodulationpar3=0.01+(pow(2,basefuncmodulationpar3*16.0)-1.0)/10.0;
- break;
- };
+ basefuncmodulationpar2=Pbasefuncmodulationpar2/127.0,
+ basefuncmodulationpar3=Pbasefuncmodulationpar3/127.0;
+
+ switch (Pbasefuncmodulation) {
+ case 1:
+ basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*5.0)-1.0)/10.0;
+ basefuncmodulationpar3=floor((pow(2,basefuncmodulationpar3*5.0)-1.0));
+ if (basefuncmodulationpar3<0.9999) basefuncmodulationpar3=-1.0;
+ break;
+ case 2:
+ basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*5.0)-1.0)/10.0;
+ basefuncmodulationpar3=1.0+floor((pow(2,basefuncmodulationpar3*5.0)-1.0));
+ break;
+ case 3:
+ basefuncmodulationpar1=(pow(2,basefuncmodulationpar1*7.0)-1.0)/10.0;
+ basefuncmodulationpar3=0.01+(pow(2,basefuncmodulationpar3*16.0)-1.0)/10.0;
+ break;
+ };
// printf("%.5f %.5f\n",basefuncmodulationpar1,basefuncmodulationpar3);
for (i=0;i<OSCIL_SIZE;i++) {
- REALTYPE t=i*1.0/OSCIL_SIZE;
-
- switch(Pbasefuncmodulation){
- case 1:t=t*basefuncmodulationpar3+sin((t+basefuncmodulationpar2)*2.0*PI)*basefuncmodulationpar1;//rev
- break;
- case 2:t=t+sin((t*basefuncmodulationpar3+basefuncmodulationpar2)*2.0*PI)*basefuncmodulationpar1;//sine
- break;
- case 3:t=t+pow((1.0-cos((t+basefuncmodulationpar2)*2.0*PI))*0.5,basefuncmodulationpar3)*basefuncmodulationpar1;//power
- break;
- };
-
- t=t-floor(t);
-
- switch (Pcurrentbasefunc){
- case 1:smps[i]=basefunc_triangle(t,par);
- break;
- case 2:smps[i]=basefunc_pulse(t,par);
- break;
- case 3:smps[i]=basefunc_saw(t,par);
- break;
- case 4:smps[i]=basefunc_power(t,par);
- break;
- case 5:smps[i]=basefunc_gauss(t,par);
- break;
- case 6:smps[i]=basefunc_diode(t,par);
- break;
- case 7:smps[i]=basefunc_abssine(t,par);
- break;
- case 8:smps[i]=basefunc_pulsesine(t,par);
- break;
- case 9:smps[i]=basefunc_stretchsine(t,par);
- break;
- case 10:smps[i]=basefunc_chirp(t,par);
- break;
- case 11:smps[i]=basefunc_absstretchsine(t,par);
- break;
- case 12:smps[i]=basefunc_chebyshev(t,par);
- break;
- case 13:smps[i]=basefunc_sqr(t,par);
- break;
- default:smps[i]=-sin(2.0*PI*i/OSCIL_SIZE);
- };
+ REALTYPE t=i*1.0/OSCIL_SIZE;
+
+ switch (Pbasefuncmodulation) {
+ case 1:
+ t=t*basefuncmodulationpar3+sin((t+basefuncmodulationpar2)*2.0*PI)*basefuncmodulationpar1;//rev
+ break;
+ case 2:
+ t=t+sin((t*basefuncmodulationpar3+basefuncmodulationpar2)*2.0*PI)*basefuncmodulationpar1;//sine
+ break;
+ case 3:
+ t=t+pow((1.0-cos((t+basefuncmodulationpar2)*2.0*PI))*0.5,basefuncmodulationpar3)*basefuncmodulationpar1;//power
+ break;
+ };
+
+ t=t-floor(t);
+
+ switch (Pcurrentbasefunc) {
+ case 1:
+ smps[i]=basefunc_triangle(t,par);
+ break;
+ case 2:
+ smps[i]=basefunc_pulse(t,par);
+ break;
+ case 3:
+ smps[i]=basefunc_saw(t,par);
+ break;
+ case 4:
+ smps[i]=basefunc_power(t,par);
+ break;
+ case 5:
+ smps[i]=basefunc_gauss(t,par);
+ break;
+ case 6:
+ smps[i]=basefunc_diode(t,par);
+ break;
+ case 7:
+ smps[i]=basefunc_abssine(t,par);
+ break;
+ case 8:
+ smps[i]=basefunc_pulsesine(t,par);
+ break;
+ case 9:
+ smps[i]=basefunc_stretchsine(t,par);
+ break;
+ case 10:
+ smps[i]=basefunc_chirp(t,par);
+ break;
+ case 11:
+ smps[i]=basefunc_absstretchsine(t,par);
+ break;
+ case 12:
+ smps[i]=basefunc_chebyshev(t,par);
+ break;
+ case 13:
+ smps[i]=basefunc_sqr(t,par);
+ break;
+ default:
+ smps[i]=-sin(2.0*PI*i/OSCIL_SIZE);
+ };
};
};
-/*
+/*
* Filter the oscillator
*/
-void OscilGen::oscilfilter(){
+void OscilGen::oscilfilter()
+{
if (Pfiltertype==0) return;
REALTYPE par=1.0-Pfilterpar1/128.0;
REALTYPE par2=Pfilterpar2/127.0;
REALTYPE max=0.0,tmp=0.0,p2,x;
- for (int i=1;i<OSCIL_SIZE/2;i++){
- REALTYPE gain=1.0;
- switch(Pfiltertype){
- case 1: gain=pow(1.0-par*par*par*0.99,i);//lp
- tmp=par2*par2*par2*par2*0.5+0.0001;
- if (gain<tmp) gain=pow(gain,10.0)/pow(tmp,9.0);
- break;
- case 2: gain=1.0-pow(1.0-par*par,i+1);//hp1
- gain=pow(gain,par2*2.0+0.1);
- break;
- case 3: if (par<0.2) par=par*0.25+0.15;
- gain=1.0-pow(1.0-par*par*0.999+0.001,i*0.05*i+1.0);//hp1b
- tmp=pow(5.0,par2*2.0);
- gain=pow(gain,tmp);
- break;
- case 4: gain=i+1-pow(2,(1.0-par)*7.5);//bp1
- gain=1.0/(1.0+gain*gain/(i+1.0));
- tmp=pow(5.0,par2*2.0);
- gain=pow(gain,tmp);
- if (gain<1e-5) gain=1e-5;
- break;
- case 5: gain=i+1-pow(2,(1.0-par)*7.5);//bs1
- gain=pow(atan(gain/(i/10.0+1))/1.57,6);
- gain=pow(gain,par2*par2*3.9+0.1);
- break;
- case 6: tmp=pow(par2,0.33);
- gain=(i+1>pow(2,(1.0-par)*10)?0.0:1.0)*par2+(1.0-par2);//lp2
- break;
- case 7: tmp=pow(par2,0.33);
- //tmp=1.0-(1.0-par2)*(1.0-par2);
- gain=(i+1>pow(2,(1.0-par)*7)?1.0:0.0)*par2+(1.0-par2);//hp2
- if (Pfilterpar1==0) gain=1.0;
- break;
- case 8: tmp=pow(par2,0.33);
- //tmp=1.0-(1.0-par2)*(1.0-par2);
- gain=(fabs(pow(2,(1.0-par)*7)-i)>i/2+1?0.0:1.0)*par2+(1.0-par2);//bp2
- break;
- case 9: tmp=pow(par2,0.33);
- gain=(fabs(pow(2,(1.0-par)*7)-i)<i/2+1?0.0:1.0)*par2+(1.0-par2);//bs2
- break;
- case 10:tmp=pow(5.0,par2*2.0-1.0);
- tmp=pow(i/32.0,tmp)*32.0;
- if (Pfilterpar2==64) tmp=i;
- gain=cos(par*par*PI/2.0*tmp);//cos
- gain*=gain;
- break;
- case 11:tmp=pow(5.0,par2*2.0-1.0);
- tmp=pow(i/32.0,tmp)*32.0;
- if (Pfilterpar2==64) tmp=i;
- gain=sin(par*par*PI/2.0*tmp);//sin
- gain*=gain;
- break;
- case 12:p2=1.0-par+0.2;
- x=i/(64.0*p2*p2);
- if (x<0.0) x=0.0;
- else if (x>1.0) x=1.0;
- tmp=pow(1.0-par2,2.0);
- gain=cos(x*PI)*(1.0-tmp)+1.01+tmp;//low shelf
- break;
- case 13:tmp=(int) (pow(2.0,(1.0-par)*7.2));
- gain=1.0;
- if (i==(int) (tmp)) gain=pow(2.0,par2*par2*8.0);
- break;
- };
-
-
- oscilFFTfreqs.s[i]*=gain;
- oscilFFTfreqs.c[i]*=gain;
- REALTYPE tmp=oscilFFTfreqs.s[i]*oscilFFTfreqs.s[i]+
- oscilFFTfreqs.c[i]*oscilFFTfreqs.c[i];
- if (max<tmp) max=tmp;
+ for (int i=1;i<OSCIL_SIZE/2;i++) {
+ REALTYPE gain=1.0;
+ switch (Pfiltertype) {
+ case 1:
+ gain=pow(1.0-par*par*par*0.99,i);//lp
+ tmp=par2*par2*par2*par2*0.5+0.0001;
+ if (gain<tmp) gain=pow(gain,10.0)/pow(tmp,9.0);
+ break;
+ case 2:
+ gain=1.0-pow(1.0-par*par,i+1);//hp1
+ gain=pow(gain,par2*2.0+0.1);
+ break;
+ case 3:
+ if (par<0.2) par=par*0.25+0.15;
+ gain=1.0-pow(1.0-par*par*0.999+0.001,i*0.05*i+1.0);//hp1b
+ tmp=pow(5.0,par2*2.0);
+ gain=pow(gain,tmp);
+ break;
+ case 4:
+ gain=i+1-pow(2,(1.0-par)*7.5);//bp1
+ gain=1.0/(1.0+gain*gain/(i+1.0));
+ tmp=pow(5.0,par2*2.0);
+ gain=pow(gain,tmp);
+ if (gain<1e-5) gain=1e-5;
+ break;
+ case 5:
+ gain=i+1-pow(2,(1.0-par)*7.5);//bs1
+ gain=pow(atan(gain/(i/10.0+1))/1.57,6);
+ gain=pow(gain,par2*par2*3.9+0.1);
+ break;
+ case 6:
+ tmp=pow(par2,0.33);
+ gain=(i+1>pow(2,(1.0-par)*10)?0.0:1.0)*par2+(1.0-par2);//lp2
+ break;
+ case 7:
+ tmp=pow(par2,0.33);
+ //tmp=1.0-(1.0-par2)*(1.0-par2);
+ gain=(i+1>pow(2,(1.0-par)*7)?1.0:0.0)*par2+(1.0-par2);//hp2
+ if (Pfilterpar1==0) gain=1.0;
+ break;
+ case 8:
+ tmp=pow(par2,0.33);
+ //tmp=1.0-(1.0-par2)*(1.0-par2);
+ gain=(fabs(pow(2,(1.0-par)*7)-i)>i/2+1?0.0:1.0)*par2+(1.0-par2);//bp2
+ break;
+ case 9:
+ tmp=pow(par2,0.33);
+ gain=(fabs(pow(2,(1.0-par)*7)-i)<i/2+1?0.0:1.0)*par2+(1.0-par2);//bs2
+ break;
+ case 10:
+ tmp=pow(5.0,par2*2.0-1.0);
+ tmp=pow(i/32.0,tmp)*32.0;
+ if (Pfilterpar2==64) tmp=i;
+ gain=cos(par*par*PI/2.0*tmp);//cos
+ gain*=gain;
+ break;
+ case 11:
+ tmp=pow(5.0,par2*2.0-1.0);
+ tmp=pow(i/32.0,tmp)*32.0;
+ if (Pfilterpar2==64) tmp=i;
+ gain=sin(par*par*PI/2.0*tmp);//sin
+ gain*=gain;
+ break;
+ case 12:
+ p2=1.0-par+0.2;
+ x=i/(64.0*p2*p2);
+ if (x<0.0) x=0.0;
+ else if (x>1.0) x=1.0;
+ tmp=pow(1.0-par2,2.0);
+ gain=cos(x*PI)*(1.0-tmp)+1.01+tmp;//low shelf
+ break;
+ case 13:
+ tmp=(int) (pow(2.0,(1.0-par)*7.2));
+ gain=1.0;
+ if (i==(int) (tmp)) gain=pow(2.0,par2*par2*8.0);
+ break;
+ };
+
+
+ oscilFFTfreqs.s[i]*=gain;
+ oscilFFTfreqs.c[i]*=gain;
+ REALTYPE tmp=oscilFFTfreqs.s[i]*oscilFFTfreqs.s[i]+
+ oscilFFTfreqs.c[i]*oscilFFTfreqs.c[i];
+ if (max<tmp) max=tmp;
};
max=sqrt(max);
if (max<1e-10) max=1.0;
REALTYPE imax=1.0/max;
for (int i=1;i<OSCIL_SIZE/2;i++) {
- oscilFFTfreqs.s[i]*=imax;
- oscilFFTfreqs.c[i]*=imax;
+ oscilFFTfreqs.s[i]*=imax;
+ oscilFFTfreqs.c[i]*=imax;
};
};
-
-/*
+
+/*
* Change the base function
*/
-void OscilGen::changebasefunction(){
+void OscilGen::changebasefunction()
+{
if (Pcurrentbasefunc!=0) {
getbasefunction(tmpsmps);
- fft->smps2freqs(tmpsmps,basefuncFFTfreqs);
- basefuncFFTfreqs.c[0]=0.0;
+ fft->smps2freqs(tmpsmps,basefuncFFTfreqs);
+ basefuncFFTfreqs.c[0]=0.0;
} else {
- for (int i=0;i<OSCIL_SIZE/2;i++){
- basefuncFFTfreqs.s[i]=0.0;
- basefuncFFTfreqs.c[i]=0.0;
- };
- //in this case basefuncFFTfreqs_ are not used
+ for (int i=0;i<OSCIL_SIZE/2;i++) {
+ basefuncFFTfreqs.s[i]=0.0;
+ basefuncFFTfreqs.c[i]=0.0;
+ };
+ //in this case basefuncFFTfreqs_ are not used
}
oscilprepared=0;
oldbasefunc=Pcurrentbasefunc;
@@ -441,10 +508,11 @@ void OscilGen::changebasefunction(){
oldbasefuncmodulationpar3=Pbasefuncmodulationpar3;
};
-/*
+/*
* Waveshape
*/
-void OscilGen::waveshape(){
+void OscilGen::waveshape()
+{
int i;
oldwaveshapingfunction=Pwaveshapingfunction;
@@ -454,30 +522,32 @@ void OscilGen::waveshape(){
oscilFFTfreqs.c[0]=0.0;//remove the DC
//reduce the amplitude of the freqs near the nyquist
for (i=1;i<OSCIL_SIZE/8;i++) {
- REALTYPE tmp=i/(OSCIL_SIZE/8.0);
- oscilFFTfreqs.s[OSCIL_SIZE/2-i]*=tmp;
- oscilFFTfreqs.c[OSCIL_SIZE/2-i]*=tmp;
+ REALTYPE tmp=i/(OSCIL_SIZE/8.0);
+ oscilFFTfreqs.s[OSCIL_SIZE/2-i]*=tmp;
+ oscilFFTfreqs.c[OSCIL_SIZE/2-i]*=tmp;
};
- fft->freqs2smps(oscilFFTfreqs,tmpsmps);
+ fft->freqs2smps(oscilFFTfreqs,tmpsmps);
//Normalize
REALTYPE max=0.0;
- for (i=0;i<OSCIL_SIZE;i++)
- if (max<fabs(tmpsmps[i])) max=fabs(tmpsmps[i]);
+ for (i=0;i<OSCIL_SIZE;i++)
+ if (max<fabs(tmpsmps[i])) max=fabs(tmpsmps[i]);
if (max<0.00001) max=1.0;
- max=1.0/max;for (i=0;i<OSCIL_SIZE;i++) tmpsmps[i]*=max;
-
+ max=1.0/max;
+ for (i=0;i<OSCIL_SIZE;i++) tmpsmps[i]*=max;
+
//Do the waveshaping
waveshapesmps(OSCIL_SIZE,tmpsmps,Pwaveshapingfunction,Pwaveshaping);
-
+
fft->smps2freqs(tmpsmps,oscilFFTfreqs);//perform FFT
};
-/*
+/*
* Do the Frequency Modulation of the Oscil
*/
-void OscilGen::modulation(){
+void OscilGen::modulation()
+{
int i;
oldmodulation=Pmodulation;
@@ -488,28 +558,31 @@ void OscilGen::modulation(){
REALTYPE modulationpar1=Pmodulationpar1/127.0,
- modulationpar2=0.5-Pmodulationpar2/127.0,
- modulationpar3=Pmodulationpar3/127.0;
-
- switch(Pmodulation){
- case 1:modulationpar1=(pow(2,modulationpar1*7.0)-1.0)/100.0;
- modulationpar3=floor((pow(2,modulationpar3*5.0)-1.0));
- if (modulationpar3<0.9999) modulationpar3=-1.0;
- break;
- case 2:modulationpar1=(pow(2,modulationpar1*7.0)-1.0)/100.0;
- modulationpar3=1.0+floor((pow(2,modulationpar3*5.0)-1.0));
- break;
- case 3:modulationpar1=(pow(2,modulationpar1*9.0)-1.0)/100.0;
- modulationpar3=0.01+(pow(2,modulationpar3*16.0)-1.0)/10.0;
- break;
- };
+ modulationpar2=0.5-Pmodulationpar2/127.0,
+ modulationpar3=Pmodulationpar3/127.0;
+
+ switch (Pmodulation) {
+ case 1:
+ modulationpar1=(pow(2,modulationpar1*7.0)-1.0)/100.0;
+ modulationpar3=floor((pow(2,modulationpar3*5.0)-1.0));
+ if (modulationpar3<0.9999) modulationpar3=-1.0;
+ break;
+ case 2:
+ modulationpar1=(pow(2,modulationpar1*7.0)-1.0)/100.0;
+ modulationpar3=1.0+floor((pow(2,modulationpar3*5.0)-1.0));
+ break;
+ case 3:
+ modulationpar1=(pow(2,modulationpar1*9.0)-1.0)/100.0;
+ modulationpar3=0.01+(pow(2,modulationpar3*16.0)-1.0)/10.0;
+ break;
+ };
oscilFFTfreqs.c[0]=0.0;//remove the DC
//reduce the amplitude of the freqs near the nyquist
for (i=1;i<OSCIL_SIZE/8;i++) {
- REALTYPE tmp=i/(OSCIL_SIZE/8.0);
- oscilFFTfreqs.s[OSCIL_SIZE/2-i]*=tmp;
- oscilFFTfreqs.c[OSCIL_SIZE/2-i]*=tmp;
+ REALTYPE tmp=i/(OSCIL_SIZE/8.0);
+ oscilFFTfreqs.s[OSCIL_SIZE/2-i]*=tmp;
+ oscilFFTfreqs.c[OSCIL_SIZE/2-i]*=tmp;
};
fft->freqs2smps(oscilFFTfreqs,tmpsmps);
int extra_points=2;
@@ -519,28 +592,32 @@ void OscilGen::modulation(){
REALTYPE max=0.0;
for (i=0;i<OSCIL_SIZE;i++) if (max<fabs(tmpsmps[i])) max=fabs(tmpsmps[i]);
if (max<0.00001) max=1.0;
- max=1.0/max;for (i=0;i<OSCIL_SIZE;i++) in[i]=tmpsmps[i]*max;
+ max=1.0/max;
+ for (i=0;i<OSCIL_SIZE;i++) in[i]=tmpsmps[i]*max;
for (i=0;i<extra_points;i++) in[i+OSCIL_SIZE]=tmpsmps[i]*max;
-
+
//Do the modulation
for (i=0;i<OSCIL_SIZE;i++) {
- REALTYPE t=i*1.0/OSCIL_SIZE;
-
- switch(Pmodulation){
- case 1:t=t*modulationpar3+sin((t+modulationpar2)*2.0*PI)*modulationpar1;//rev
- break;
- case 2:t=t+sin((t*modulationpar3+modulationpar2)*2.0*PI)*modulationpar1;//sine
- break;
- case 3:t=t+pow((1.0-cos((t+modulationpar2)*2.0*PI))*0.5,modulationpar3)*modulationpar1;//power
- break;
- };
-
- t=(t-floor(t))*OSCIL_SIZE;
-
- int poshi=(int) t;
- REALTYPE poslo=t-floor(t);
-
- tmpsmps[i]=in[poshi]*(1.0-poslo)+in[poshi+1]*poslo;
+ REALTYPE t=i*1.0/OSCIL_SIZE;
+
+ switch (Pmodulation) {
+ case 1:
+ t=t*modulationpar3+sin((t+modulationpar2)*2.0*PI)*modulationpar1;//rev
+ break;
+ case 2:
+ t=t+sin((t*modulationpar3+modulationpar2)*2.0*PI)*modulationpar1;//sine
+ break;
+ case 3:
+ t=t+pow((1.0-cos((t+modulationpar2)*2.0*PI))*0.5,modulationpar3)*modulationpar1;//power
+ break;
+ };
+
+ t=(t-floor(t))*OSCIL_SIZE;
+
+ int poshi=(int) t;
+ REALTYPE poslo=t-floor(t);
+
+ tmpsmps[i]=in[poshi]*(1.0-poslo)+in[poshi+1]*poslo;
};
delete [] in;
@@ -549,267 +626,291 @@ void OscilGen::modulation(){
-/*
+/*
* Adjust the spectrum
*/
-void OscilGen::spectrumadjust(){
+void OscilGen::spectrumadjust()
+{
if (Psatype==0) return;
REALTYPE par=Psapar/127.0;
- switch(Psatype){
- case 1: par=1.0-par*2.0;
- if (par>=0.0) par=pow(5.0,par);
- else par=pow(8.0,par);
- break;
- case 2: par=pow(10.0,(1.0-par)*3.0)*0.25;
- break;
- case 3: par=pow(10.0,(1.0-par)*3.0)*0.25;
- break;
+ switch (Psatype) {
+ case 1:
+ par=1.0-par*2.0;
+ if (par>=0.0) par=pow(5.0,par);
+ else par=pow(8.0,par);
+ break;
+ case 2:
+ par=pow(10.0,(1.0-par)*3.0)*0.25;
+ break;
+ case 3:
+ par=pow(10.0,(1.0-par)*3.0)*0.25;
+ break;
};
REALTYPE max=0.0;
- for (int i=0;i<OSCIL_SIZE/2;i++){
- REALTYPE tmp=pow(oscilFFTfreqs.c[i],2)+pow(oscilFFTfreqs.s[i],2.0);
- if (max<tmp) max=tmp;
+ for (int i=0;i<OSCIL_SIZE/2;i++) {
+ REALTYPE tmp=pow(oscilFFTfreqs.c[i],2)+pow(oscilFFTfreqs.s[i],2.0);
+ if (max<tmp) max=tmp;
};
max=sqrt(max)/OSCIL_SIZE*2.0;
if (max<1e-8) max=1.0;
-
- for (int i=0;i<OSCIL_SIZE/2;i++){
+
+ for (int i=0;i<OSCIL_SIZE/2;i++) {
REALTYPE mag=sqrt(pow(oscilFFTfreqs.s[i],2)+pow(oscilFFTfreqs.c[i],2.0))/max;
- REALTYPE phase=atan2(oscilFFTfreqs.s[i],oscilFFTfreqs.c[i]);
-
- switch (Psatype){
- case 1: mag=pow(mag,par);
- break;
- case 2: if (mag<par) mag=0.0;
- break;
- case 3: mag/=par;
- if (mag>1.0) mag=1.0;
- break;
- };
- oscilFFTfreqs.c[i]=mag*cos(phase);
- oscilFFTfreqs.s[i]=mag*sin(phase);
+ REALTYPE phase=atan2(oscilFFTfreqs.s[i],oscilFFTfreqs.c[i]);
+
+ switch (Psatype) {
+ case 1:
+ mag=pow(mag,par);
+ break;
+ case 2:
+ if (mag<par) mag=0.0;
+ break;
+ case 3:
+ mag/=par;
+ if (mag>1.0) mag=1.0;
+ break;
+ };
+ oscilFFTfreqs.c[i]=mag*cos(phase);
+ oscilFFTfreqs.s[i]=mag*sin(phase);
};
-
+
};
-void OscilGen::shiftharmonics(){
+void OscilGen::shiftharmonics()
+{
if (Pharmonicshift==0) return;
-
+
REALTYPE hc,hs;
int harmonicshift=-Pharmonicshift;
-
- if (harmonicshift>0){
- for (int i=OSCIL_SIZE/2-2;i>=0;i--){
- int oldh=i-harmonicshift;
- if (oldh<0){
- hc=0.0;
- hs=0.0;
- } else {
- hc=oscilFFTfreqs.c[oldh+1];
- hs=oscilFFTfreqs.s[oldh+1];
- };
- oscilFFTfreqs.c[i+1]=hc;
- oscilFFTfreqs.s[i+1]=hs;
- };
+
+ if (harmonicshift>0) {
+ for (int i=OSCIL_SIZE/2-2;i>=0;i--) {
+ int oldh=i-harmonicshift;
+ if (oldh<0) {
+ hc=0.0;
+ hs=0.0;
+ } else {
+ hc=oscilFFTfreqs.c[oldh+1];
+ hs=oscilFFTfreqs.s[oldh+1];
+ };
+ oscilFFTfreqs.c[i+1]=hc;
+ oscilFFTfreqs.s[i+1]=hs;
+ };
} else {
- for (int i=0;i<OSCIL_SIZE/2-1;i++){
- int oldh=i+abs(harmonicshift);
- if (oldh>=(OSCIL_SIZE/2-1)){
- hc=0.0;
- hs=0.0;
- } else {
- hc=oscilFFTfreqs.c[oldh+1];
- hs=oscilFFTfreqs.s[oldh+1];
- if (fabs(hc)<0.000001) hc=0.0;
- if (fabs(hs)<0.000001) hs=0.0;
- };
-
- oscilFFTfreqs.c[i+1]=hc;
- oscilFFTfreqs.s[i+1]=hs;
- };
+ for (int i=0;i<OSCIL_SIZE/2-1;i++) {
+ int oldh=i+abs(harmonicshift);
+ if (oldh>=(OSCIL_SIZE/2-1)) {
+ hc=0.0;
+ hs=0.0;
+ } else {
+ hc=oscilFFTfreqs.c[oldh+1];
+ hs=oscilFFTfreqs.s[oldh+1];
+ if (fabs(hc)<0.000001) hc=0.0;
+ if (fabs(hs)<0.000001) hs=0.0;
+ };
+
+ oscilFFTfreqs.c[i+1]=hc;
+ oscilFFTfreqs.s[i+1]=hs;
+ };
};
-
+
oscilFFTfreqs.c[0]=0.0;
};
-/*
+/*
* Prepare the Oscillator
*/
-void OscilGen::prepare(){
- int i,j,k;
- REALTYPE a,b,c,d,hmagnew;
-
- if ((oldbasepar!=Pbasefuncpar)||(oldbasefunc!=Pcurrentbasefunc)||
- (oldbasefuncmodulation!=Pbasefuncmodulation)||
- (oldbasefuncmodulationpar1!=Pbasefuncmodulationpar1)||
- (oldbasefuncmodulationpar2!=Pbasefuncmodulationpar2)||
- (oldbasefuncmodulationpar3!=Pbasefuncmodulationpar3))
- changebasefunction();
-
- for (i=0;i<MAX_AD_HARMONICS;i++) hphase[i]=(Phphase[i]-64.0)/64.0*PI/(i+1);
-
- for (i=0;i<MAX_AD_HARMONICS;i++){
- hmagnew=1.0-fabs(Phmag[i]/64.0-1.0);
- switch(Phmagtype){
- case 1:hmag[i]=exp(hmagnew*log(0.01)); break;
- case 2:hmag[i]=exp(hmagnew*log(0.001));break;
- case 3:hmag[i]=exp(hmagnew*log(0.0001));break;
- case 4:hmag[i]=exp(hmagnew*log(0.00001));break;
- default:hmag[i]=1.0-hmagnew;
- break;
- };
-
- if (Phmag[i]<64) hmag[i]=-hmag[i];
- };
-
- //remove the harmonics where Phmag[i]==64
- for (i=0;i<MAX_AD_HARMONICS;i++) if (Phmag[i]==64) hmag[i]=0.0;
-
-
- for (i=0;i<OSCIL_SIZE/2;i++) {
- oscilFFTfreqs.c[i]=0.0;
- oscilFFTfreqs.s[i]=0.0;
- };
- if (Pcurrentbasefunc==0) {//the sine case
- for (i=0;i<MAX_AD_HARMONICS;i++){
- oscilFFTfreqs.c[i+1]=-hmag[i]*sin(hphase[i]*(i+1))/2.0;
- oscilFFTfreqs.s[i+1]=hmag[i]*cos(hphase[i]*(i+1))/2.0;
- };
- } else {
- for (j=0;j<MAX_AD_HARMONICS;j++){
- if (Phmag[j]==64) continue;
- for (i=1;i<OSCIL_SIZE/2;i++){
- k=i*(j+1);if (k>=OSCIL_SIZE/2) break;
- a=basefuncFFTfreqs.c[i];
- b=basefuncFFTfreqs.s[i];
- c=hmag[j]*cos(hphase[j]*k);
- d=hmag[j]*sin(hphase[j]*k);
- oscilFFTfreqs.c[k]+=a*c-b*d;
- oscilFFTfreqs.s[k]+=a*d+b*c;
- };
- };
-
- };
-
- if (Pharmonicshiftfirst!=0) shiftharmonics();
-
-
-
- if (Pfilterbeforews==0){
+void OscilGen::prepare()
+{
+ int i,j,k;
+ REALTYPE a,b,c,d,hmagnew;
+
+ if ((oldbasepar!=Pbasefuncpar)||(oldbasefunc!=Pcurrentbasefunc)||
+ (oldbasefuncmodulation!=Pbasefuncmodulation)||
+ (oldbasefuncmodulationpar1!=Pbasefuncmodulationpar1)||
+ (oldbasefuncmodulationpar2!=Pbasefuncmodulationpar2)||
+ (oldbasefuncmodulationpar3!=Pbasefuncmodulationpar3))
+ changebasefunction();
+
+ for (i=0;i<MAX_AD_HARMONICS;i++) hphase[i]=(Phphase[i]-64.0)/64.0*PI/(i+1);
+
+ for (i=0;i<MAX_AD_HARMONICS;i++) {
+ hmagnew=1.0-fabs(Phmag[i]/64.0-1.0);
+ switch (Phmagtype) {
+ case 1:
+ hmag[i]=exp(hmagnew*log(0.01));
+ break;
+ case 2:
+ hmag[i]=exp(hmagnew*log(0.001));
+ break;
+ case 3:
+ hmag[i]=exp(hmagnew*log(0.0001));
+ break;
+ case 4:
+ hmag[i]=exp(hmagnew*log(0.00001));
+ break;
+ default:
+ hmag[i]=1.0-hmagnew;
+ break;
+ };
+
+ if (Phmag[i]<64) hmag[i]=-hmag[i];
+ };
+
+ //remove the harmonics where Phmag[i]==64
+ for (i=0;i<MAX_AD_HARMONICS;i++) if (Phmag[i]==64) hmag[i]=0.0;
+
+
+ for (i=0;i<OSCIL_SIZE/2;i++) {
+ oscilFFTfreqs.c[i]=0.0;
+ oscilFFTfreqs.s[i]=0.0;
+ };
+ if (Pcurrentbasefunc==0) {//the sine case
+ for (i=0;i<MAX_AD_HARMONICS;i++) {
+ oscilFFTfreqs.c[i+1]=-hmag[i]*sin(hphase[i]*(i+1))/2.0;
+ oscilFFTfreqs.s[i+1]=hmag[i]*cos(hphase[i]*(i+1))/2.0;
+ };
+ } else {
+ for (j=0;j<MAX_AD_HARMONICS;j++) {
+ if (Phmag[j]==64) continue;
+ for (i=1;i<OSCIL_SIZE/2;i++) {
+ k=i*(j+1);
+ if (k>=OSCIL_SIZE/2) break;
+ a=basefuncFFTfreqs.c[i];
+ b=basefuncFFTfreqs.s[i];
+ c=hmag[j]*cos(hphase[j]*k);
+ d=hmag[j]*sin(hphase[j]*k);
+ oscilFFTfreqs.c[k]+=a*c-b*d;
+ oscilFFTfreqs.s[k]+=a*d+b*c;
+ };
+ };
+
+ };
+
+ if (Pharmonicshiftfirst!=0) shiftharmonics();
+
+
+
+ if (Pfilterbeforews==0) {
waveshape();
- oscilfilter();
+ oscilfilter();
} else {
- oscilfilter();
+ oscilfilter();
waveshape();
};
- modulation();
- spectrumadjust();
- if (Pharmonicshiftfirst==0) shiftharmonics();
+ modulation();
+ spectrumadjust();
+ if (Pharmonicshiftfirst==0) shiftharmonics();
- oscilFFTfreqs.c[0]=0.0;
+ oscilFFTfreqs.c[0]=0.0;
- oldhmagtype=Phmagtype;
- oldharmonicshift=Pharmonicshift+Pharmonicshiftfirst*256;
+ oldhmagtype=Phmagtype;
+ oldharmonicshift=Pharmonicshift+Pharmonicshiftfirst*256;
- oscilprepared=1;
+ oscilprepared=1;
};
-void OscilGen::adaptiveharmonic(FFTFREQS f,REALTYPE freq){
- if ((Padaptiveharmonics==0)/*||(freq<1.0)*/) return;
- if (freq<1.0) freq=440.0;
-
- FFTFREQS inf;
- newFFTFREQS(&inf,OSCIL_SIZE/2);
- for (int i=0;i<OSCIL_SIZE/2;i++) {
- inf.s[i]=f.s[i];
- inf.c[i]=f.c[i];
- f.s[i]=0.0;
- f.c[i]=0.0;
- };
- inf.c[0]=0.0;inf.s[0]=0.0;
-
- REALTYPE hc=0.0,hs=0.0;
- REALTYPE basefreq=30.0*pow(10.0,Padaptiveharmonicsbasefreq/128.0);
- REALTYPE power=(Padaptiveharmonicspower+1.0)/101.0;
-
- REALTYPE rap=freq/basefreq;
-
- rap=pow(rap,power);
-
- bool down=false;
- if (rap>1.0) {
- rap=1.0/rap;
- down=true;
- };
-
- for (int i=0;i<OSCIL_SIZE/2-2;i++){
- REALTYPE h=i*rap;
- int high=(int)(i*rap);
- REALTYPE low=fmod(h,1.0);
-
- if (high>=(OSCIL_SIZE/2-2)){
- break;
- } else {
- if (down){
- f.c[high]+=inf.c[i]*(1.0-low);
- f.s[high]+=inf.s[i]*(1.0-low);
- f.c[high+1]+=inf.c[i]*low;
- f.s[high+1]+=inf.s[i]*low;
- } else {
- hc=inf.c[high]*(1.0-low)+inf.c[high+1]*low;
- hs=inf.s[high]*(1.0-low)+inf.s[high+1]*low;
- };
- if (fabs(hc)<0.000001) hc=0.0;
- if (fabs(hs)<0.000001) hs=0.0;
- };
-
- if (!down){
- if (i==0) {//corect the aplitude of the first harmonic
- hc*=rap;
- hs*=rap;
- };
- f.c[i]=hc;
- f.s[i]=hs;
- };
- };
-
- f.c[1]+=f.c[0];f.s[1]+=f.s[0];
- f.c[0]=0.0;f.s[0]=0.0;
- deleteFFTFREQS(&inf);
+void OscilGen::adaptiveharmonic(FFTFREQS f,REALTYPE freq)
+{
+ if ((Padaptiveharmonics==0)/*||(freq<1.0)*/) return;
+ if (freq<1.0) freq=440.0;
+
+ FFTFREQS inf;
+ newFFTFREQS(&inf,OSCIL_SIZE/2);
+ for (int i=0;i<OSCIL_SIZE/2;i++) {
+ inf.s[i]=f.s[i];
+ inf.c[i]=f.c[i];
+ f.s[i]=0.0;
+ f.c[i]=0.0;
+ };
+ inf.c[0]=0.0;
+ inf.s[0]=0.0;
+
+ REALTYPE hc=0.0,hs=0.0;
+ REALTYPE basefreq=30.0*pow(10.0,Padaptiveharmonicsbasefreq/128.0);
+ REALTYPE power=(Padaptiveharmonicspower+1.0)/101.0;
+
+ REALTYPE rap=freq/basefreq;
+
+ rap=pow(rap,power);
+
+ bool down=false;
+ if (rap>1.0) {
+ rap=1.0/rap;
+ down=true;
+ };
+
+ for (int i=0;i<OSCIL_SIZE/2-2;i++) {
+ REALTYPE h=i*rap;
+ int high=(int)(i*rap);
+ REALTYPE low=fmod(h,1.0);
+
+ if (high>=(OSCIL_SIZE/2-2)) {
+ break;
+ } else {
+ if (down) {
+ f.c[high]+=inf.c[i]*(1.0-low);
+ f.s[high]+=inf.s[i]*(1.0-low);
+ f.c[high+1]+=inf.c[i]*low;
+ f.s[high+1]+=inf.s[i]*low;
+ } else {
+ hc=inf.c[high]*(1.0-low)+inf.c[high+1]*low;
+ hs=inf.s[high]*(1.0-low)+inf.s[high+1]*low;
+ };
+ if (fabs(hc)<0.000001) hc=0.0;
+ if (fabs(hs)<0.000001) hs=0.0;
+ };
+
+ if (!down) {
+ if (i==0) {//corect the aplitude of the first harmonic
+ hc*=rap;
+ hs*=rap;
+ };
+ f.c[i]=hc;
+ f.s[i]=hs;
+ };
+ };
+
+ f.c[1]+=f.c[0];
+ f.s[1]+=f.s[0];
+ f.c[0]=0.0;
+ f.s[0]=0.0;
+ deleteFFTFREQS(&inf);
};
-void OscilGen::adaptiveharmonicpostprocess(REALTYPE *f,int size){
+void OscilGen::adaptiveharmonicpostprocess(REALTYPE *f,int size)
+{
if (Padaptiveharmonics<=1) return;
REALTYPE *inf=new REALTYPE[size];
REALTYPE par=Padaptiveharmonicspar*0.01;
par=1.0-pow((1.0-par),1.5);
-
+
for (int i=0;i<size;i++) {
- inf[i]=f[i]*par;
- f[i]=f[i]*(1.0-par);
+ inf[i]=f[i]*par;
+ f[i]=f[i]*(1.0-par);
};
-
- if (Padaptiveharmonics==2){//2n+1
+
+ if (Padaptiveharmonics==2) {//2n+1
for (int i=0;i<size;i++) if ((i%2)==0) f[i]+=inf[i];//i=0 pt prima armonica,etc.
- } else{//celelalte moduri
+ } else {//celelalte moduri
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++) {
- if (((i+1)%nh)==0){
- f[i]+=inf[i];
- };
- };
+ if (sub_vs_add==0) {
+ for (int i=0;i<size;i++) {
+ if (((i+1)%nh)==0) {
+ f[i]+=inf[i];
+ };
+ };
} else {
- for (int i=0;i<size/nh-1;i++) {
- f[(i+1)*nh-1]+=inf[i];
- };
- };
+ for (int i=0;i<size/nh-1;i++) {
+ f[(i+1)*nh-1]+=inf[i];
+ };
+ };
};
delete(inf);
@@ -817,67 +918,70 @@ void OscilGen::adaptiveharmonicpostprocess(REALTYPE *f,int size){
-/*
+/*
* Get the oscillator function
*/
-short int OscilGen::get(REALTYPE *smps,REALTYPE freqHz){
+short int OscilGen::get(REALTYPE *smps,REALTYPE freqHz)
+{
return(this->get(smps,freqHz,0));
};
-void OscilGen::newrandseed(unsigned int randseed){
+void OscilGen::newrandseed(unsigned int randseed)
+{
this->randseed=randseed;
};
-/*
+/*
* Get the oscillator function
*/
-short int OscilGen::get(REALTYPE *smps,REALTYPE freqHz,int resonance){
+short int OscilGen::get(REALTYPE *smps,REALTYPE freqHz,int resonance)
+{
int i;
int nyquist,outpos;
-
+
if ((oldbasepar!=Pbasefuncpar)||(oldbasefunc!=Pcurrentbasefunc)||(oldhmagtype!=Phmagtype)
- ||(oldwaveshaping!=Pwaveshaping)||(oldwaveshapingfunction!=Pwaveshapingfunction)) oscilprepared=0;
- if (oldfilterpars!=Pfiltertype*256+Pfilterpar1+Pfilterpar2*65536+Pfilterbeforews*16777216){
- oscilprepared=0;
- oldfilterpars=Pfiltertype*256+Pfilterpar1+Pfilterpar2*65536+Pfilterbeforews*16777216;
+ ||(oldwaveshaping!=Pwaveshaping)||(oldwaveshapingfunction!=Pwaveshapingfunction)) oscilprepared=0;
+ if (oldfilterpars!=Pfiltertype*256+Pfilterpar1+Pfilterpar2*65536+Pfilterbeforews*16777216) {
+ oscilprepared=0;
+ oldfilterpars=Pfiltertype*256+Pfilterpar1+Pfilterpar2*65536+Pfilterbeforews*16777216;
};
- if (oldsapars!=Psatype*256+Psapar){
- oscilprepared=0;
- oldsapars=Psatype*256+Psapar;
+ if (oldsapars!=Psatype*256+Psapar) {
+ oscilprepared=0;
+ oldsapars=Psatype*256+Psapar;
};
if ((oldbasefuncmodulation!=Pbasefuncmodulation)||
- (oldbasefuncmodulationpar1!=Pbasefuncmodulationpar1)||
- (oldbasefuncmodulationpar2!=Pbasefuncmodulationpar2)||
- (oldbasefuncmodulationpar3!=Pbasefuncmodulationpar3))
- oscilprepared=0;
+ (oldbasefuncmodulationpar1!=Pbasefuncmodulationpar1)||
+ (oldbasefuncmodulationpar2!=Pbasefuncmodulationpar2)||
+ (oldbasefuncmodulationpar3!=Pbasefuncmodulationpar3))
+ oscilprepared=0;
if ((oldmodulation!=Pmodulation)||
- (oldmodulationpar1!=Pmodulationpar1)||
- (oldmodulationpar2!=Pmodulationpar2)||
- (oldmodulationpar3!=Pmodulationpar3))
- oscilprepared=0;
+ (oldmodulationpar1!=Pmodulationpar1)||
+ (oldmodulationpar2!=Pmodulationpar2)||
+ (oldmodulationpar3!=Pmodulationpar3))
+ oscilprepared=0;
if (oldharmonicshift!=Pharmonicshift+Pharmonicshiftfirst*256) oscilprepared=0;
-
+
if (oscilprepared!=1) prepare();
outpos=(int)((RND*2.0-1.0)*(REALTYPE) OSCIL_SIZE*(Prand-64.0)/64.0);
outpos=(outpos+2*OSCIL_SIZE) % OSCIL_SIZE;
- for (i=0;i<OSCIL_SIZE/2;i++){
- outoscilFFTfreqs.c[i]=0.0;
- outoscilFFTfreqs.s[i]=0.0;
+ for (i=0;i<OSCIL_SIZE/2;i++) {
+ outoscilFFTfreqs.c[i]=0.0;
+ outoscilFFTfreqs.s[i]=0.0;
};
nyquist=(int)(0.5*SAMPLE_RATE/fabs(freqHz))+2;
if (ADvsPAD) nyquist=(int)(OSCIL_SIZE/2);
if (nyquist>OSCIL_SIZE/2) nyquist=OSCIL_SIZE/2;
-
-
+
+
int realnyquist=nyquist;
-
+
if (Padaptiveharmonics!=0) nyquist=OSCIL_SIZE/2;
for (i=1;i<nyquist-1;i++) {
outoscilFFTfreqs.c[i]=oscilFFTfreqs.c[i];
@@ -889,55 +993,57 @@ short int OscilGen::get(REALTYPE *smps,REALTYPE freqHz,int resonance){
adaptiveharmonicpostprocess(&outoscilFFTfreqs.s[1],OSCIL_SIZE/2-1);
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 (i=nyquist;i<OSCIL_SIZE/2;i++) {
- outoscilFFTfreqs.s[i]=0;
- outoscilFFTfreqs.c[i]=0;
- };
+ outoscilFFTfreqs.s[i]=0;
+ outoscilFFTfreqs.c[i]=0;
+ };
};
- // Randomness (each harmonic), the block type is computed
+ // Randomness (each harmonic), the block type is computed
// in ADnote by setting start position according to this setting
- if ((Prand>64)&&(freqHz>=0.0)&&(!ADvsPAD)){
+ if ((Prand>64)&&(freqHz>=0.0)&&(!ADvsPAD)) {
REALTYPE rnd,angle,a,b,c,d;
rnd=PI*pow((Prand-64.0)/64.0,2.0);
- for (i=1;i<nyquist-1;i++){//to Nyquist only for AntiAliasing
- angle=rnd*i*RND;
- a=outoscilFFTfreqs.c[i];
- b=outoscilFFTfreqs.s[i];
- c=cos(angle);
- d=sin(angle);
- outoscilFFTfreqs.c[i]=a*c-b*d;
- outoscilFFTfreqs.s[i]=a*d+b*c;
- };
+ for (i=1;i<nyquist-1;i++) {//to Nyquist only for AntiAliasing
+ angle=rnd*i*RND;
+ a=outoscilFFTfreqs.c[i];
+ b=outoscilFFTfreqs.s[i];
+ c=cos(angle);
+ d=sin(angle);
+ outoscilFFTfreqs.c[i]=a*c-b*d;
+ outoscilFFTfreqs.s[i]=a*d+b*c;
+ };
};
//Harmonic Amplitude Randomness
if ((freqHz>0.1)&&(!ADvsPAD)) {
- unsigned int realrnd=rand();
- srand(randseed);
- REALTYPE power=Pamprandpower/127.0;
- REALTYPE normalize=1.0/(1.2-power);
- switch (Pamprandtype){
- case 1: power=power*2.0-0.5;
- power=pow(15.0,power);
- for (i=1;i<nyquist-1;i++){
- REALTYPE amp=pow(RND,power)*normalize;
- outoscilFFTfreqs.c[i]*=amp;
- outoscilFFTfreqs.s[i]*=amp;
- };
- break;
- case 2: power=power*2.0-0.5;
- power=pow(15.0,power)*2.0;
- REALTYPE rndfreq=2*PI*RND;
- for (i=1;i<nyquist-1;i++){
- REALTYPE amp=pow(fabs(sin(i*rndfreq)),power)*normalize;
- outoscilFFTfreqs.c[i]*=amp;
- outoscilFFTfreqs.s[i]*=amp;
- };
- break;
- };
- srand(realrnd+1);
+ unsigned int realrnd=rand();
+ srand(randseed);
+ REALTYPE power=Pamprandpower/127.0;
+ REALTYPE normalize=1.0/(1.2-power);
+ switch (Pamprandtype) {
+ case 1:
+ power=power*2.0-0.5;
+ power=pow(15.0,power);
+ for (i=1;i<nyquist-1;i++) {
+ REALTYPE amp=pow(RND,power)*normalize;
+ outoscilFFTfreqs.c[i]*=amp;
+ outoscilFFTfreqs.s[i]*=amp;
+ };
+ break;
+ case 2:
+ power=power*2.0-0.5;
+ power=pow(15.0,power)*2.0;
+ REALTYPE rndfreq=2*PI*RND;
+ for (i=1;i<nyquist-1;i++) {
+ REALTYPE amp=pow(fabs(sin(i*rndfreq)),power)*normalize;
+ outoscilFFTfreqs.c[i]*=amp;
+ outoscilFFTfreqs.s[i]*=amp;
+ };
+ break;
+ };
+ srand(realrnd+1);
};
if ((freqHz>0.1)&&(resonance!=0)) res->applyres(nyquist-1,outoscilFFTfreqs,freqHz);
@@ -946,85 +1052,89 @@ short int OscilGen::get(REALTYPE *smps,REALTYPE freqHz,int resonance){
REALTYPE sum=0;
for (int j=1;j<OSCIL_SIZE/2;j++) {
REALTYPE term=outoscilFFTfreqs.c[j]*outoscilFFTfreqs.c[j]
- +outoscilFFTfreqs.s[j]*outoscilFFTfreqs.s[j];
+ +outoscilFFTfreqs.s[j]*outoscilFFTfreqs.s[j];
sum+=term;
};
- if (sum<0.000001) sum=1.0;
+ if (sum<0.000001) sum=1.0;
sum=1.0/sqrt(sum);
for (int j=1;j<OSCIL_SIZE/2;j++) {
- outoscilFFTfreqs.c[j]*=sum;
- outoscilFFTfreqs.s[j]*=sum;
+ outoscilFFTfreqs.c[j]*=sum;
+ outoscilFFTfreqs.s[j]*=sum;
};
-
- if ((ADvsPAD)&&(freqHz>0.1)){//in this case the smps will contain the freqs
+
+ if ((ADvsPAD)&&(freqHz>0.1)) {//in this case the smps will contain the freqs
for (i=1;i<OSCIL_SIZE/2;i++) smps[i-1]=sqrt(outoscilFFTfreqs.c[i]*outoscilFFTfreqs.c[i]
- +outoscilFFTfreqs.s[i]*outoscilFFTfreqs.s[i]);
+ +outoscilFFTfreqs.s[i]*outoscilFFTfreqs.s[i]);
} else {
- fft->freqs2smps(outoscilFFTfreqs,smps);
+ fft->freqs2smps(outoscilFFTfreqs,smps);
for (i=0;i<OSCIL_SIZE;i++) smps[i]*=0.25;//correct the amplitude
};
if (Prand<64) return(outpos);
- else return(0);
+ else return(0);
};
-/*
+/*
* Get the spectrum of the oscillator for the UI
*/
-void OscilGen::getspectrum(int n, REALTYPE *spc,int what){
+void OscilGen::getspectrum(int n, REALTYPE *spc,int what)
+{
if (n>OSCIL_SIZE/2) n=OSCIL_SIZE/2;
- for (int i=1;i<n;i++){
- if (what==0){
- spc[i-1]=sqrt(oscilFFTfreqs.c[i]*oscilFFTfreqs.c[i]
- +oscilFFTfreqs.s[i]*oscilFFTfreqs.s[i]);
- } else {
- if (Pcurrentbasefunc==0) spc[i-1]=((i==1)?(1.0):(0.0));
- else spc[i-1]=sqrt(basefuncFFTfreqs.c[i]*basefuncFFTfreqs.c[i]+
- basefuncFFTfreqs.s[i]*basefuncFFTfreqs.s[i]);
- };
+ for (int i=1;i<n;i++) {
+ if (what==0) {
+ spc[i-1]=sqrt(oscilFFTfreqs.c[i]*oscilFFTfreqs.c[i]
+ +oscilFFTfreqs.s[i]*oscilFFTfreqs.s[i]);
+ } else {
+ if (Pcurrentbasefunc==0) spc[i-1]=((i==1)?(1.0):(0.0));
+ else spc[i-1]=sqrt(basefuncFFTfreqs.c[i]*basefuncFFTfreqs.c[i]+
+ basefuncFFTfreqs.s[i]*basefuncFFTfreqs.s[i]);
+ };
};
-
+
if (what==0) {
for (int i=0;i<n;i++) outoscilFFTfreqs.s[i]=outoscilFFTfreqs.c[i]=spc[i];
- for (int i=n;i<OSCIL_SIZE/2;i++) outoscilFFTfreqs.s[i]=outoscilFFTfreqs.c[i]=0.0;
- adaptiveharmonic(outoscilFFTfreqs,0.0);
- for (int i=0;i<n;i++) spc[i]=outoscilFFTfreqs.s[i];
- adaptiveharmonicpostprocess(spc,n-1);
+ for (int i=n;i<OSCIL_SIZE/2;i++) outoscilFFTfreqs.s[i]=outoscilFFTfreqs.c[i]=0.0;
+ adaptiveharmonic(outoscilFFTfreqs,0.0);
+ for (int i=0;i<n;i++) spc[i]=outoscilFFTfreqs.s[i];
+ adaptiveharmonicpostprocess(spc,n-1);
};
};
-/*
+/*
* Convert the oscillator as base function
*/
-void OscilGen::useasbase(){
- int i;
+void OscilGen::useasbase()
+{
+ int i;
- for (i=0;i<OSCIL_SIZE/2;i++) {
- basefuncFFTfreqs.c[i]=oscilFFTfreqs.c[i];
- basefuncFFTfreqs.s[i]=oscilFFTfreqs.s[i];
- };
+ for (i=0;i<OSCIL_SIZE/2;i++) {
+ basefuncFFTfreqs.c[i]=oscilFFTfreqs.c[i];
+ basefuncFFTfreqs.s[i]=oscilFFTfreqs.s[i];
+ };
- oldbasefunc=Pcurrentbasefunc=127;
+ oldbasefunc=Pcurrentbasefunc=127;
- prepare();
+ prepare();
};
-/*
+/*
* Get the base function for UI
*/
-void OscilGen::getcurrentbasefunction(REALTYPE *smps){
+void OscilGen::getcurrentbasefunction(REALTYPE *smps)
+{
if (Pcurrentbasefunc!=0) {
- fft->freqs2smps(basefuncFFTfreqs,smps);
+ fft->freqs2smps(basefuncFFTfreqs,smps);
} else getbasefunction(smps);//the sine case
};
-void OscilGen::add2XML(XMLwrapper *xml){
+void OscilGen::add2XML(XMLwrapper *xml)
+{
xml->addpar("harmonic_mag_type",Phmagtype);
xml->addpar("base_function",Pcurrentbasefunc);
@@ -1062,41 +1172,42 @@ void OscilGen::add2XML(XMLwrapper *xml){
xml->addpar("adaptive_harmonics_power",Padaptiveharmonicspower);
xml->beginbranch("HARMONICS");
- for (int n=0;n<MAX_AD_HARMONICS;n++){
- if ((Phmag[n]==64)&&(Phphase[n]==64)) continue;
- xml->beginbranch("HARMONIC",n+1);
- xml->addpar("mag",Phmag[n]);
- xml->addpar("phase",Phphase[n]);
- xml->endbranch();
- };
+ for (int n=0;n<MAX_AD_HARMONICS;n++) {
+ if ((Phmag[n]==64)&&(Phphase[n]==64)) continue;
+ xml->beginbranch("HARMONIC",n+1);
+ xml->addpar("mag",Phmag[n]);
+ xml->addpar("phase",Phphase[n]);
+ xml->endbranch();
+ };
xml->endbranch();
-
- if (Pcurrentbasefunc==127){
- REALTYPE max=0.0;
-
- for (int i=0;i<OSCIL_SIZE/2;i++){
- if (max<fabs(basefuncFFTfreqs.c[i])) max=fabs(basefuncFFTfreqs.c[i]);
- if (max<fabs(basefuncFFTfreqs.s[i])) max=fabs(basefuncFFTfreqs.s[i]);
- };
- if (max<0.00000001) max=1.0;
-
- xml->beginbranch("BASE_FUNCTION");
- for (int i=1;i<OSCIL_SIZE/2;i++){
- REALTYPE xc=basefuncFFTfreqs.c[i]/max;
- REALTYPE xs=basefuncFFTfreqs.s[i]/max;
- if ((fabs(xs)>0.00001)&&(fabs(xs)>0.00001)){
- xml->beginbranch("BF_HARMONIC",i);
- xml->addparreal("cos",xc);
- xml->addparreal("sin",xs);
- xml->endbranch();
- };
- };
- xml->endbranch();
+
+ if (Pcurrentbasefunc==127) {
+ REALTYPE max=0.0;
+
+ for (int i=0;i<OSCIL_SIZE/2;i++) {
+ if (max<fabs(basefuncFFTfreqs.c[i])) max=fabs(basefuncFFTfreqs.c[i]);
+ if (max<fabs(basefuncFFTfreqs.s[i])) max=fabs(basefuncFFTfreqs.s[i]);
+ };
+ if (max<0.00000001) max=1.0;
+
+ xml->beginbranch("BASE_FUNCTION");
+ for (int i=1;i<OSCIL_SIZE/2;i++) {
+ REALTYPE xc=basefuncFFTfreqs.c[i]/max;
+ REALTYPE xs=basefuncFFTfreqs.s[i]/max;
+ if ((fabs(xs)>0.00001)&&(fabs(xs)>0.00001)) {
+ xml->beginbranch("BF_HARMONIC",i);
+ xml->addparreal("cos",xc);
+ xml->addparreal("sin",xs);
+ xml->endbranch();
+ };
+ };
+ xml->endbranch();
};
};
-void OscilGen::getfromXML(XMLwrapper *xml){
+void OscilGen::getfromXML(XMLwrapper *xml)
+{
Phmagtype=xml->getpar127("harmonic_mag_type",Phmagtype);
@@ -1136,45 +1247,46 @@ void OscilGen::getfromXML(XMLwrapper *xml){
Padaptiveharmonicspower=xml->getpar("adaptive_harmonics_power",Padaptiveharmonicspower,0,200);
- if (xml->enterbranch("HARMONICS")){
- Phmag[0]=64;Phphase[0]=64;
- for (int n=0;n<MAX_AD_HARMONICS;n++){
- if (xml->enterbranch("HARMONIC",n+1)==0) continue;
- Phmag[n]=xml->getpar127("mag",64);
- Phphase[n]=xml->getpar127("phase",64);
- xml->exitbranch();
- };
- xml->exitbranch();
+ if (xml->enterbranch("HARMONICS")) {
+ Phmag[0]=64;
+ Phphase[0]=64;
+ for (int n=0;n<MAX_AD_HARMONICS;n++) {
+ if (xml->enterbranch("HARMONIC",n+1)==0) continue;
+ Phmag[n]=xml->getpar127("mag",64);
+ Phphase[n]=xml->getpar127("phase",64);
+ xml->exitbranch();
+ };
+ xml->exitbranch();
};
-
+
if (Pcurrentbasefunc!=0) changebasefunction();
-
-
- if (xml->enterbranch("BASE_FUNCTION")){
- for (int i=1;i<OSCIL_SIZE/2;i++){
- if (xml->enterbranch("BF_HARMONIC",i)){
- basefuncFFTfreqs.c[i]=xml->getparreal("cos",0.0);
- basefuncFFTfreqs.s[i]=xml->getparreal("sin",0.0);
- xml->exitbranch();
- };
-
-
- };
- xml->exitbranch();
-
- REALTYPE max=0.0;
-
- basefuncFFTfreqs.c[0]=0.0;
- for (int i=0;i<OSCIL_SIZE/2;i++) {
- if (max<fabs(basefuncFFTfreqs.c[i])) max=fabs(basefuncFFTfreqs.c[i]);
- if (max<fabs(basefuncFFTfreqs.s[i])) max=fabs(basefuncFFTfreqs.s[i]);
- };
- if (max<0.00000001) max=1.0;
-
- for (int i=0;i<OSCIL_SIZE/2;i++) {
- if (basefuncFFTfreqs.c[i]) basefuncFFTfreqs.c[i]/=max;
- if (basefuncFFTfreqs.s[i]) basefuncFFTfreqs.s[i]/=max;
- };
+
+
+ if (xml->enterbranch("BASE_FUNCTION")) {
+ for (int i=1;i<OSCIL_SIZE/2;i++) {
+ if (xml->enterbranch("BF_HARMONIC",i)) {
+ basefuncFFTfreqs.c[i]=xml->getparreal("cos",0.0);
+ basefuncFFTfreqs.s[i]=xml->getparreal("sin",0.0);
+ xml->exitbranch();
+ };
+
+
+ };
+ xml->exitbranch();
+
+ REALTYPE max=0.0;
+
+ basefuncFFTfreqs.c[0]=0.0;
+ for (int i=0;i<OSCIL_SIZE/2;i++) {
+ if (max<fabs(basefuncFFTfreqs.c[i])) max=fabs(basefuncFFTfreqs.c[i]);
+ if (max<fabs(basefuncFFTfreqs.s[i])) max=fabs(basefuncFFTfreqs.s[i]);
+ };
+ if (max<0.00000001) max=1.0;
+
+ for (int i=0;i<OSCIL_SIZE/2;i++) {
+ if (basefuncFFTfreqs.c[i]) basefuncFFTfreqs.c[i]/=max;
+ if (basefuncFFTfreqs.s[i]) basefuncFFTfreqs.s[i]/=max;
+ };
};
};
diff --git a/src/Synth/OscilGen.h b/src/Synth/OscilGen.h
@@ -6,7 +6,7 @@
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,151 +26,152 @@
#include "../globals.h"
#include "../Misc/XMLwrapper.h"
#include "Resonance.h"
-#include "../DSP/FFTwrapper.h"
+#include "../DSP/FFTwrapper.h"
#include "../Params/Presets.h"
-class OscilGen:public Presets{
- public:
- OscilGen(FFTwrapper *fft_,Resonance *res_);
- ~OscilGen();
-
- /**computes the full spectrum of oscil from harmonics,phases and basefunc*/
- void prepare();
-
- /**do the antialiasing(cut off higher freqs.),apply randomness and do a IFFT*/
- short get(REALTYPE *smps,REALTYPE freqHz);//returns where should I start getting samples, used in block type randomness
- short get(REALTYPE *smps,REALTYPE freqHz,int resonance);
- //if freqHz is smaller than 0, return the "un-randomized" sample for UI
-
- void getbasefunction(REALTYPE *smps);
-
- //called by UI
- void getspectrum(int n,REALTYPE *spc,int what);//what=0 pt. oscil,1 pt. basefunc
- void getcurrentbasefunction(REALTYPE *smps);
- /**convert oscil to base function*/
- void useasbase();
-
- void add2XML(XMLwrapper *xml);
- void defaults();
- void getfromXML(XMLwrapper *xml);
-
- void convert2sine(int magtype);
-
- //Parameters
-
- /**
- * The hmag and hphase starts counting from 0, so the first harmonic(1) has the index 0,
- * 2-nd harmonic has index 1, ..the 128 harminic has index 127
- */
- unsigned char Phmag[MAX_AD_HARMONICS],Phphase[MAX_AD_HARMONICS];//the MIDI parameters for mag. and phases
-
-
- /**The Type of magnitude:
- * 0 - Linear
- * 1 - dB scale (-40)
- * 2 - dB scale (-60)
- * 3 - dB scale (-80)
- * 4 - dB scale (-100)*/
- unsigned char Phmagtype;
-
- unsigned char Pcurrentbasefunc;//The base function used - 0=sin, 1=...
- unsigned char Pbasefuncpar;//the parameter of the base function
-
- unsigned char Pbasefuncmodulation;//what modulation is applied to the basefunc
- unsigned char Pbasefuncmodulationpar1,Pbasefuncmodulationpar2,Pbasefuncmodulationpar3;//the parameter of the base function modulation
-
- /*the Randomness:
- 64=no randomness
- 63..0 - block type randomness - 0 is maximum
- 65..127 - each harmonic randomness - 127 is maximum*/
- unsigned char Prand;
- unsigned char Pwaveshaping,Pwaveshapingfunction;
- unsigned char Pfiltertype,Pfilterpar1,Pfilterpar2;
- unsigned char Pfilterbeforews;
- unsigned char Psatype,Psapar;//spectrum adjust
-
- unsigned char Pamprandpower, Pamprandtype;//amplitude randomness
- int Pharmonicshift;//how the harmonics are shifted
- int Pharmonicshiftfirst;//if the harmonic shift is done before waveshaping and filter
-
- unsigned char Padaptiveharmonics;//the adaptive harmonics status (off=0,on=1,etc..)
- unsigned char Padaptiveharmonicsbasefreq;//the base frequency of the adaptive harmonic (30..3000Hz)
- unsigned char Padaptiveharmonicspower;//the strength of the effect (0=off,100=full)
- unsigned char Padaptiveharmonicspar;//the parameters in 2,3,4.. modes of adaptive harmonics
-
- unsigned char Pmodulation;//what modulation is applied to the oscil
- unsigned char Pmodulationpar1,Pmodulationpar2,Pmodulationpar3;//the parameter of the parameters
-
-
- //makes a new random seed for Amplitude Randomness
- //this should be called every note on event
- void newrandseed(unsigned int randseed);
-
- bool ADvsPAD;//if it is used by ADsynth or by PADsynth
-
- static REALTYPE *tmpsmps;//this array stores some termporary data and it has SOUND_BUFFER_SIZE elements
- static FFTFREQS outoscilFFTfreqs;
-
- private:
-
- REALTYPE hmag[MAX_AD_HARMONICS],hphase[MAX_AD_HARMONICS];//the magnituides and the phases of the sine/nonsine harmonics
+class OscilGen:public Presets
+{
+public:
+ OscilGen(FFTwrapper *fft_,Resonance *res_);
+ ~OscilGen();
+
+ /**computes the full spectrum of oscil from harmonics,phases and basefunc*/
+ void prepare();
+
+ /**do the antialiasing(cut off higher freqs.),apply randomness and do a IFFT*/
+ short get(REALTYPE *smps,REALTYPE freqHz);//returns where should I start getting samples, used in block type randomness
+ short get(REALTYPE *smps,REALTYPE freqHz,int resonance);
+ //if freqHz is smaller than 0, return the "un-randomized" sample for UI
+
+ void getbasefunction(REALTYPE *smps);
+
+ //called by UI
+ void getspectrum(int n,REALTYPE *spc,int what);//what=0 pt. oscil,1 pt. basefunc
+ void getcurrentbasefunction(REALTYPE *smps);
+ /**convert oscil to base function*/
+ void useasbase();
+
+ void add2XML(XMLwrapper *xml);
+ void defaults();
+ void getfromXML(XMLwrapper *xml);
+
+ void convert2sine(int magtype);
+
+ //Parameters
+
+ /**
+ * The hmag and hphase starts counting from 0, so the first harmonic(1) has the index 0,
+ * 2-nd harmonic has index 1, ..the 128 harminic has index 127
+ */
+ unsigned char Phmag[MAX_AD_HARMONICS],Phphase[MAX_AD_HARMONICS];//the MIDI parameters for mag. and phases
+
+
+ /**The Type of magnitude:
+ * 0 - Linear
+ * 1 - dB scale (-40)
+ * 2 - dB scale (-60)
+ * 3 - dB scale (-80)
+ * 4 - dB scale (-100)*/
+ unsigned char Phmagtype;
+
+ unsigned char Pcurrentbasefunc;//The base function used - 0=sin, 1=...
+ unsigned char Pbasefuncpar;//the parameter of the base function
+
+ unsigned char Pbasefuncmodulation;//what modulation is applied to the basefunc
+ unsigned char Pbasefuncmodulationpar1,Pbasefuncmodulationpar2,Pbasefuncmodulationpar3;//the parameter of the base function modulation
+
+ /*the Randomness:
+ 64=no randomness
+ 63..0 - block type randomness - 0 is maximum
+ 65..127 - each harmonic randomness - 127 is maximum*/
+ unsigned char Prand;
+ unsigned char Pwaveshaping,Pwaveshapingfunction;
+ unsigned char Pfiltertype,Pfilterpar1,Pfilterpar2;
+ unsigned char Pfilterbeforews;
+ unsigned char Psatype,Psapar;//spectrum adjust
+
+ unsigned char Pamprandpower, Pamprandtype;//amplitude randomness
+ int Pharmonicshift;//how the harmonics are shifted
+ int Pharmonicshiftfirst;//if the harmonic shift is done before waveshaping and filter
+
+ unsigned char Padaptiveharmonics;//the adaptive harmonics status (off=0,on=1,etc..)
+ unsigned char Padaptiveharmonicsbasefreq;//the base frequency of the adaptive harmonic (30..3000Hz)
+ unsigned char Padaptiveharmonicspower;//the strength of the effect (0=off,100=full)
+ unsigned char Padaptiveharmonicspar;//the parameters in 2,3,4.. modes of adaptive harmonics
+
+ unsigned char Pmodulation;//what modulation is applied to the oscil
+ unsigned char Pmodulationpar1,Pmodulationpar2,Pmodulationpar3;//the parameter of the parameters
+
+
+ //makes a new random seed for Amplitude Randomness
+ //this should be called every note on event
+ void newrandseed(unsigned int randseed);
+
+ bool ADvsPAD;//if it is used by ADsynth or by PADsynth
+
+ static REALTYPE *tmpsmps;//this array stores some termporary data and it has SOUND_BUFFER_SIZE elements
+ static FFTFREQS outoscilFFTfreqs;
+
+private:
+
+ REALTYPE hmag[MAX_AD_HARMONICS],hphase[MAX_AD_HARMONICS];//the magnituides and the phases of the sine/nonsine harmonics
// private:
- FFTwrapper *fft;
- //computes the basefunction and make the FFT; newbasefunc<0 = same basefunc
- void changebasefunction();
- //Waveshaping
- void waveshape();
-
- //Filter the oscillator accotding to Pfiltertype and Pfilterpar
- void oscilfilter();
-
- //Adjust the spectrum
- void spectrumadjust();
-
- //Shift the harmonics
- void shiftharmonics();
-
- //Do the oscil modulation stuff
- void modulation();
-
- //Do the adaptive harmonic stuff
- void adaptiveharmonic(FFTFREQS f,REALTYPE freq);
-
- //Do the adaptive harmonic postprocessing (2n+1,2xS,2xA,etc..)
- //this function is called even for the user interface
- //this can be called for the sine and components, and for the spectrum
- //(that's why the sine and cosine components should be processed with a separate call)
- void adaptiveharmonicpostprocess(REALTYPE *f, int size);
-
- //Basic/base functions (Functiile De Baza)
- REALTYPE basefunc_pulse(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_saw(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_triangle(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_power(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_gauss(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_diode(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_abssine(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_pulsesine(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_stretchsine(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_chirp(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_absstretchsine(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_chebyshev(REALTYPE x,REALTYPE a);
- REALTYPE basefunc_sqr(REALTYPE x,REALTYPE a);
-
- //Internal Data
- unsigned char oldbasefunc,oldbasepar,oldhmagtype,oldwaveshapingfunction,oldwaveshaping;
- int oldfilterpars,oldsapars,oldbasefuncmodulation,oldbasefuncmodulationpar1,oldbasefuncmodulationpar2,oldbasefuncmodulationpar3,oldharmonicshift;
- int oldmodulation,oldmodulationpar1,oldmodulationpar2,oldmodulationpar3;
-
-
- FFTFREQS basefuncFFTfreqs;//Base Function Frequencies
- FFTFREQS 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()
-
- Resonance *res;
-
- unsigned int randseed;
-
+ FFTwrapper *fft;
+ //computes the basefunction and make the FFT; newbasefunc<0 = same basefunc
+ void changebasefunction();
+ //Waveshaping
+ void waveshape();
+
+ //Filter the oscillator accotding to Pfiltertype and Pfilterpar
+ void oscilfilter();
+
+ //Adjust the spectrum
+ void spectrumadjust();
+
+ //Shift the harmonics
+ void shiftharmonics();
+
+ //Do the oscil modulation stuff
+ void modulation();
+
+ //Do the adaptive harmonic stuff
+ void adaptiveharmonic(FFTFREQS f,REALTYPE freq);
+
+ //Do the adaptive harmonic postprocessing (2n+1,2xS,2xA,etc..)
+ //this function is called even for the user interface
+ //this can be called for the sine and components, and for the spectrum
+ //(that's why the sine and cosine components should be processed with a separate call)
+ void adaptiveharmonicpostprocess(REALTYPE *f, int size);
+
+ //Basic/base functions (Functiile De Baza)
+ REALTYPE basefunc_pulse(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_saw(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_triangle(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_power(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_gauss(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_diode(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_abssine(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_pulsesine(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_stretchsine(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_chirp(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_absstretchsine(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_chebyshev(REALTYPE x,REALTYPE a);
+ REALTYPE basefunc_sqr(REALTYPE x,REALTYPE a);
+
+ //Internal Data
+ unsigned char oldbasefunc,oldbasepar,oldhmagtype,oldwaveshapingfunction,oldwaveshaping;
+ int oldfilterpars,oldsapars,oldbasefuncmodulation,oldbasefuncmodulationpar1,oldbasefuncmodulationpar2,oldbasefuncmodulationpar3,oldharmonicshift;
+ int oldmodulation,oldmodulationpar1,oldmodulationpar2,oldmodulationpar3;
+
+
+ FFTFREQS basefuncFFTfreqs;//Base Function Frequencies
+ FFTFREQS 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()
+
+ Resonance *res;
+
+ unsigned int randseed;
+
};
diff --git a/src/Synth/PADnote.C b/src/Synth/PADnote.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
PADnote.C - The "pad" synthesizer
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -22,7 +22,8 @@
#include "PADnote.h"
#include "../Misc/Config.h"
-PADnote::PADnote(PADnoteParameters *parameters, Controller *ctl_,REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool besilent){
+PADnote::PADnote(PADnoteParameters *parameters, Controller *ctl_,REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool besilent)
+{
ready=0;
// Initialise some legato-specific vars
@@ -42,99 +43,99 @@ PADnote::PADnote(PADnoteParameters *parameters, Controller *ctl_,REALTYPE freq,
ctl=ctl_;
this->velocity=velocity;
finished_=false;
-
+
if (pars->Pfixedfreq==0) basefreq=freq;
- else {
- basefreq=440.0;
- int fixedfreqET=pars->PfixedfreqET;
- if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
- REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
- if (fixedfreqET<=64) basefreq*=pow(2.0,tmp);
- else basefreq*=pow(3.0,tmp);
- };
+ else {
+ basefreq=440.0;
+ int fixedfreqET=pars->PfixedfreqET;
+ if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
+ REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
+ if (fixedfreqET<=64) basefreq*=pow(2.0,tmp);
+ else basefreq*=pow(3.0,tmp);
+ };
- };
+ };
firsttime=true;
released=false;
realfreq=basefreq;
NoteGlobalPar.Detune=getdetune(pars->PDetuneType
- ,pars->PCoarseDetune,pars->PDetune);
+ ,pars->PCoarseDetune,pars->PDetune);
//find out the closest note
REALTYPE logfreq=log(basefreq*pow(2.0,NoteGlobalPar.Detune/1200.0));
REALTYPE mindist=fabs(logfreq-log(pars->sample[0].basefreq+0.0001));
nsample=0;
- for (int i=1;i<PAD_MAX_SAMPLES;i++){
- if (pars->sample[i].smp==NULL) break;
- REALTYPE dist=fabs(logfreq-log(pars->sample[i].basefreq+0.0001));
+ for (int i=1;i<PAD_MAX_SAMPLES;i++) {
+ if (pars->sample[i].smp==NULL) break;
+ REALTYPE dist=fabs(logfreq-log(pars->sample[i].basefreq+0.0001));
// printf("(mindist=%g) %i %g %g\n",mindist,i,dist,pars->sample[i].basefreq);
-
- if (dist<mindist){
- nsample=i;
- mindist=dist;
- };
+
+ if (dist<mindist) {
+ nsample=i;
+ mindist=dist;
+ };
};
int size=pars->sample[nsample].size;
if (size==0) size=1;
-
-
+
+
poshi_l=(int)(RND*(size-1));
if (pars->PStereo!=0) poshi_r=(poshi_l+size/2)%size;
- else poshi_r=poshi_l;
+ else poshi_r=poshi_l;
poslo=0.0;
-
+
tmpwave=new REALTYPE [SOUND_BUFFER_SIZE];
-
+
if (pars->PPanning==0) NoteGlobalPar.Panning=RND;
- else NoteGlobalPar.Panning=pars->PPanning/128.0;
+ else NoteGlobalPar.Panning=pars->PPanning/128.0;
NoteGlobalPar.FilterCenterPitch=pars->GlobalFilter->getfreq()+//center freq
- pars->PFilterVelocityScale/127.0*6.0* //velocity sensing
- (VelF(velocity,pars->PFilterVelocityScaleFunction)-1);
+ pars->PFilterVelocityScale/127.0*6.0* //velocity sensing
+ (VelF(velocity,pars->PFilterVelocityScaleFunction)-1);
if (pars->PPunchStrength!=0) {
NoteGlobalPar.Punch.Enabled=1;
- NoteGlobalPar.Punch.t=1.0;//start from 1.0 and to 0.0
- NoteGlobalPar.Punch.initialvalue=( (pow(10,1.5*pars->PPunchStrength/127.0)-1.0)
- *VelF(velocity,pars->PPunchVelocitySensing) );
- REALTYPE time=pow(10,3.0*pars->PPunchTime/127.0)/10000.0;//0.1 .. 100 ms
- REALTYPE stretch=pow(440.0/freq,pars->PPunchStretch/64.0);
- NoteGlobalPar.Punch.dt=1.0/(time*SAMPLE_RATE*stretch);
+ NoteGlobalPar.Punch.t=1.0;//start from 1.0 and to 0.0
+ NoteGlobalPar.Punch.initialvalue=( (pow(10,1.5*pars->PPunchStrength/127.0)-1.0)
+ *VelF(velocity,pars->PPunchVelocitySensing) );
+ REALTYPE time=pow(10,3.0*pars->PPunchTime/127.0)/10000.0;//0.1 .. 100 ms
+ REALTYPE stretch=pow(440.0/freq,pars->PPunchStretch/64.0);
+ NoteGlobalPar.Punch.dt=1.0/(time*SAMPLE_RATE*stretch);
} else NoteGlobalPar.Punch.Enabled=0;
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.Volume=4.0*pow(0.1,3.0*(1.0-pars->PVolume/96.0))//-60 dB .. 0 dB
- *VelF(velocity,pars->PAmpVelocityScaleFunction);//velocity sensing
+ *VelF(velocity,pars->PAmpVelocityScaleFunction);//velocity sensing
NoteGlobalPar.AmpEnvelope->envout_dB();//discard the first envelope output
globaloldamplitude=globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout();
NoteGlobalPar.GlobalFilterL=new Filter(pars->GlobalFilter);
NoteGlobalPar.GlobalFilterR=new Filter(pars->GlobalFilter);
-
+
NoteGlobalPar.FilterEnvelope=new Envelope(pars->FilterEnvelope,basefreq);
NoteGlobalPar.FilterLfo=new LFO(pars->FilterLfo,basefreq);
NoteGlobalPar.FilterQ=pars->GlobalFilter->getq();
NoteGlobalPar.FilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq);
-
+
ready=1;///sa il pun pe asta doar cand e chiar gata
- if (parameters->sample[nsample].smp==NULL){
- finished_=true;
- return;
+ if (parameters->sample[nsample].smp==NULL) {
+ finished_=true;
+ return;
};
};
@@ -143,97 +144,99 @@ PADnote::PADnote(PADnoteParameters *parameters, Controller *ctl_,REALTYPE freq,
// with some lines removed so that it only alter the already playing
// note (to perform legato). It is possible I left stuff that is not
// required for this.
-void PADnote::PADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool externcall){
- PADnoteParameters *parameters=pars;
- //Controller *ctl_=ctl;
-
- // Manage legato stuff
- if (externcall) Legato.msg=LM_Norm;
- if (Legato.msg!=LM_CatchUp){
- Legato.lastfreq=Legato.param.freq;
- Legato.param.freq=freq;
- Legato.param.vel=velocity;
- Legato.param.portamento=portamento_;
- Legato.param.midinote=midinote;
- if (Legato.msg==LM_Norm){
- if (Legato.silent){
- Legato.fade.m=0.0;
- Legato.msg=LM_FadeIn;
- } else {
- Legato.fade.m=1.0;
- Legato.msg=LM_FadeOut;
- return;
- }
- }
- if (Legato.msg==LM_ToNorm) Legato.msg=LM_Norm;
- }
-
- portamento=portamento_;
- this->velocity=velocity;
- finished_=false;
-
- if (pars->Pfixedfreq==0) basefreq=freq;
- else {
- basefreq=440.0;
- int fixedfreqET=pars->PfixedfreqET;
- if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
- REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
- if (fixedfreqET<=64) basefreq*=pow(2.0,tmp);
- else basefreq*=pow(3.0,tmp);
- };
- };
-
- released=false;
- realfreq=basefreq;
-
- getdetune(pars->PDetuneType,pars->PCoarseDetune,pars->PDetune);
-
-
- //find out the closest note
- REALTYPE logfreq=log(basefreq*pow(2.0,NoteGlobalPar.Detune/1200.0));
- REALTYPE mindist=fabs(logfreq-log(pars->sample[0].basefreq+0.0001));
- nsample=0;
- for (int i=1;i<PAD_MAX_SAMPLES;i++){
- if (pars->sample[i].smp==NULL) break;
- REALTYPE dist=fabs(logfreq-log(pars->sample[i].basefreq+0.0001));
-
- if (dist<mindist){
- nsample=i;
- mindist=dist;
- };
- };
-
- int size=pars->sample[nsample].size;
- if (size==0) size=1;
-
- if (pars->PPanning==0) NoteGlobalPar.Panning=RND;
- else NoteGlobalPar.Panning=pars->PPanning/128.0;
-
- NoteGlobalPar.FilterCenterPitch=pars->GlobalFilter->getfreq()+//center freq
- pars->PFilterVelocityScale/127.0*6.0* //velocity sensing
- (VelF(velocity,pars->PFilterVelocityScaleFunction)-1);
-
-
- NoteGlobalPar.Volume=4.0*pow(0.1,3.0*(1.0-pars->PVolume/96.0))//-60 dB .. 0 dB
- *VelF(velocity,pars->PAmpVelocityScaleFunction);//velocity sensing
-
- NoteGlobalPar.AmpEnvelope->envout_dB();//discard the first envelope output
- globaloldamplitude=globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout();
-
- NoteGlobalPar.FilterQ=pars->GlobalFilter->getq();
- NoteGlobalPar.FilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq);
-
-
- if (parameters->sample[nsample].smp==NULL){
- finished_=true;
- return;
- };
-
- // End of the PADlegatonote function.
+void PADnote::PADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool externcall)
+{
+ PADnoteParameters *parameters=pars;
+ //Controller *ctl_=ctl;
+
+ // Manage legato stuff
+ if (externcall) Legato.msg=LM_Norm;
+ if (Legato.msg!=LM_CatchUp) {
+ Legato.lastfreq=Legato.param.freq;
+ Legato.param.freq=freq;
+ Legato.param.vel=velocity;
+ Legato.param.portamento=portamento_;
+ Legato.param.midinote=midinote;
+ if (Legato.msg==LM_Norm) {
+ if (Legato.silent) {
+ Legato.fade.m=0.0;
+ Legato.msg=LM_FadeIn;
+ } else {
+ Legato.fade.m=1.0;
+ Legato.msg=LM_FadeOut;
+ return;
+ }
+ }
+ if (Legato.msg==LM_ToNorm) Legato.msg=LM_Norm;
+ }
+
+ portamento=portamento_;
+ this->velocity=velocity;
+ finished_=false;
+
+ if (pars->Pfixedfreq==0) basefreq=freq;
+ else {
+ basefreq=440.0;
+ int fixedfreqET=pars->PfixedfreqET;
+ if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
+ REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
+ if (fixedfreqET<=64) basefreq*=pow(2.0,tmp);
+ else basefreq*=pow(3.0,tmp);
+ };
+ };
+
+ released=false;
+ realfreq=basefreq;
+
+ getdetune(pars->PDetuneType,pars->PCoarseDetune,pars->PDetune);
+
+
+ //find out the closest note
+ REALTYPE logfreq=log(basefreq*pow(2.0,NoteGlobalPar.Detune/1200.0));
+ REALTYPE mindist=fabs(logfreq-log(pars->sample[0].basefreq+0.0001));
+ nsample=0;
+ for (int i=1;i<PAD_MAX_SAMPLES;i++) {
+ if (pars->sample[i].smp==NULL) break;
+ REALTYPE dist=fabs(logfreq-log(pars->sample[i].basefreq+0.0001));
+
+ if (dist<mindist) {
+ nsample=i;
+ mindist=dist;
+ };
+ };
+
+ int size=pars->sample[nsample].size;
+ if (size==0) size=1;
+
+ if (pars->PPanning==0) NoteGlobalPar.Panning=RND;
+ else NoteGlobalPar.Panning=pars->PPanning/128.0;
+
+ NoteGlobalPar.FilterCenterPitch=pars->GlobalFilter->getfreq()+//center freq
+ pars->PFilterVelocityScale/127.0*6.0* //velocity sensing
+ (VelF(velocity,pars->PFilterVelocityScaleFunction)-1);
+
+
+ NoteGlobalPar.Volume=4.0*pow(0.1,3.0*(1.0-pars->PVolume/96.0))//-60 dB .. 0 dB
+ *VelF(velocity,pars->PAmpVelocityScaleFunction);//velocity sensing
+
+ NoteGlobalPar.AmpEnvelope->envout_dB();//discard the first envelope output
+ globaloldamplitude=globalnewamplitude=NoteGlobalPar.Volume*NoteGlobalPar.AmpEnvelope->envout_dB()*NoteGlobalPar.AmpLfo->amplfoout();
+
+ NoteGlobalPar.FilterQ=pars->GlobalFilter->getq();
+ NoteGlobalPar.FilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq);
+
+
+ if (parameters->sample[nsample].smp==NULL) {
+ finished_=true;
+ return;
+ };
+
+ // End of the PADlegatonote function.
};
-PADnote::~PADnote(){
+PADnote::~PADnote()
+{
delete (NoteGlobalPar.FreqEnvelope);
delete (NoteGlobalPar.FreqLfo);
delete (NoteGlobalPar.AmpEnvelope);
@@ -246,271 +249,279 @@ PADnote::~PADnote(){
};
-inline void PADnote::fadein(REALTYPE *smps){
+inline void PADnote::fadein(REALTYPE *smps)
+{
int zerocrossings=0;
for (int i=1;i<SOUND_BUFFER_SIZE;i++)
- if ((smps[i-1]<0.0) && (smps[i]>0.0)) zerocrossings++;//this is only the possitive crossings
+ if ((smps[i-1]<0.0) && (smps[i]>0.0)) zerocrossings++;//this is only the possitive crossings
REALTYPE tmp=(SOUND_BUFFER_SIZE-1.0)/(zerocrossings+1)/3.0;
if (tmp<8.0) tmp=8.0;
int n;
- F2I(tmp,n);//how many samples is the fade-in
+ F2I(tmp,n);//how many samples is the fade-in
if (n>SOUND_BUFFER_SIZE) n=SOUND_BUFFER_SIZE;
for (int i=0;i<n;i++) {//fade-in
- REALTYPE tmp=0.5-cos((REALTYPE)i/(REALTYPE) n*PI)*0.5;
- smps[i]*=tmp;
+ REALTYPE tmp=0.5-cos((REALTYPE)i/(REALTYPE) n*PI)*0.5;
+ smps[i]*=tmp;
};
};
-void PADnote::computecurrentparameters(){
+void PADnote::computecurrentparameters()
+{
REALTYPE globalpitch,globalfilterpitch;
globalpitch=0.01*(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()*NoteGlobalPar.AmpLfo->amplfoout();
-
+
globalfilterpitch=NoteGlobalPar.FilterEnvelope->envout()+NoteGlobalPar.FilterLfo->lfoout()
+NoteGlobalPar.FilterCenterPitch;
-
+
REALTYPE tmpfilterfreq=globalfilterpitch+ctl->filtercutoff.relfreq
- +NoteGlobalPar.FilterFreqTracking;
-
+ +NoteGlobalPar.FilterFreqTracking;
+
tmpfilterfreq=NoteGlobalPar.GlobalFilterL->getrealfreq(tmpfilterfreq);
-
+
REALTYPE globalfilterq=NoteGlobalPar.FilterQ*ctl->filterq.relq;
NoteGlobalPar.GlobalFilterL->setfreq_and_q(tmpfilterfreq,globalfilterq);
NoteGlobalPar.GlobalFilterR->setfreq_and_q(tmpfilterfreq,globalfilterq);
//compute the portamento, if it is used by this note
- REALTYPE portamentofreqrap=1.0;
- if (portamento!=0){//this voice use portamento
- portamentofreqrap=ctl->portamento.freqrap;
- if (ctl->portamento.used==0){//the portamento has finished
- portamento=0;//this note is no longer "portamented"
- };
+ REALTYPE portamentofreqrap=1.0;
+ if (portamento!=0) {//this voice use portamento
+ portamentofreqrap=ctl->portamento.freqrap;
+ if (ctl->portamento.used==0) {//the portamento has finished
+ portamento=0;//this note is no longer "portamented"
+ };
};
realfreq=basefreq*portamentofreqrap*pow(2.0,globalpitch/12.0)*ctl->pitchwheel.relfreq;
};
-int PADnote::Compute_Linear(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo){
+int PADnote::Compute_Linear(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo)
+{
REALTYPE *smps=pars->sample[nsample].smp;
- if (smps==NULL){
- finished_=true;
- return(1);
+ if (smps==NULL) {
+ finished_=true;
+ return(1);
};
int size=pars->sample[nsample].size;
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- poshi_l+=freqhi;
- poshi_r+=freqhi;
- poslo+=freqlo;
- if (poslo>=1.0){
- poshi_l+=1;
- poshi_r+=1;
- poslo-=1.0;
- };
- if (poshi_l>=size) poshi_l%=size;
- if (poshi_r>=size) poshi_r%=size;
-
- outl[i]=smps[poshi_l]*(1.0-poslo)+smps[poshi_l+1]*poslo;
- outr[i]=smps[poshi_r]*(1.0-poslo)+smps[poshi_r+1]*poslo;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ poshi_l+=freqhi;
+ poshi_r+=freqhi;
+ poslo+=freqlo;
+ if (poslo>=1.0) {
+ poshi_l+=1;
+ poshi_r+=1;
+ poslo-=1.0;
+ };
+ if (poshi_l>=size) poshi_l%=size;
+ if (poshi_r>=size) poshi_r%=size;
+
+ outl[i]=smps[poshi_l]*(1.0-poslo)+smps[poshi_l+1]*poslo;
+ outr[i]=smps[poshi_r]*(1.0-poslo)+smps[poshi_r+1]*poslo;
};
return(1);
};
-int PADnote::Compute_Cubic(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo){
+int PADnote::Compute_Cubic(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo)
+{
REALTYPE *smps=pars->sample[nsample].smp;
- if (smps==NULL){
- finished_=true;
- return(1);
+ if (smps==NULL) {
+ finished_=true;
+ return(1);
};
int size=pars->sample[nsample].size;
REALTYPE xm1,x0,x1,x2,a,b,c;
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- poshi_l+=freqhi;
- poshi_r+=freqhi;
- poslo+=freqlo;
- if (poslo>=1.0){
- poshi_l+=1;
- poshi_r+=1;
- poslo-=1.0;
- };
- if (poshi_l>=size) poshi_l%=size;
- if (poshi_r>=size) poshi_r%=size;
-
-
- //left
- xm1=smps[poshi_l];
- x0=smps[poshi_l + 1];
- x1=smps[poshi_l + 2];
- x2=smps[poshi_l + 3];
- a = (3.0 * (x0-x1) - xm1 + x2)*0.5;
- b = 2.0*x1 + xm1 - (5.0*x0 + x2)*0.5;
- c = (x1 - xm1)*0.5;
- outl[i] = (((a * poslo) + b) * poslo + c) * poslo + x0;
- //right
- xm1=smps[poshi_r];
- x0=smps[poshi_r + 1];
- x1=smps[poshi_r + 2];
- x2=smps[poshi_r + 3];
- a = (3.0 * (x0-x1) - xm1 + x2)*0.5;
- b = 2.0*x1 + xm1 - (5.0*x0 + x2)*0.5;
- c = (x1 - xm1)*0.5;
- outr[i] = (((a * poslo) + b) * poslo + c) * poslo + x0;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ poshi_l+=freqhi;
+ poshi_r+=freqhi;
+ poslo+=freqlo;
+ if (poslo>=1.0) {
+ poshi_l+=1;
+ poshi_r+=1;
+ poslo-=1.0;
+ };
+ if (poshi_l>=size) poshi_l%=size;
+ if (poshi_r>=size) poshi_r%=size;
+
+
+ //left
+ xm1=smps[poshi_l];
+ x0=smps[poshi_l + 1];
+ x1=smps[poshi_l + 2];
+ x2=smps[poshi_l + 3];
+ a = (3.0 * (x0-x1) - xm1 + x2)*0.5;
+ b = 2.0*x1 + xm1 - (5.0*x0 + x2)*0.5;
+ c = (x1 - xm1)*0.5;
+ outl[i] = (((a * poslo) + b) * poslo + c) * poslo + x0;
+ //right
+ xm1=smps[poshi_r];
+ x0=smps[poshi_r + 1];
+ x1=smps[poshi_r + 2];
+ x2=smps[poshi_r + 3];
+ a = (3.0 * (x0-x1) - xm1 + x2)*0.5;
+ b = 2.0*x1 + xm1 - (5.0*x0 + x2)*0.5;
+ c = (x1 - xm1)*0.5;
+ outr[i] = (((a * poslo) + b) * poslo + c) * poslo + x0;
};
return(1);
};
-int PADnote::noteout(REALTYPE *outl,REALTYPE *outr){
+int PADnote::noteout(REALTYPE *outl,REALTYPE *outr)
+{
computecurrentparameters();
REALTYPE *smps=pars->sample[nsample].smp;
- if (smps==NULL){
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- outl[i]=0.0;
- outr[i]=0.0;
- };
- return(1);
+ if (smps==NULL) {
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]=0.0;
+ outr[i]=0.0;
+ };
+ return(1);
};
REALTYPE smpfreq=pars->sample[nsample].basefreq;
-
+
REALTYPE freqrap=realfreq/smpfreq;
int freqhi=(int) (floor(freqrap));
REALTYPE freqlo=freqrap-floor(freqrap);
-
+
if (config.cfg.Interpolation) Compute_Cubic(outl,outr,freqhi,freqlo);
- else Compute_Linear(outl,outr,freqhi,freqlo);
-
-
- if (firsttime){
- fadein(outl);
- fadein(outr);
- firsttime=false;
+ else Compute_Linear(outl,outr,freqhi,freqlo);
+
+
+ if (firsttime) {
+ fadein(outl);
+ fadein(outr);
+ firsttime=false;
};
- NoteGlobalPar.GlobalFilterL->filterout(outl);
- NoteGlobalPar.GlobalFilterR->filterout(outr);
-
- //Apply the punch
- if (NoteGlobalPar.Punch.Enabled!=0){
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- REALTYPE punchamp=NoteGlobalPar.Punch.initialvalue*NoteGlobalPar.Punch.t+1.0;
- outl[i]*=punchamp;
- outr[i]*=punchamp;
- NoteGlobalPar.Punch.t-=NoteGlobalPar.Punch.dt;
- if (NoteGlobalPar.Punch.t<0.0) {
- NoteGlobalPar.Punch.Enabled=0;
- break;
- };
- };
+ NoteGlobalPar.GlobalFilterL->filterout(outl);
+ NoteGlobalPar.GlobalFilterR->filterout(outr);
+
+ //Apply the punch
+ if (NoteGlobalPar.Punch.Enabled!=0) {
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE punchamp=NoteGlobalPar.Punch.initialvalue*NoteGlobalPar.Punch.t+1.0;
+ outl[i]*=punchamp;
+ outr[i]*=punchamp;
+ NoteGlobalPar.Punch.t-=NoteGlobalPar.Punch.dt;
+ if (NoteGlobalPar.Punch.t<0.0) {
+ NoteGlobalPar.Punch.Enabled=0;
+ break;
+ };
};
+ };
- if (ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude,globalnewamplitude)){
- // Amplitude Interpolation
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(globaloldamplitude,globalnewamplitude,i,SOUND_BUFFER_SIZE);
- outl[i]*=tmpvol*NoteGlobalPar.Panning;
- outr[i]*=tmpvol*(1.0-NoteGlobalPar.Panning);
- };
- } else {
- for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
- outl[i]*=globalnewamplitude*NoteGlobalPar.Panning;
- outr[i]*=globalnewamplitude*(1.0-NoteGlobalPar.Panning);
- };
+ if (ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude,globalnewamplitude)) {
+ // Amplitude Interpolation
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(globaloldamplitude,globalnewamplitude,i,SOUND_BUFFER_SIZE);
+ outl[i]*=tmpvol*NoteGlobalPar.Panning;
+ outr[i]*=tmpvol*(1.0-NoteGlobalPar.Panning);
};
+ } else {
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]*=globalnewamplitude*NoteGlobalPar.Panning;
+ outr[i]*=globalnewamplitude*(1.0-NoteGlobalPar.Panning);
+ };
+ };
// Apply legato-specific sound signal modifications
- if (Legato.silent){ // Silencer
- if (Legato.msg!=LM_FadeIn){
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- outl[i]=0.0;
- outr[i]=0.0;
- }
- }
+ if (Legato.silent) { // Silencer
+ if (Legato.msg!=LM_FadeIn) {
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]=0.0;
+ outr[i]=0.0;
+ }
+ }
}
- switch (Legato.msg){
+ switch (Legato.msg) {
case LM_CatchUp : // Continue the catch-up...
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){//Yea, could be done without the loop...
- Legato.decounter--;
- if (Legato.decounter<1){
- // Catching-up done, we can finally set
- // the note to the actual parameters.
- Legato.decounter=-10;
- Legato.msg=LM_ToNorm;
- PADlegatonote(Legato.param.freq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
- break;
- }
- }
- break;
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {//Yea, could be done without the loop...
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ // Catching-up done, we can finally set
+ // the note to the actual parameters.
+ Legato.decounter=-10;
+ Legato.msg=LM_ToNorm;
+ PADlegatonote(Legato.param.freq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
+ break;
+ }
+ }
+ break;
case LM_FadeIn : // Fade-in
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- Legato.silent=false;
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- Legato.decounter--;
- if (Legato.decounter<1){
- Legato.decounter=-10;
- Legato.msg=LM_Norm;
- break;
- }
- Legato.fade.m+=Legato.fade.step;
- outl[i]*=Legato.fade.m;
- outr[i]*=Legato.fade.m;
- }
- break;
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ Legato.silent=false;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ Legato.decounter=-10;
+ Legato.msg=LM_Norm;
+ break;
+ }
+ Legato.fade.m+=Legato.fade.step;
+ outl[i]*=Legato.fade.m;
+ outr[i]*=Legato.fade.m;
+ }
+ break;
case LM_FadeOut : // Fade-out, then set the catch-up
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- for (int i=0;i<SOUND_BUFFER_SIZE;i++){
- Legato.decounter--;
- if (Legato.decounter<1){
- for (int j=i;j<SOUND_BUFFER_SIZE;j++){
- outl[j]=0.0;
- outr[j]=0.0;
- }
- Legato.decounter=-10;
- Legato.silent=true;
- // Fading-out done, now set the catch-up :
- Legato.decounter=Legato.fade.length;
- Legato.msg=LM_CatchUp;
- REALTYPE catchupfreq=Legato.param.freq*(Legato.param.freq/Legato.lastfreq);//This freq should make this now silent note to catch-up (or should I say resync ?) with the heard note for the same length it stayed at the previous freq during the fadeout.
- PADlegatonote(catchupfreq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
- break;
- }
- Legato.fade.m-=Legato.fade.step;
- outl[i]*=Legato.fade.m;
- outr[i]*=Legato.fade.m;
- }
- break;
- default : break;
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ for (int j=i;j<SOUND_BUFFER_SIZE;j++) {
+ outl[j]=0.0;
+ outr[j]=0.0;
+ }
+ Legato.decounter=-10;
+ Legato.silent=true;
+ // Fading-out done, now set the catch-up :
+ Legato.decounter=Legato.fade.length;
+ Legato.msg=LM_CatchUp;
+ REALTYPE catchupfreq=Legato.param.freq*(Legato.param.freq/Legato.lastfreq);//This freq should make this now silent note to catch-up (or should I say resync ?) with the heard note for the same length it stayed at the previous freq during the fadeout.
+ PADlegatonote(catchupfreq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
+ break;
+ }
+ Legato.fade.m-=Legato.fade.step;
+ outl[i]*=Legato.fade.m;
+ outr[i]*=Legato.fade.m;
+ }
+ break;
+ default :
+ break;
}
- // Check if the global amplitude is finished.
- // If it does, disable the note
- if (NoteGlobalPar.AmpEnvelope->finished()!=0) {
- for (int i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out
- REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
- outl[i]*=tmp;
- outr[i]*=tmp;
- };
- finished_=1;
- };
+ // Check if the global amplitude is finished.
+ // If it does, disable the note
+ if (NoteGlobalPar.AmpEnvelope->finished()!=0) {
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out
+ REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
+ outl[i]*=tmp;
+ outr[i]*=tmp;
+ };
+ finished_=1;
+ };
return(1);
};
-int PADnote::finished(){
+int PADnote::finished()
+{
return(finished_);
};
-void PADnote::relasekey(){
- NoteGlobalPar.FreqEnvelope->relasekey();
- NoteGlobalPar.FilterEnvelope->relasekey();
- NoteGlobalPar.AmpEnvelope->relasekey();
+void PADnote::relasekey()
+{
+ NoteGlobalPar.FreqEnvelope->relasekey();
+ NoteGlobalPar.FilterEnvelope->relasekey();
+ NoteGlobalPar.AmpEnvelope->relasekey();
};
diff --git a/src/Synth/PADnote.h b/src/Synth/PADnote.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
PADnote.h - The "pad" synthesizer
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -30,95 +30,96 @@
#include "../Params/Controller.h"
/**The "pad" synthesizer*/
-class PADnote{
- public:
- PADnote(PADnoteParameters *parameters, Controller *ctl_,REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool besilent);
- ~PADnote();
-
- void PADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool externcall);
-
- int noteout(REALTYPE *outl,REALTYPE *outr);
- int finished();
- void relasekey();
-
- int ready;
-
- private:
- void fadein(REALTYPE *smps);
- void computecurrentparameters();
- bool finished_;
- PADnoteParameters *pars;
-
- int poshi_l,poshi_r;
- REALTYPE poslo;
-
- REALTYPE basefreq;
- bool firsttime,released;
-
- int nsample,portamento;
-
- int Compute_Linear(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo);
- int Compute_Cubic(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo);
-
-
- struct{
- /******************************************
- * FREQUENCY GLOBAL PARAMETERS *
- ******************************************/
- REALTYPE Detune;//cents
-
- Envelope *FreqEnvelope;
- LFO *FreqLfo;
-
- /********************************************
- * AMPLITUDE GLOBAL PARAMETERS *
- ********************************************/
- REALTYPE Volume;// [ 0 .. 1 ]
-
- REALTYPE Panning;// [ 0 .. 1 ]
-
- Envelope *AmpEnvelope;
- LFO *AmpLfo;
-
- struct {
- int Enabled;
- REALTYPE initialvalue,dt,t;
- } Punch;
-
- /******************************************
- * FILTER GLOBAL PARAMETERS *
- ******************************************/
- Filter *GlobalFilterL,*GlobalFilterR;
-
- REALTYPE FilterCenterPitch;//octaves
- REALTYPE FilterQ;
- REALTYPE FilterFreqTracking;
-
- Envelope *FilterEnvelope;
-
- LFO *FilterLfo;
- } NoteGlobalPar;
-
-
- REALTYPE globaloldamplitude,globalnewamplitude,velocity,realfreq;
- REALTYPE *tmpwave;
- Controller *ctl;
-
- // Legato vars
- struct {
- bool silent;
- REALTYPE lastfreq;
- LegatoMsg msg;
- int decounter;
- struct { // Fade In/Out vars
- int length;
- REALTYPE m, step;
- } fade;
- struct { // Note parameters
- REALTYPE freq, vel;
- int portamento, midinote;
- } param;
- } Legato;
+class PADnote
+{
+public:
+ PADnote(PADnoteParameters *parameters, Controller *ctl_,REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool besilent);
+ ~PADnote();
+
+ void PADlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool externcall);
+
+ int noteout(REALTYPE *outl,REALTYPE *outr);
+ int finished();
+ void relasekey();
+
+ int ready;
+
+private:
+ void fadein(REALTYPE *smps);
+ void computecurrentparameters();
+ bool finished_;
+ PADnoteParameters *pars;
+
+ int poshi_l,poshi_r;
+ REALTYPE poslo;
+
+ REALTYPE basefreq;
+ bool firsttime,released;
+
+ int nsample,portamento;
+
+ int Compute_Linear(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo);
+ int Compute_Cubic(REALTYPE *outl,REALTYPE *outr,int freqhi,REALTYPE freqlo);
+
+
+ struct {
+ /******************************************
+ * FREQUENCY GLOBAL PARAMETERS *
+ ******************************************/
+ REALTYPE Detune;//cents
+
+ Envelope *FreqEnvelope;
+ LFO *FreqLfo;
+
+ /********************************************
+ * AMPLITUDE GLOBAL PARAMETERS *
+ ********************************************/
+ REALTYPE Volume;// [ 0 .. 1 ]
+
+ REALTYPE Panning;// [ 0 .. 1 ]
+
+ Envelope *AmpEnvelope;
+ LFO *AmpLfo;
+
+ struct {
+ int Enabled;
+ REALTYPE initialvalue,dt,t;
+ } Punch;
+
+ /******************************************
+ * FILTER GLOBAL PARAMETERS *
+ ******************************************/
+ Filter *GlobalFilterL,*GlobalFilterR;
+
+ REALTYPE FilterCenterPitch;//octaves
+ REALTYPE FilterQ;
+ REALTYPE FilterFreqTracking;
+
+ Envelope *FilterEnvelope;
+
+ LFO *FilterLfo;
+ } NoteGlobalPar;
+
+
+ REALTYPE globaloldamplitude,globalnewamplitude,velocity,realfreq;
+ REALTYPE *tmpwave;
+ Controller *ctl;
+
+ // Legato vars
+ struct {
+ bool silent;
+ REALTYPE lastfreq;
+ LegatoMsg msg;
+ int decounter;
+ struct { // Fade In/Out vars
+ int length;
+ REALTYPE m, step;
+ } fade;
+ struct { // Note parameters
+ REALTYPE freq, vel;
+ int portamento, midinote;
+ } param;
+ } Legato;
};
diff --git a/src/Synth/Resonance.C b/src/Synth/Resonance.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
Resonance.C - Resonance
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -26,16 +26,19 @@
#include <stdio.h>
-Resonance::Resonance():Presets(){
+Resonance::Resonance():Presets()
+{
setpresettype("Presonance");
defaults();
};
-Resonance::~Resonance(){
+Resonance::~Resonance()
+{
};
-void Resonance::defaults(){
+void Resonance::defaults()
+{
Penabled=0;
PmaxdB=20;
Pcenterfreq=64;//1 kHz
@@ -49,7 +52,8 @@ void Resonance::defaults(){
/*
* Set a point of resonance function with a value
*/
-void Resonance::setpoint(int n,unsigned char p){
+void Resonance::setpoint(int n,unsigned char p)
+{
if ((n<0)||(n>=N_RES_POINTS)) return;
Prespoints[n]=p;
};
@@ -57,51 +61,59 @@ void Resonance::setpoint(int n,unsigned char p){
/*
* Apply the resonance to FFT data
*/
-void Resonance::applyres(int n,FFTFREQS fftdata,REALTYPE freq){
+void Resonance::applyres(int n,FFTFREQS fftdata,REALTYPE freq)
+{
if (Penabled==0) return;//if the resonance is disabled
REALTYPE sum=0.0,
- l1=log(getfreqx(0.0)*ctlcenter),
- l2=log(2.0)*getoctavesfreq()*ctlbw;
+ l1=log(getfreqx(0.0)*ctlcenter),
+ l2=log(2.0)*getoctavesfreq()*ctlbw;
for (int i=0;i<N_RES_POINTS;i++) if (sum<Prespoints[i]) sum=Prespoints[i];
if (sum<1.0) sum=1.0;
- for (int i=1;i<n;i++){
+ for (int i=1;i<n;i++) {
REALTYPE x=(log(freq*i)-l1)/l2;//compute where the n-th hamonics fits to the graph
- if (x<0.0) x=0.0;
-
- x*=N_RES_POINTS;
- REALTYPE 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;if (kx2>=N_RES_POINTS) kx2=N_RES_POINTS-1;
- REALTYPE y=(Prespoints[kx1]*(1.0-dx)+Prespoints[kx2]*dx)/127.0-sum/127.0;
-
- y=pow(10.0,y*PmaxdB/20.0);
-
- if ((Pprotectthefundamental!=0)&&(i==1)) y=1.0;
-
+ if (x<0.0) x=0.0;
+
+ x*=N_RES_POINTS;
+ REALTYPE 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;
+ if (kx2>=N_RES_POINTS) kx2=N_RES_POINTS-1;
+ REALTYPE y=(Prespoints[kx1]*(1.0-dx)+Prespoints[kx2]*dx)/127.0-sum/127.0;
+
+ y=pow(10.0,y*PmaxdB/20.0);
+
+ if ((Pprotectthefundamental!=0)&&(i==1)) y=1.0;
+
fftdata.c[i]*=y;
fftdata.s[i]*=y;
- };
+ };
};
/*
* Gets the response at the frequency "freq"
*/
-REALTYPE Resonance::getfreqresponse(REALTYPE freq){
+REALTYPE Resonance::getfreqresponse(REALTYPE freq)
+{
REALTYPE l1=log(getfreqx(0.0)*ctlcenter),
- l2=log(2.0)*getoctavesfreq()*ctlbw,sum=0.0;
-
+ l2=log(2.0)*getoctavesfreq()*ctlbw,sum=0.0;
+
for (int i=0;i<N_RES_POINTS;i++) if (sum<Prespoints[i]) sum=Prespoints[i];
if (sum<1.0) sum=1.0;
REALTYPE x=(log(freq)-l1)/l2;//compute where the n-th hamonics fits to the graph
if (x<0.0) x=0.0;
x*=N_RES_POINTS;
- REALTYPE 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;if (kx2>=N_RES_POINTS) kx2=N_RES_POINTS-1;
+ REALTYPE 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;
+ if (kx2>=N_RES_POINTS) kx2=N_RES_POINTS-1;
REALTYPE result=(Prespoints[kx1]*(1.0-dx)+Prespoints[kx2]*dx)/127.0-sum/127.0;
result=pow(10.0,result*PmaxdB/20.0);
return(result);
@@ -111,30 +123,32 @@ REALTYPE Resonance::getfreqresponse(REALTYPE freq){
/*
* Smooth the resonance function
*/
-void Resonance::smooth(){
+void Resonance::smooth()
+{
REALTYPE old=Prespoints[0];
- for (int i=0;i<N_RES_POINTS;i++){
+ for (int i=0;i<N_RES_POINTS;i++) {
old=old*0.4+Prespoints[i]*0.6;
- Prespoints[i]=(int) old;
+ Prespoints[i]=(int) old;
};
old=Prespoints[N_RES_POINTS-1];
- for (int i=N_RES_POINTS-1;i>0;i--){
+ for (int i=N_RES_POINTS-1;i>0;i--) {
old=old*0.4+Prespoints[i]*0.6;
- Prespoints[i]=(int) old+1;
- if (Prespoints[i]>127) Prespoints[i]=127;
+ Prespoints[i]=(int) old+1;
+ if (Prespoints[i]>127) Prespoints[i]=127;
};
};
/*
* Randomize the resonance function
*/
-void Resonance::randomize(int type){
+void Resonance::randomize(int type)
+{
int r=(int)(RND*127.0);
- for (int i=0;i<N_RES_POINTS;i++){
- Prespoints[i]=r;
- if ((RND<0.1)&&(type==0)) r=(int)(RND*127.0);
- if ((RND<0.3)&&(type==1)) r=(int)(RND*127.0);
- if (type==2) r=(int)(RND*127.0);
+ for (int i=0;i<N_RES_POINTS;i++) {
+ Prespoints[i]=r;
+ if ((RND<0.1)&&(type==0)) r=(int)(RND*127.0);
+ if ((RND<0.3)&&(type==1)) r=(int)(RND*127.0);
+ if (type==2) r=(int)(RND*127.0);
};
smooth();
};
@@ -142,26 +156,28 @@ void Resonance::randomize(int type){
/*
* Interpolate the peaks
*/
-void Resonance::interpolatepeaks(int type){
+void Resonance::interpolatepeaks(int type)
+{
int x1=0,y1=Prespoints[0];
- 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++){
- float x=(float) k/(i-x1);
- if (type==0) x=(1-cos(x*PI))*0.5;
- Prespoints[x1+k]=(int)(y1*(1.0-x)+y2*x);
- };
- x1=i;
- y1=y2;
- };
+ 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++) {
+ float x=(float) k/(i-x1);
+ if (type==0) x=(1-cos(x*PI))*0.5;
+ Prespoints[x1+k]=(int)(y1*(1.0-x)+y2*x);
+ };
+ x1=i;
+ y1=y2;
+ };
};
};
/*
* Get the frequency from x, where x is [0..1]; x is the x coordinate
*/
-REALTYPE Resonance::getfreqx(REALTYPE x){
+REALTYPE Resonance::getfreqx(REALTYPE x)
+{
if (x>1.0) x=1.0;
REALTYPE octf=pow(2.0,getoctavesfreq());
return(getcenterfreq()/sqrt(octf)*pow(octf,x));
@@ -170,35 +186,40 @@ REALTYPE Resonance::getfreqx(REALTYPE x){
/*
* Get the x coordinate from frequency (used by the UI)
*/
-REALTYPE Resonance::getfreqpos(REALTYPE freq){
+REALTYPE Resonance::getfreqpos(REALTYPE freq)
+{
return((log(freq)-log(getfreqx(0.0)))/log(2.0)/getoctavesfreq());
};
/*
* Get the center frequency of the resonance graph
*/
-REALTYPE Resonance::getcenterfreq(){
+REALTYPE Resonance::getcenterfreq()
+{
return(10000.0*pow(10,-(1.0-Pcenterfreq/127.0)*2.0));
};
/*
* Get the number of octave that the resonance functions applies to
*/
-REALTYPE Resonance::getoctavesfreq(){
+REALTYPE Resonance::getoctavesfreq()
+{
return(0.25+10.0*Poctavesfreq/127.0);
};
-void Resonance::sendcontroller(MidiControllers ctl,REALTYPE par){
+void Resonance::sendcontroller(MidiControllers ctl,REALTYPE par)
+{
if (ctl==C_resonance_center) ctlcenter=par;
- else ctlbw=par;
+ else ctlbw=par;
};
-void Resonance::add2XML(XMLwrapper *xml){
+void Resonance::add2XML(XMLwrapper *xml)
+{
xml->addparbool("enabled",Penabled);
-
+
if ((Penabled==0)&&(xml->minimal)) return;
xml->addpar("max_db",PmaxdB);
@@ -206,25 +227,26 @@ void Resonance::add2XML(XMLwrapper *xml){
xml->addpar("octaves_freq",Poctavesfreq);
xml->addparbool("protect_fundamental_frequency",Pprotectthefundamental);
xml->addpar("resonance_points",N_RES_POINTS);
- for (int i=0;i<N_RES_POINTS;i++){
- xml->beginbranch("RESPOINT",i);
- xml->addpar("val",Prespoints[i]);
- xml->endbranch();
+ for (int i=0;i<N_RES_POINTS;i++) {
+ xml->beginbranch("RESPOINT",i);
+ xml->addpar("val",Prespoints[i]);
+ xml->endbranch();
};
};
-void Resonance::getfromXML(XMLwrapper *xml){
+void Resonance::getfromXML(XMLwrapper *xml)
+{
Penabled=xml->getparbool("enabled",Penabled);
PmaxdB=xml->getpar127("max_db",PmaxdB);
Pcenterfreq=xml->getpar127("center_freq",Pcenterfreq);
Poctavesfreq=xml->getpar127("octaves_freq",Poctavesfreq);
Pprotectthefundamental=xml->getparbool("protect_fundamental_frequency",Pprotectthefundamental);
- for (int i=0;i<N_RES_POINTS;i++){
- if (xml->enterbranch("RESPOINT",i)==0) continue;
- Prespoints[i]=xml->getpar127("val",Prespoints[i]);
- xml->exitbranch();
+ for (int i=0;i<N_RES_POINTS;i++) {
+ if (xml->enterbranch("RESPOINT",i)==0) continue;
+ Prespoints[i]=xml->getpar127("val",Prespoints[i]);
+ xml->exitbranch();
};
};
diff --git a/src/Synth/Resonance.h b/src/Synth/Resonance.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
- Resonance.h - Resonance
+
+ Resonance.h - Resonance
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,30 +29,31 @@
#define N_RES_POINTS 256
-class Resonance:public Presets{
- public:
- Resonance();
- ~Resonance();
- void setpoint(int n,unsigned char p);
- void applyres(int n,FFTFREQS fftdata,REALTYPE freq);
- void smooth();
- void interpolatepeaks(int type);
- void randomize(int type);
+class Resonance:public Presets
+{
+public:
+ Resonance();
+ ~Resonance();
+ void setpoint(int n,unsigned char p);
+ void applyres(int n,FFTFREQS fftdata,REALTYPE freq);
+ void smooth();
+ void interpolatepeaks(int type);
+ void randomize(int type);
- void add2XML(XMLwrapper *xml);
- void defaults();
- void getfromXML(XMLwrapper *xml);
+ void add2XML(XMLwrapper *xml);
+ void defaults();
+ void getfromXML(XMLwrapper *xml);
- REALTYPE getfreqpos(REALTYPE freq);
- REALTYPE getfreqx(REALTYPE x);
- REALTYPE getfreqresponse(REALTYPE freq);
- REALTYPE getcenterfreq();
- REALTYPE getoctavesfreq();
- void sendcontroller(MidiControllers ctl,REALTYPE par);
+ REALTYPE getfreqpos(REALTYPE freq);
+ REALTYPE getfreqx(REALTYPE x);
+ REALTYPE getfreqresponse(REALTYPE freq);
+ REALTYPE getcenterfreq();
+ REALTYPE getoctavesfreq();
+ void sendcontroller(MidiControllers ctl,REALTYPE par);
//parameters
- unsigned char Penabled; //if the ressonance is enabled
+ unsigned char Penabled; //if the ressonance is enabled
unsigned char Prespoints[N_RES_POINTS]; //how many points define the resonance function
unsigned char PmaxdB; //how many dB the signal may be amplified
unsigned char Pcenterfreq,Poctavesfreq; //the center frequency of the res. func., and the number of octaves
@@ -61,8 +62,8 @@ class Resonance:public Presets{
//controllers
REALTYPE ctlcenter;//center frequency(relative)
REALTYPE ctlbw;//bandwidth(relative)
-
- private:
+
+private:
};
#endif
diff --git a/src/Synth/SUBnote.C b/src/Synth/SUBnote.C
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
SUBnote.C - The "subtractive" synthesizer
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -27,9 +27,10 @@
#include "SUBnote.h"
#include "../Misc/Util.h"
-SUBnote::SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote,bool besilent){
+SUBnote::SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote,bool besilent)
+{
ready=0;
-
+
tmpsmp=new REALTYPE[SOUND_BUFFER_SIZE];
tmprnd=new REALTYPE[SOUND_BUFFER_SIZE];
@@ -44,7 +45,7 @@ SUBnote::SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,RE
Legato.param.portamento=portamento_;
Legato.param.midinote=midinote;
Legato.silent=besilent;
-
+
pars=parameters;
ctl=ctl_;
portamento=portamento_;
@@ -52,7 +53,7 @@ SUBnote::SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,RE
volume=pow(0.1,3.0*(1.0-pars->PVolume/96.0));//-60 dB .. 0 dB
volume*=VelF(velocity,pars->PAmpVelocityScaleFunction);
if (pars->PPanning!=0) panning=pars->PPanning/127.0;
- else panning=RND;
+ else panning=RND;
numstages=pars->Pnumstages;
stereo=pars->Pstereo;
start=pars->Pstart;
@@ -60,95 +61,105 @@ SUBnote::SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,RE
int pos[MAX_SUB_HARMONICS];
if (pars->Pfixedfreq==0) basefreq=freq;
- else {
- basefreq=440.0;
- int fixedfreqET=pars->PfixedfreqET;
- if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
- REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
- if (fixedfreqET<=64) basefreq*=pow(2.0,tmp);
- else basefreq*=pow(3.0,tmp);
- };
-
- };
+ else {
+ basefreq=440.0;
+ int fixedfreqET=pars->PfixedfreqET;
+ if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
+ REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
+ if (fixedfreqET<=64) basefreq*=pow(2.0,tmp);
+ else basefreq*=pow(3.0,tmp);
+ };
+
+ };
REALTYPE detune=getdetune(pars->PDetuneType,pars->PCoarseDetune,pars->PDetune);
basefreq*=pow(2.0,detune/1200.0);//detune
// basefreq*=ctl->pitchwheel.relfreq;//pitch wheel
-
- //global filter
- GlobalFilterCenterPitch=pars->GlobalFilter->getfreq()+//center freq
- (pars->PGlobalFilterVelocityScale/127.0*6.0)* //velocity sensing
- (VelF(velocity,pars->PGlobalFilterVelocityScaleFunction)-1);
- GlobalFilterL=NULL;GlobalFilterR=NULL;
+ //global filter
+ GlobalFilterCenterPitch=pars->GlobalFilter->getfreq()+//center freq
+ (pars->PGlobalFilterVelocityScale/127.0*6.0)* //velocity sensing
+ (VelF(velocity,pars->PGlobalFilterVelocityScaleFunction)-1);
+
+ GlobalFilterL=NULL;
+ GlobalFilterR=NULL;
GlobalFilterEnvelope=NULL;
- //select only harmonics that desire to compute
+ //select only harmonics that desire to compute
numharmonics=0;
- for (int n=0;n<MAX_SUB_HARMONICS;n++){
- if (pars->Phmag[n]==0)continue;
- if (n*basefreq>SAMPLE_RATE/2.0) break;//remove the freqs above the Nyquist freq
- pos[numharmonics++]=n;
+ for (int n=0;n<MAX_SUB_HARMONICS;n++) {
+ if (pars->Phmag[n]==0)continue;
+ if (n*basefreq>SAMPLE_RATE/2.0) break;//remove the freqs above the Nyquist freq
+ pos[numharmonics++]=n;
};
firstnumharmonics=numharmonics;//(gf)Useful in legato mode.
-
+
if (numharmonics==0) {
- NoteEnabled=OFF;
- return;
+ NoteEnabled=OFF;
+ return;
};
-
-
+
+
lfilter=new bpfilter[numstages*numharmonics];
if (stereo!=0) rfilter=new bpfilter[numstages*numharmonics];
-
+
//how much the amplitude is normalised (because the harmonics)
REALTYPE reduceamp=0.0;
-
- for (int n=0;n<numharmonics;n++){
- REALTYPE freq=basefreq*(pos[n]+1);
-
- //the bandwidth is not absolute(Hz); it is relative to frequency
- REALTYPE bw=pow(10,(pars->Pbandwidth-127.0)/127.0*4)*numstages;
+ for (int n=0;n<numharmonics;n++) {
- //Bandwidth Scale
- bw*=pow(1000/freq,(pars->Pbwscale-64.0)/64.0*3.0);
+ REALTYPE freq=basefreq*(pos[n]+1);
- //Relative BandWidth
- bw*=pow(100,(pars->Phrelbw[pos[n]]-64.0)/64.0);
+ //the bandwidth is not absolute(Hz); it is relative to frequency
+ REALTYPE bw=pow(10,(pars->Pbandwidth-127.0)/127.0*4)*numstages;
- if (bw>25.0) bw=25.0;
+ //Bandwidth Scale
+ bw*=pow(1000/freq,(pars->Pbwscale-64.0)/64.0*3.0);
- //try to keep same amplitude on all freqs and bw. (empirically)
- REALTYPE gain=sqrt(1500.0/(bw*freq));
+ //Relative BandWidth
+ bw*=pow(100,(pars->Phrelbw[pos[n]]-64.0)/64.0);
+
+ if (bw>25.0) bw=25.0;
+
+ //try to keep same amplitude on all freqs and bw. (empirically)
+ REALTYPE gain=sqrt(1500.0/(bw*freq));
REALTYPE hmagnew=1.0-pars->Phmag[pos[n]]/127.0;
- REALTYPE hgain;
-
- switch(pars->Phmagtype){
- case 1:hgain=exp(hmagnew*log(0.01)); break;
- case 2:hgain=exp(hmagnew*log(0.001));break;
- case 3:hgain=exp(hmagnew*log(0.0001));break;
- case 4:hgain=exp(hmagnew*log(0.00001));break;
- default:hgain=1.0-hmagnew;
- };
- gain*=hgain;
+ REALTYPE hgain;
+
+ switch (pars->Phmagtype) {
+ case 1:
+ hgain=exp(hmagnew*log(0.01));
+ break;
+ case 2:
+ hgain=exp(hmagnew*log(0.001));
+ break;
+ case 3:
+ hgain=exp(hmagnew*log(0.0001));
+ break;
+ case 4:
+ hgain=exp(hmagnew*log(0.00001));
+ break;
+ default:
+ hgain=1.0-hmagnew;
+ };
+ gain*=hgain;
reduceamp+=hgain;
- for (int nph=0;nph<numstages;nph++){
- REALTYPE amp=1.0;
- if (nph==0) amp=gain;
- initfilter(lfilter[nph+n*numstages],freq,bw,amp,hgain);
- if (stereo!=0) initfilter(rfilter[nph+n*numstages],freq,bw,amp,hgain);
- };
+ for (int nph=0;nph<numstages;nph++) {
+ REALTYPE amp=1.0;
+ if (nph==0) amp=gain;
+ initfilter(lfilter[nph+n*numstages],freq,bw,amp,hgain);
+ if (stereo!=0) initfilter(rfilter[nph+n*numstages],freq,bw,amp,hgain);
+ };
};
-
+
if (reduceamp<0.001) reduceamp=1.0;
volume/=reduceamp;
-
+
oldpitchwheel=0;
oldbandwidth=64;
if (pars->Pfixedfreq==0) initparameters(basefreq);
- else initparameters(basefreq/440.0*freq);
+ else initparameters(basefreq/440.0*freq);
oldamplitude=newamplitude;
ready=1;
@@ -159,29 +170,30 @@ SUBnote::SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,RE
// initparameters(...) stuck together with some lines removed so that
// it only alter the already playing note (to perform legato). It is
// possible I left stuff that is not required for this.
-void SUBnote::SUBlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool externcall){
+void SUBnote::SUBlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool externcall)
+{
//SUBnoteParameters *parameters=pars;
//Controller *ctl_=ctl;
// Manage legato stuff
if (externcall) Legato.msg=LM_Norm;
- if (Legato.msg!=LM_CatchUp){
- Legato.lastfreq=Legato.param.freq;
- Legato.param.freq=freq;
- Legato.param.vel=velocity;
- Legato.param.portamento=portamento_;
- Legato.param.midinote=midinote;
- if (Legato.msg==LM_Norm){
- if (Legato.silent){
- Legato.fade.m=0.0;
- Legato.msg=LM_FadeIn;
- } else {
- Legato.fade.m=1.0;
- Legato.msg=LM_FadeOut;
- return;
- }
- }
- if (Legato.msg==LM_ToNorm) Legato.msg=LM_Norm;
+ if (Legato.msg!=LM_CatchUp) {
+ Legato.lastfreq=Legato.param.freq;
+ Legato.param.freq=freq;
+ Legato.param.vel=velocity;
+ Legato.param.portamento=portamento_;
+ Legato.param.midinote=midinote;
+ if (Legato.msg==LM_Norm) {
+ if (Legato.silent) {
+ Legato.fade.m=0.0;
+ Legato.msg=LM_FadeIn;
+ } else {
+ Legato.fade.m=1.0;
+ Legato.msg=LM_FadeOut;
+ return;
+ }
+ }
+ if (Legato.msg==LM_ToNorm) Legato.msg=LM_Norm;
}
portamento=portamento_;
@@ -197,78 +209,87 @@ void SUBnote::SUBlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, i
if (pars->Pfixedfreq==0) basefreq=freq;
else {
- basefreq=440.0;
- int fixedfreqET=pars->PfixedfreqET;
- if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
- REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
- if (fixedfreqET<=64) basefreq*=pow(2.0,tmp);
- else basefreq*=pow(3.0,tmp);
- };
+ basefreq=440.0;
+ int fixedfreqET=pars->PfixedfreqET;
+ if (fixedfreqET!=0) {//if the frequency varies according the keyboard note
+ REALTYPE tmp=(midinote-69.0)/12.0*(pow(2.0,(fixedfreqET-1)/63.0)-1.0);
+ if (fixedfreqET<=64) basefreq*=pow(2.0,tmp);
+ else basefreq*=pow(3.0,tmp);
+ };
};
REALTYPE detune=getdetune(pars->PDetuneType,pars->PCoarseDetune,pars->PDetune);
basefreq*=pow(2.0,detune/1200.0);//detune
//global filter
GlobalFilterCenterPitch=pars->GlobalFilter->getfreq()+//center freq
- (pars->PGlobalFilterVelocityScale/127.0*6.0)* //velocity sensing
- (VelF(velocity,pars->PGlobalFilterVelocityScaleFunction)-1);
+ (pars->PGlobalFilterVelocityScale/127.0*6.0)* //velocity sensing
+ (VelF(velocity,pars->PGlobalFilterVelocityScaleFunction)-1);
int legatonumharmonics=0;
- for (int n=0;n<MAX_SUB_HARMONICS;n++){
- if (pars->Phmag[n]==0)continue;
- if (n*basefreq>SAMPLE_RATE/2.0) break;//remove the freqs above the Nyquist freq
- pos[legatonumharmonics++]=n;
+ for (int n=0;n<MAX_SUB_HARMONICS;n++) {
+ if (pars->Phmag[n]==0)continue;
+ if (n*basefreq>SAMPLE_RATE/2.0) break;//remove the freqs above the Nyquist freq
+ pos[legatonumharmonics++]=n;
};
if (legatonumharmonics>firstnumharmonics) numharmonics=firstnumharmonics;
else numharmonics=legatonumharmonics;
if (numharmonics==0) {
- NoteEnabled=OFF;
- return;
+ NoteEnabled=OFF;
+ return;
};
//how much the amplitude is normalised (because the harmonics)
REALTYPE reduceamp=0.0;
- for (int n=0;n<numharmonics;n++){
-
- REALTYPE freq=basefreq*(pos[n]+1);
+ for (int n=0;n<numharmonics;n++) {
- //the bandwidth is not absolute(Hz); it is relative to frequency
- REALTYPE bw=pow(10,(pars->Pbandwidth-127.0)/127.0*4)*numstages;
+ REALTYPE freq=basefreq*(pos[n]+1);
- //Bandwidth Scale
- bw*=pow(1000/freq,(pars->Pbwscale-64.0)/64.0*3.0);
+ //the bandwidth is not absolute(Hz); it is relative to frequency
+ REALTYPE bw=pow(10,(pars->Pbandwidth-127.0)/127.0*4)*numstages;
- //Relative BandWidth
- bw*=pow(100,(pars->Phrelbw[pos[n]]-64.0)/64.0);
+ //Bandwidth Scale
+ bw*=pow(1000/freq,(pars->Pbwscale-64.0)/64.0*3.0);
- if (bw>25.0) bw=25.0;
+ //Relative BandWidth
+ bw*=pow(100,(pars->Phrelbw[pos[n]]-64.0)/64.0);
- //try to keep same amplitude on all freqs and bw. (empirically)
- REALTYPE gain=sqrt(1500.0/(bw*freq));
+ if (bw>25.0) bw=25.0;
- REALTYPE hmagnew=1.0-pars->Phmag[pos[n]]/127.0;
- REALTYPE hgain;
+ //try to keep same amplitude on all freqs and bw. (empirically)
+ REALTYPE gain=sqrt(1500.0/(bw*freq));
- switch(pars->Phmagtype){
- case 1:hgain=exp(hmagnew*log(0.01)); break;
- case 2:hgain=exp(hmagnew*log(0.001));break;
- case 3:hgain=exp(hmagnew*log(0.0001));break;
- case 4:hgain=exp(hmagnew*log(0.00001));break;
- default:hgain=1.0-hmagnew;
- };
- gain*=hgain;
- reduceamp+=hgain;
+ REALTYPE hmagnew=1.0-pars->Phmag[pos[n]]/127.0;
+ REALTYPE hgain;
+
+ switch (pars->Phmagtype) {
+ case 1:
+ hgain=exp(hmagnew*log(0.01));
+ break;
+ case 2:
+ hgain=exp(hmagnew*log(0.001));
+ break;
+ case 3:
+ hgain=exp(hmagnew*log(0.0001));
+ break;
+ case 4:
+ hgain=exp(hmagnew*log(0.00001));
+ break;
+ default:
+ hgain=1.0-hmagnew;
+ };
+ gain*=hgain;
+ reduceamp+=hgain;
- for (int nph=0;nph<numstages;nph++){
- REALTYPE amp=1.0;
- if (nph==0) amp=gain;
- initfilter(lfilter[nph+n*numstages],freq,bw,amp,hgain);
- if (stereo!=0) initfilter(rfilter[nph+n*numstages],freq,bw,amp,hgain);
- };
+ for (int nph=0;nph<numstages;nph++) {
+ REALTYPE amp=1.0;
+ if (nph==0) amp=gain;
+ initfilter(lfilter[nph+n*numstages],freq,bw,amp,hgain);
+ if (stereo!=0) initfilter(rfilter[nph+n*numstages],freq,bw,amp,hgain);
+ };
};
if (reduceamp<0.001) reduceamp=1.0;
@@ -281,24 +302,25 @@ void SUBnote::SUBlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, i
else freq*=basefreq/440.0;
- ///////////////
- // Altered initparameters(...) content:
+ ///////////////
+ // Altered initparameters(...) content:
- if (pars->PGlobalFilterEnabled!=0){
- globalfiltercenterq=pars->GlobalFilter->getq();
- GlobalFilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq);
+ if (pars->PGlobalFilterEnabled!=0) {
+ globalfiltercenterq=pars->GlobalFilter->getq();
+ GlobalFilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq);
};
- // end of the altered initparameters function content.
- ///////////////
+ // end of the altered initparameters function content.
+ ///////////////
oldamplitude=newamplitude;
- // End of the SUBlegatonote function.
+ // End of the SUBlegatonote function.
};
-SUBnote::~SUBnote(){
+SUBnote::~SUBnote()
+{
if (NoteEnabled!=OFF) KillNote();
delete [] tmpsmp;
delete [] tmprnd;
@@ -307,39 +329,42 @@ SUBnote::~SUBnote(){
/*
* Kill the note
*/
-void SUBnote::KillNote(){
- if (NoteEnabled!=OFF){
- delete [] lfilter;
- lfilter=NULL;
- if (stereo!=0) delete [] rfilter;
- rfilter=NULL;
- delete(AmpEnvelope);
- if (FreqEnvelope!=NULL) delete(FreqEnvelope);
- if (BandWidthEnvelope!=NULL) delete(BandWidthEnvelope);
- NoteEnabled=OFF;
+void SUBnote::KillNote()
+{
+ if (NoteEnabled!=OFF) {
+ delete [] lfilter;
+ lfilter=NULL;
+ if (stereo!=0) delete [] rfilter;
+ rfilter=NULL;
+ delete(AmpEnvelope);
+ if (FreqEnvelope!=NULL) delete(FreqEnvelope);
+ if (BandWidthEnvelope!=NULL) delete(BandWidthEnvelope);
+ NoteEnabled=OFF;
};
-
+
};
/*
* Compute the filters coefficients
*/
-void SUBnote::computefiltercoefs(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE gain){
+void SUBnote::computefiltercoefs(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE gain)
+{
if (freq>SAMPLE_RATE/2.0-200.0) {
- freq=SAMPLE_RATE/2.0-200.0;
+ freq=SAMPLE_RATE/2.0-200.0;
};
REALTYPE omega=2.0*PI*freq/SAMPLE_RATE;
- REALTYPE sn=sin(omega);REALTYPE cs=cos(omega);
+ REALTYPE sn=sin(omega);
+ REALTYPE cs=cos(omega);
REALTYPE alpha=sn*sinh(LOG_2/2.0*bw*omega/sn);
if (alpha>1) alpha=1;
if (alpha>bw) alpha=bw;
-
+
filter.b0=alpha/(1.0+alpha)*filter.amp*gain;
filter.b2=-alpha/(1.0+alpha)*filter.amp*gain;
- filter.a1=-2.0*cs/(1.0+alpha);
+ filter.a1=-2.0*cs/(1.0+alpha);
filter.a2=(1.0-alpha)/(1.0+alpha);
};
@@ -348,29 +373,31 @@ void SUBnote::computefiltercoefs(bpfilter &filter,REALTYPE freq,REALTYPE bw,REAL
/*
* Initialise the filters
*/
-void SUBnote::initfilter(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE amp,REALTYPE mag){
- filter.xn1=0.0;filter.xn2=0.0;
-
+void SUBnote::initfilter(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE amp,REALTYPE mag)
+{
+ filter.xn1=0.0;
+ filter.xn2=0.0;
+
if (start==0) {
- filter.yn1=0.0;
- filter.yn2=0.0;
+ filter.yn1=0.0;
+ filter.yn2=0.0;
} else {
- REALTYPE a=0.1*mag;//empirically
+ REALTYPE a=0.1*mag;//empirically
REALTYPE p=RND*2.0*PI;
- if (start==1) a*=RND;
- filter.yn1=a*cos(p);
- filter.yn2=a*cos(p+freq*2.0*PI/SAMPLE_RATE);
-
- //correct the error of computation the start amplitude
- //at very high frequencies
- if (freq>SAMPLE_RATE*0.96) {
- filter.yn1=0.0;
- filter.yn2=0.0;
-
- };
+ if (start==1) a*=RND;
+ filter.yn1=a*cos(p);
+ filter.yn2=a*cos(p+freq*2.0*PI/SAMPLE_RATE);
+
+ //correct the error of computation the start amplitude
+ //at very high frequencies
+ if (freq>SAMPLE_RATE*0.96) {
+ filter.yn1=0.0;
+ filter.yn2=0.0;
+
+ };
};
- filter.amp=amp;
+ filter.amp=amp;
filter.freq=freq;
filter.bw=bw;
computefiltercoefs(filter,freq,bw,1.0);
@@ -379,17 +406,18 @@ void SUBnote::initfilter(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE amp
/*
* Do the filtering
*/
-void SUBnote::filter(bpfilter &filter,REALTYPE *smps){
+void SUBnote::filter(bpfilter &filter,REALTYPE *smps)
+{
int i;
REALTYPE out;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- out=smps[i] * filter.b0 + filter.b2 * filter.xn2
- -filter.a1 * filter.yn1 - filter.a2 * filter.yn2;
- filter.xn2=filter.xn1;
- filter.xn1=smps[i];
- filter.yn2=filter.yn1;
- filter.yn1=out;
- smps[i]=out;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ out=smps[i] * filter.b0 + filter.b2 * filter.xn2
+ -filter.a1 * filter.yn1 - filter.a2 * filter.yn2;
+ filter.xn2=filter.xn1;
+ filter.xn1=smps[i];
+ filter.yn2=filter.yn1;
+ filter.yn1=out;
+ smps[i]=out;
};
};
@@ -397,18 +425,19 @@ void SUBnote::filter(bpfilter &filter,REALTYPE *smps){
/*
* Init Parameters
*/
-void SUBnote::initparameters(REALTYPE freq){
+void SUBnote::initparameters(REALTYPE freq)
+{
AmpEnvelope=new Envelope(pars->AmpEnvelope,freq);
if (pars->PFreqEnvelopeEnabled!=0) FreqEnvelope=new Envelope(pars->FreqEnvelope,freq);
- else FreqEnvelope=NULL;
+ else FreqEnvelope=NULL;
if (pars->PBandWidthEnvelopeEnabled!=0) BandWidthEnvelope=new Envelope(pars->BandWidthEnvelope,freq);
- else BandWidthEnvelope=NULL;
- if (pars->PGlobalFilterEnabled!=0){
- globalfiltercenterq=pars->GlobalFilter->getq();
- GlobalFilterL=new Filter(pars->GlobalFilter);
- if (stereo!=0) GlobalFilterR=new Filter(pars->GlobalFilter);
- GlobalFilterEnvelope=new Envelope(pars->GlobalFilterEnvelope,freq);
- GlobalFilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq);
+ else BandWidthEnvelope=NULL;
+ if (pars->PGlobalFilterEnabled!=0) {
+ globalfiltercenterq=pars->GlobalFilter->getq();
+ GlobalFilterL=new Filter(pars->GlobalFilter);
+ if (stereo!=0) GlobalFilterR=new Filter(pars->GlobalFilter);
+ GlobalFilterEnvelope=new Envelope(pars->GlobalFilterEnvelope,freq);
+ GlobalFilterFreqTracking=pars->GlobalFilter->getfreqtracking(basefreq);
};
computecurrentparameters();
};
@@ -417,65 +446,68 @@ void SUBnote::initparameters(REALTYPE freq){
/*
* Compute Parameters of SUBnote for each tick
*/
-void SUBnote::computecurrentparameters(){
+void SUBnote::computecurrentparameters()
+{
if ((FreqEnvelope!=NULL)||(BandWidthEnvelope!=NULL)||
- (oldpitchwheel!=ctl->pitchwheel.data)||
- (oldbandwidth!=ctl->bandwidth.data)||
- (portamento!=0)){
- REALTYPE envfreq=1.0;
- REALTYPE envbw=1.0;
- REALTYPE gain=1.0;
-
- if (FreqEnvelope!=NULL) {
- envfreq=FreqEnvelope->envout()/1200;
- envfreq=pow(2.0,envfreq);
- };
- envfreq*=ctl->pitchwheel.relfreq;//pitch wheel
- if (portamento!=0) {//portamento is used
- envfreq*=ctl->portamento.freqrap;
- if (ctl->portamento.used==0){//the portamento has finished
- portamento=0;//this note is no longer "portamented"
- };
- };
-
- if (BandWidthEnvelope!=NULL) {
- envbw=BandWidthEnvelope->envout();
- envbw=pow(2,envbw);
- };
- envbw*=ctl->bandwidth.relbw;//bandwidth controller
-
- REALTYPE tmpgain=1.0/sqrt(envbw*envfreq);
-
- for (int n=0;n<numharmonics;n++){
- for (int nph=0;nph<numstages;nph++) {
- if (nph==0) gain=tmpgain;else gain=1.0;
- computefiltercoefs( lfilter[nph+n*numstages],
- lfilter[nph+n*numstages].freq*envfreq,
- lfilter[nph+n*numstages].bw*envbw,gain);
- };
- };
- if (stereo!=0)
- for (int n=0;n<numharmonics;n++){
- for (int nph=0;nph<numstages;nph++) {
- if (nph==0) gain=tmpgain;else gain=1.0;
- computefiltercoefs( rfilter[nph+n*numstages],
- rfilter[nph+n*numstages].freq*envfreq,
- rfilter[nph+n*numstages].bw*envbw,gain);
- };
- };
- oldbandwidth=ctl->bandwidth.data;
- oldpitchwheel=ctl->pitchwheel.data;
+ (oldpitchwheel!=ctl->pitchwheel.data)||
+ (oldbandwidth!=ctl->bandwidth.data)||
+ (portamento!=0)) {
+ REALTYPE envfreq=1.0;
+ REALTYPE envbw=1.0;
+ REALTYPE gain=1.0;
+
+ if (FreqEnvelope!=NULL) {
+ envfreq=FreqEnvelope->envout()/1200;
+ envfreq=pow(2.0,envfreq);
+ };
+ envfreq*=ctl->pitchwheel.relfreq;//pitch wheel
+ if (portamento!=0) {//portamento is used
+ envfreq*=ctl->portamento.freqrap;
+ if (ctl->portamento.used==0) {//the portamento has finished
+ portamento=0;//this note is no longer "portamented"
+ };
+ };
+
+ if (BandWidthEnvelope!=NULL) {
+ envbw=BandWidthEnvelope->envout();
+ envbw=pow(2,envbw);
+ };
+ envbw*=ctl->bandwidth.relbw;//bandwidth controller
+
+ REALTYPE tmpgain=1.0/sqrt(envbw*envfreq);
+
+ for (int n=0;n<numharmonics;n++) {
+ for (int nph=0;nph<numstages;nph++) {
+ if (nph==0) gain=tmpgain;
+ else gain=1.0;
+ computefiltercoefs( lfilter[nph+n*numstages],
+ lfilter[nph+n*numstages].freq*envfreq,
+ lfilter[nph+n*numstages].bw*envbw,gain);
+ };
+ };
+ if (stereo!=0)
+ for (int n=0;n<numharmonics;n++) {
+ for (int nph=0;nph<numstages;nph++) {
+ if (nph==0) gain=tmpgain;
+ else gain=1.0;
+ computefiltercoefs( rfilter[nph+n*numstages],
+ rfilter[nph+n*numstages].freq*envfreq,
+ rfilter[nph+n*numstages].bw*envbw,gain);
+ };
+ };
+ oldbandwidth=ctl->bandwidth.data;
+ oldpitchwheel=ctl->pitchwheel.data;
};
newamplitude=volume*AmpEnvelope->envout_dB()*2.0;
-
+
//Filter
- if (GlobalFilterL!=NULL){
- REALTYPE globalfilterpitch=GlobalFilterCenterPitch+GlobalFilterEnvelope->envout();
- REALTYPE filterfreq=globalfilterpitch+ctl->filtercutoff.relfreq+GlobalFilterFreqTracking;
- filterfreq=GlobalFilterL->getrealfreq(filterfreq);
-
- GlobalFilterL->setfreq_and_q(filterfreq,globalfiltercenterq*ctl->filterq.relq);
- if (GlobalFilterR!=NULL) GlobalFilterR->setfreq_and_q(filterfreq,globalfiltercenterq*ctl->filterq.relq);
+ if (GlobalFilterL!=NULL) {
+ REALTYPE globalfilterpitch=GlobalFilterCenterPitch+GlobalFilterEnvelope->envout();
+ REALTYPE filterfreq=globalfilterpitch+ctl->filtercutoff.relfreq+GlobalFilterFreqTracking;
+ filterfreq=GlobalFilterL->getrealfreq(filterfreq);
+
+ GlobalFilterL->setfreq_and_q(filterfreq,globalfiltercenterq*ctl->filterq.relq);
+ if (GlobalFilterR!=NULL) GlobalFilterR->setfreq_and_q(filterfreq,globalfiltercenterq*ctl->filterq.relq);
};
};
@@ -483,140 +515,143 @@ void SUBnote::computecurrentparameters(){
/*
* Note Output
*/
-int SUBnote::noteout(REALTYPE *outl,REALTYPE *outr){
+int SUBnote::noteout(REALTYPE *outl,REALTYPE *outr)
+{
int i;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- outl[i]=denormalkillbuf[i];
- outr[i]=denormalkillbuf[i];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]=denormalkillbuf[i];
+ outr[i]=denormalkillbuf[i];
};
-
+
if (NoteEnabled==OFF) return(0);
//left channel
for (i=0;i<SOUND_BUFFER_SIZE;i++) tmprnd[i]=RND*2.0-1.0;
- for (int n=0;n<numharmonics;n++){
- for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpsmp[i]=tmprnd[i];
- for (int nph=0;nph<numstages;nph++)
- filter(lfilter[nph+n*numstages],tmpsmp);
- for (i=0;i<SOUND_BUFFER_SIZE;i++) outl[i]+=tmpsmp[i];
+ for (int n=0;n<numharmonics;n++) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpsmp[i]=tmprnd[i];
+ for (int nph=0;nph<numstages;nph++)
+ filter(lfilter[nph+n*numstages],tmpsmp);
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) outl[i]+=tmpsmp[i];
};
- if (GlobalFilterL!=NULL) GlobalFilterL->filterout(&outl[0]);
-
+ if (GlobalFilterL!=NULL) GlobalFilterL->filterout(&outl[0]);
+
//right channel
- if (stereo!=0){
- for (i=0;i<SOUND_BUFFER_SIZE;i++) tmprnd[i]=RND*2.0-1.0;
- for (int n=0;n<numharmonics;n++){
- for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpsmp[i]=tmprnd[i];
- for (int nph=0;nph<numstages;nph++)
- filter(rfilter[nph+n*numstages],tmpsmp);
- for (i=0;i<SOUND_BUFFER_SIZE;i++) outr[i]+=tmpsmp[i];
- };
- if (GlobalFilterR!=NULL) GlobalFilterR->filterout(&outr[0]);
+ if (stereo!=0) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) tmprnd[i]=RND*2.0-1.0;
+ for (int n=0;n<numharmonics;n++) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) tmpsmp[i]=tmprnd[i];
+ for (int nph=0;nph<numstages;nph++)
+ filter(rfilter[nph+n*numstages],tmpsmp);
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) outr[i]+=tmpsmp[i];
+ };
+ if (GlobalFilterR!=NULL) GlobalFilterR->filterout(&outr[0]);
} else for (i=0;i<SOUND_BUFFER_SIZE;i++) outr[i]=outl[i];
-
- if (firsttick!=0){
- int n=10;if (n>SOUND_BUFFER_SIZE) n=SOUND_BUFFER_SIZE;
- for (i=0;i<n;i++) {
- REALTYPE ampfadein=0.5-0.5*cos((REALTYPE) i/(REALTYPE) n*PI);
- outl[i]*=ampfadein;
- outr[i]*=ampfadein;
- };
- firsttick=0;
+
+ if (firsttick!=0) {
+ int n=10;
+ if (n>SOUND_BUFFER_SIZE) n=SOUND_BUFFER_SIZE;
+ for (i=0;i<n;i++) {
+ REALTYPE ampfadein=0.5-0.5*cos((REALTYPE) i/(REALTYPE) n*PI);
+ outl[i]*=ampfadein;
+ outr[i]*=ampfadein;
+ };
+ firsttick=0;
};
- if (ABOVE_AMPLITUDE_THRESHOLD(oldamplitude,newamplitude)){
- // Amplitude interpolation
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(oldamplitude
- ,newamplitude,i,SOUND_BUFFER_SIZE);
- outl[i]*=tmpvol*panning;
- outr[i]*=tmpvol*(1.0-panning);
+ if (ABOVE_AMPLITUDE_THRESHOLD(oldamplitude,newamplitude)) {
+ // Amplitude interpolation
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(oldamplitude
+ ,newamplitude,i,SOUND_BUFFER_SIZE);
+ outl[i]*=tmpvol*panning;
+ outr[i]*=tmpvol*(1.0-panning);
};
} else {
- for (i=0;i<SOUND_BUFFER_SIZE;i++) {
- outl[i]*=newamplitude*panning;
- outr[i]*=newamplitude*(1.0-panning);
- };
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]*=newamplitude*panning;
+ outr[i]*=newamplitude*(1.0-panning);
+ };
};
- oldamplitude=newamplitude;
+ oldamplitude=newamplitude;
computecurrentparameters();
// Apply legato-specific sound signal modifications
- if (Legato.silent){ // Silencer
- if (Legato.msg!=LM_FadeIn){
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- outl[i]=0.0;
- outr[i]=0.0;
- }
- }
+ if (Legato.silent) { // Silencer
+ if (Legato.msg!=LM_FadeIn) {
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ outl[i]=0.0;
+ outr[i]=0.0;
+ }
+ }
}
- switch (Legato.msg){
+ switch (Legato.msg) {
case LM_CatchUp : // Continue the catch-up...
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){//Yea, could be done without the loop...
- Legato.decounter--;
- if (Legato.decounter<1){
- // Catching-up done, we can finally set
- // the note to the actual parameters.
- Legato.decounter=-10;
- Legato.msg=LM_ToNorm;
- SUBlegatonote(Legato.param.freq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
- break;
- }
- }
- break;
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {//Yea, could be done without the loop...
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ // Catching-up done, we can finally set
+ // the note to the actual parameters.
+ Legato.decounter=-10;
+ Legato.msg=LM_ToNorm;
+ SUBlegatonote(Legato.param.freq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
+ break;
+ }
+ }
+ break;
case LM_FadeIn : // Fade-in
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- Legato.silent=false;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- Legato.decounter--;
- if (Legato.decounter<1){
- Legato.decounter=-10;
- Legato.msg=LM_Norm;
- break;
- }
- Legato.fade.m+=Legato.fade.step;
- outl[i]*=Legato.fade.m;
- outr[i]*=Legato.fade.m;
- }
- break;
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ Legato.silent=false;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ Legato.decounter=-10;
+ Legato.msg=LM_Norm;
+ break;
+ }
+ Legato.fade.m+=Legato.fade.step;
+ outl[i]*=Legato.fade.m;
+ outr[i]*=Legato.fade.m;
+ }
+ break;
case LM_FadeOut : // Fade-out, then set the catch-up
- if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
- for (i=0;i<SOUND_BUFFER_SIZE;i++){
- Legato.decounter--;
- if (Legato.decounter<1){
- for (int j=i;j<SOUND_BUFFER_SIZE;j++){
- outl[j]=0.0;
- outr[j]=0.0;
- }
- Legato.decounter=-10;
- Legato.silent=true;
- // Fading-out done, now set the catch-up :
- Legato.decounter=Legato.fade.length;
- Legato.msg=LM_CatchUp;
- REALTYPE catchupfreq=Legato.param.freq*(Legato.param.freq/Legato.lastfreq);//This freq should make this now silent note to catch-up (or should I say resync ?) with the heard note for the same length it stayed at the previous freq during the fadeout.
- SUBlegatonote(catchupfreq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
- break;
- }
- Legato.fade.m-=Legato.fade.step;
- outl[i]*=Legato.fade.m;
- outr[i]*=Legato.fade.m;
- }
- break;
- default : break;
+ if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
+ for (i=0;i<SOUND_BUFFER_SIZE;i++) {
+ Legato.decounter--;
+ if (Legato.decounter<1) {
+ for (int j=i;j<SOUND_BUFFER_SIZE;j++) {
+ outl[j]=0.0;
+ outr[j]=0.0;
+ }
+ Legato.decounter=-10;
+ Legato.silent=true;
+ // Fading-out done, now set the catch-up :
+ Legato.decounter=Legato.fade.length;
+ Legato.msg=LM_CatchUp;
+ REALTYPE catchupfreq=Legato.param.freq*(Legato.param.freq/Legato.lastfreq);//This freq should make this now silent note to catch-up (or should I say resync ?) with the heard note for the same length it stayed at the previous freq during the fadeout.
+ SUBlegatonote(catchupfreq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
+ break;
+ }
+ Legato.fade.m-=Legato.fade.step;
+ outl[i]*=Legato.fade.m;
+ outr[i]*=Legato.fade.m;
+ }
+ break;
+ default :
+ break;
}
// Check if the note needs to be computed more
- if (AmpEnvelope->finished()!=0){
+ if (AmpEnvelope->finished()!=0) {
for (i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out
- REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
- outl[i]*=tmp;
- outr[i]*=tmp;
- };
- KillNote();
+ REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
+ outl[i]*=tmp;
+ outr[i]*=tmp;
+ };
+ KillNote();
};
return(1);
};
@@ -624,7 +659,8 @@ int SUBnote::noteout(REALTYPE *outl,REALTYPE *outr){
/*
* Relase Key (Note Off)
*/
-void SUBnote::relasekey(){
+void SUBnote::relasekey()
+{
AmpEnvelope->relasekey();
if (FreqEnvelope!=NULL) FreqEnvelope->relasekey();
if (BandWidthEnvelope!=NULL) BandWidthEnvelope->relasekey();
@@ -634,8 +670,9 @@ void SUBnote::relasekey(){
/*
* Check if the note is finished
*/
-int SUBnote::finished(){
- if (NoteEnabled==OFF) return(1);
- else return(0);
+int SUBnote::finished()
+{
+ if (NoteEnabled==OFF) return(1);
+ else return(0);
};
diff --git a/src/Synth/SUBnote.h b/src/Synth/SUBnote.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
SUBnote.h - The subtractive synthesizer
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,85 +29,86 @@
#include "Envelope.h"
#include "../DSP/Filter.h"
-class SUBnote{
- public:
- SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote,bool besilent);
- ~SUBnote();
-
- void SUBlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool externcall);
-
- int noteout(REALTYPE *outl,REALTYPE *outr);//note output,return 0 if the note is finished
- void relasekey();
- int finished();
-
- int ready; //if I can get the sampledata
-
- private:
-
- void computecurrentparameters();
- void initparameters(REALTYPE freq);
- void KillNote();
-
- 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
- REALTYPE basefreq;
- REALTYPE panning;
- Envelope *AmpEnvelope;
- Envelope *FreqEnvelope;
- Envelope *BandWidthEnvelope;
-
- Filter *GlobalFilterL,*GlobalFilterR;
-
- Envelope *GlobalFilterEnvelope;
-
- //internal values
- ONOFFTYPE NoteEnabled;
- int firsttick,portamento;
- REALTYPE volume,oldamplitude,newamplitude;
-
- REALTYPE GlobalFilterCenterPitch;//octaves
- REALTYPE GlobalFilterFreqTracking;
-
- struct bpfilter{
- REALTYPE freq,bw,amp; //filter parameters
- REALTYPE a1,a2,b0,b2;//filter coefs. b1=0
- REALTYPE xn1,xn2,yn1,yn2; //filter internal values
- };
-
- void initfilter(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE amp,REALTYPE mag);
- void computefiltercoefs(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE gain);
- void filter(bpfilter &filter,REALTYPE *smps);
-
- bpfilter *lfilter,*rfilter;
-
- REALTYPE *tmpsmp;
- REALTYPE *tmprnd;//this is filled with random numbers
-
- Controller *ctl;
- int oldpitchwheel,oldbandwidth;
- REALTYPE globalfiltercenterq;
-
- // Legato vars
- struct {
- bool silent;
- REALTYPE lastfreq;
- LegatoMsg msg;
- int decounter;
- struct { // Fade In/Out vars
- int length;
- REALTYPE m, step;
- } fade;
- struct { // Note parameters
- REALTYPE freq, vel;
- int portamento, midinote;
- } param;
- } Legato;
+class SUBnote
+{
+public:
+ SUBnote(SUBnoteParameters *parameters,Controller *ctl_,REALTYPE freq,REALTYPE velocity,int portamento_,int midinote,bool besilent);
+ ~SUBnote();
+
+ void SUBlegatonote(REALTYPE freq, REALTYPE velocity, int portamento_, int midinote, bool externcall);
+
+ int noteout(REALTYPE *outl,REALTYPE *outr);//note output,return 0 if the note is finished
+ void relasekey();
+ int finished();
+
+ int ready; //if I can get the sampledata
+
+private:
+
+ void computecurrentparameters();
+ void initparameters(REALTYPE freq);
+ void KillNote();
+
+ 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
+ REALTYPE basefreq;
+ REALTYPE panning;
+ Envelope *AmpEnvelope;
+ Envelope *FreqEnvelope;
+ Envelope *BandWidthEnvelope;
+
+ Filter *GlobalFilterL,*GlobalFilterR;
+
+ Envelope *GlobalFilterEnvelope;
+
+ //internal values
+ ONOFFTYPE NoteEnabled;
+ int firsttick,portamento;
+ REALTYPE volume,oldamplitude,newamplitude;
+
+ REALTYPE GlobalFilterCenterPitch;//octaves
+ REALTYPE GlobalFilterFreqTracking;
+
+ struct bpfilter {
+ REALTYPE freq,bw,amp; //filter parameters
+ REALTYPE a1,a2,b0,b2;//filter coefs. b1=0
+ REALTYPE xn1,xn2,yn1,yn2; //filter internal values
+ };
+
+ void initfilter(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE amp,REALTYPE mag);
+ void computefiltercoefs(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE gain);
+ void filter(bpfilter &filter,REALTYPE *smps);
+
+ bpfilter *lfilter,*rfilter;
+
+ REALTYPE *tmpsmp;
+ REALTYPE *tmprnd;//this is filled with random numbers
+
+ Controller *ctl;
+ int oldpitchwheel,oldbandwidth;
+ REALTYPE globalfiltercenterq;
+
+ // Legato vars
+ struct {
+ bool silent;
+ REALTYPE lastfreq;
+ LegatoMsg msg;
+ int decounter;
+ struct { // Fade In/Out vars
+ int length;
+ REALTYPE m, step;
+ } fade;
+ struct { // Note parameters
+ REALTYPE freq, vel;
+ int portamento, midinote;
+ } param;
+ } Legato;
};
diff --git a/src/Tests/EchoTest.h b/src/Tests/EchoTest.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
EchoTest.h - CxxTest for Effect/Echo
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -29,101 +29,98 @@ int SAMPLE_RATE=1024;
class EchoTest : public CxxTest::TestSuite
{
- public:
- void setUp()
- {
- outL=new float[SOUND_BUFFER_SIZE];
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(outL+i)=0;
- outR=new float[SOUND_BUFFER_SIZE];
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(outR+i)=0;
- inL=new float[SOUND_BUFFER_SIZE];
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(inL+i)=0;
- inR=new float[SOUND_BUFFER_SIZE];
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(inR+i)=0;
- testFX=new Echo(true,outL,outR);
- }
+public:
+ void setUp() {
+ outL=new float[SOUND_BUFFER_SIZE];
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(outL+i)=0;
+ outR=new float[SOUND_BUFFER_SIZE];
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(outR+i)=0;
+ inL=new float[SOUND_BUFFER_SIZE];
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(inL+i)=0;
+ inR=new float[SOUND_BUFFER_SIZE];
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(inR+i)=0;
+ testFX=new Echo(true,outL,outR);
+ }
- void tearDown()
- {
- delete[] inL;
- delete[] inR;
- delete[] outL;
- delete[] outR;
- delete testFX;
- }
+ void tearDown() {
+ delete[] inL;
+ delete[] inR;
+ delete[] outL;
+ delete[] outR;
+ delete testFX;
+ }
-
- void testInit()
- {
- //Make sure that the output will be zero at start
- //(given a zero input)
- testFX->out(inL,inR);
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- TS_ASSERT_EQUALS(outL[i],0.0);
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- TS_ASSERT_EQUALS(outR[i],0.0);
- }
- void testClear()
- {
- char DELAY=2;
- testFX->changepar(DELAY,127);
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(inL+i)=1.0;
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(inR+i)=1.0;
- for(int i=0;i<50;++i)
- testFX->out(inL,inR);
- for(int i=0;i<SOUND_BUFFER_SIZE;++i){
- TS_ASSERT_DIFFERS(outL[i],0.0);
- TS_ASSERT_DIFFERS(outR[i],0.0)}
- //After making sure the internal buffer has a nonzero value
- //cleanup
- //Then get the next output, which should be zereoed out if DELAY
- //is large enough
- testFX->cleanup();
+ void testInit() {
+ //Make sure that the output will be zero at start
+ //(given a zero input)
+ testFX->out(inL,inR);
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ TS_ASSERT_EQUALS(outL[i],0.0);
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ TS_ASSERT_EQUALS(outR[i],0.0);
+ }
+
+ void testClear() {
+ char DELAY=2;
+ testFX->changepar(DELAY,127);
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(inL+i)=1.0;
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(inR+i)=1.0;
+ for (int i=0;i<50;++i)
testFX->out(inL,inR);
- for(int i=0;i<SOUND_BUFFER_SIZE;++i){
- TS_ASSERT_DELTA(outL[i],0.0,0.0001);
- TS_ASSERT_DELTA(outR[i],0.0,0.0001);
- }
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i) {
+ TS_ASSERT_DIFFERS(outL[i],0.0);
+ TS_ASSERT_DIFFERS(outR[i],0.0)
+ }
+ //After making sure the internal buffer has a nonzero value
+ //cleanup
+ //Then get the next output, which should be zereoed out if DELAY
+ //is large enough
+ testFX->cleanup();
+ testFX->out(inL,inR);
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i) {
+ TS_ASSERT_DELTA(outL[i],0.0,0.0001);
+ TS_ASSERT_DELTA(outR[i],0.0,0.0001);
}
- //Insures that the proper decay occurs with high feedback
- void testDecaywFb()
- {
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(inL+i)=1.0;
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(inR+i)=1.0;
- char FEEDBACK=5;
- testFX->changepar(FEEDBACK,127);
- for(int i=0;i<4;++i)
- testFX->out(inL,inR);
- for(int i=0;i<SOUND_BUFFER_SIZE;++i){
- TS_ASSERT_DIFFERS(outL[i],0.0);
- TS_ASSERT_DIFFERS(outR[i],0.0)}
- float amp=abs(outL[0]+outR[0])/2;
- //reset input to zero
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(inL+i)=0.0;
- for(int i=0;i<SOUND_BUFFER_SIZE;++i)
- *(inR+i)=0.0;
- //give the echo time to fade based upon zero input and high feedback
- for(int i=0;i<50;++i)
- testFX->out(inL,inR);
- TS_ASSERT_LESS_THAN(abs(outL[0]+outR[0])/2, amp);
+ }
+ //Insures that the proper decay occurs with high feedback
+ void testDecaywFb() {
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(inL+i)=1.0;
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(inR+i)=1.0;
+ char FEEDBACK=5;
+ testFX->changepar(FEEDBACK,127);
+ for (int i=0;i<4;++i)
+ testFX->out(inL,inR);
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i) {
+ TS_ASSERT_DIFFERS(outL[i],0.0);
+ TS_ASSERT_DIFFERS(outR[i],0.0)
}
+ float amp=abs(outL[0]+outR[0])/2;
+ //reset input to zero
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(inL+i)=0.0;
+ for (int i=0;i<SOUND_BUFFER_SIZE;++i)
+ *(inR+i)=0.0;
+ //give the echo time to fade based upon zero input and high feedback
+ for (int i=0;i<50;++i)
+ testFX->out(inL,inR);
+ TS_ASSERT_LESS_THAN(abs(outL[0]+outR[0])/2, amp);
+ }
+
-
- private:
- float *inL,*inR,*outR,*outL;
- Echo *testFX;
+private:
+ float *inL,*inR,*outR,*outL;
+ Echo *testFX;
};
diff --git a/src/Tests/SampleTest.h b/src/Tests/SampleTest.h
@@ -1,12 +1,12 @@
/*
ZynAddSubFX - a software synthesizer
-
+
SampleTest.h - CxxTest for Samples
Copyright (C) 2009-2009 Mark McCurry
Author: Mark McCurry
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -24,54 +24,49 @@
class SampleTest : public CxxTest::TestSuite
{
- public:
- void testInit()
- {
- AuSample smp(10);
- TS_ASSERT_EQUALS(smp.size(),10);
- for(int i=0;i<20;++i)
- TS_ASSERT_EQUALS(smp[i],0.0);
- }
+public:
+ void testInit() {
+ AuSample smp(10);
+ TS_ASSERT_EQUALS(smp.size(),10);
+ for (int i=0;i<20;++i)
+ TS_ASSERT_EQUALS(smp[i],0.0);
+ }
- void testAssign()
- {
- AuSample smp(3);
- smp[0]=0;
- smp[1]=1;
- smp[2]=2;
- AuSample nsmp(40);
- nsmp=smp;
- TS_ASSERT_EQUALS(smp.size(),nsmp.size());
- for(int i=0;i<29;++i)
- TS_ASSERT_EQUALS(smp[i],nsmp[i]);
- }
- void testBounds()
- {
- AuSample smp(0);
- TS_ASSERT(smp.size()!=0);
- }
+ void testAssign() {
+ AuSample smp(3);
+ smp[0]=0;
+ smp[1]=1;
+ smp[2]=2;
+ AuSample nsmp(40);
+ nsmp=smp;
+ TS_ASSERT_EQUALS(smp.size(),nsmp.size());
+ for (int i=0;i<29;++i)
+ TS_ASSERT_EQUALS(smp[i],nsmp[i]);
+ }
+ void testBounds() {
+ AuSample smp(0);
+ TS_ASSERT(smp.size()!=0);
+ }
- void testAllocDealloc()
- {
- float * fl=new float[50];
- for(int i=0;i<50;++i)
- *(fl+i)=i;
- AuSample smp(2);
- smp = AuSample(fl, 50);
- delete [] fl;
- for(int i=0;i<50;++i)
- TS_ASSERT_DELTA(smp[i],i,0.001);
- smp = AuSample(3);
- }
+ void testAllocDealloc() {
+ float * fl=new float[50];
+ for (int i=0;i<50;++i)
+ *(fl+i)=i;
+ AuSample smp(2);
+ smp = AuSample(fl, 50);
+ delete [] fl;
+ for (int i=0;i<50;++i)
+ TS_ASSERT_DELTA(smp[i],i,0.001);
+ smp = AuSample(3);
+ }
- void testClear()
- {
- AuSample smp(50);
- for(int i=0;i<50;++i)
- smp[i]=10;
- smp.clear();
- for(int i=0;i<50;++i)
- TS_ASSERT_EQUALS(smp[i],0);
- }
+ void testClear() {
+ AuSample smp(50);
+ for (int i=0;i<50;++i)
+ smp[i]=10;
+ smp.clear();
+ for (int i=0;i<50;++i)
+ TS_ASSERT_EQUALS(smp[i],0);
+ }
};
diff --git a/src/globals.h b/src/globals.h
@@ -1,13 +1,13 @@
/*
ZynAddSubFX - a software synthesizer
- globals.h - it contains program settings and the program capabilities
+ globals.h - it contains program settings and the program capabilities
like number of parts, of effects
Copyright (C) 2002-2005 Nasca Octavian Paul
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -28,7 +28,7 @@
//What float type I use for internal sampledata
#define REALTYPE float
-struct FFTFREQS{
+struct FFTFREQS {
REALTYPE *s,*c;//sine and cosine components
};
@@ -38,12 +38,12 @@ extern void deleteFFTFREQS(FFTFREQS *f);
/**Sampling rate*/
extern int SAMPLE_RATE;
-/**
+/**
* The size of a sound buffer (or the granularity)
* All internal transfer of sound data use buffer of this size
* All parameters are constant during this period of time, exception
* some parameters(like amplitudes) which are linear interpolated.
- * If you increase this you'll ecounter big latencies, but if you
+ * If you increase this you'll ecounter big latencies, but if you
* decrease this the CPU requirements gets high.
*/
extern int SOUND_BUFFER_SIZE;
@@ -107,7 +107,7 @@ extern int OSCIL_SIZE;
#define NUM_INS_EFX 8
/*
- * Number of part's insertion effects
+ * Number of part's insertion effects
*/
#define NUM_PART_EFX 3
@@ -142,7 +142,7 @@ extern int OSCIL_SIZE;
#define MAX_FILTER_STAGES 5
/*
- * Formant filter (FF) limits
+ * Formant filter (FF) limits
*/
#define FF_MAX_VOWELS 6
#define FF_MAX_FORMANTS 12
@@ -173,7 +173,7 @@ extern int OSCIL_SIZE;
/*
- * dB
+ * dB
*/
#define dB2rap(dB) ((exp((dB)*LOG_10/20.0)))
#define rap2dB(rap) ((20*log(rap)/LOG_10))
@@ -185,16 +185,17 @@ extern int OSCIL_SIZE;
#define ZERO(data,size) {char *data_=(char *) data;for (int i=0;i<size;i++) data_[i]=0;};
-enum ONOFFTYPE{OFF=0,ON=1};
+enum ONOFFTYPE {OFF=0,ON=1};
-enum MidiControllers{C_NULL=0,C_pitchwheel=1000,C_expression=11,C_panning=10,
- C_filtercutoff=74,C_filterq=71,C_bandwidth=75,C_modwheel=1,C_fmamp=76,
- C_volume=7,C_sustain=64,C_allnotesoff=123,C_allsoundsoff=120,C_resetallcontrollers=121,
- C_portamento=65,C_resonance_center=77,C_resonance_bandwidth=78,
+enum MidiControllers {C_NULL=0,C_pitchwheel=1000,C_expression=11,C_panning=10,
+ C_filtercutoff=74,C_filterq=71,C_bandwidth=75,C_modwheel=1,C_fmamp=76,
+ C_volume=7,C_sustain=64,C_allnotesoff=123,C_allsoundsoff=120,C_resetallcontrollers=121,
+ C_portamento=65,C_resonance_center=77,C_resonance_bandwidth=78,
- C_dataentryhi=0x06,C_dataentrylo=0x26,C_nrpnhi=99,C_nrpnlo=98};
+ C_dataentryhi=0x06,C_dataentrylo=0x26,C_nrpnhi=99,C_nrpnlo=98
+ };
-enum LegatoMsg{LM_Norm, LM_FadeIn, LM_FadeOut, LM_CatchUp, LM_ToNorm};
+enum LegatoMsg {LM_Norm, LM_FadeIn, LM_FadeOut, LM_CatchUp, LM_ToNorm};
//is like i=(int)(floor(f))
#ifdef ASM_F2I_YES
@@ -206,7 +207,7 @@ enum LegatoMsg{LM_Norm, LM_FadeIn, LM_FadeOut, LM_CatchUp, LM_ToNorm};
#ifndef O_BINARY
-#define O_BINARY 0
+#define O_BINARY 0
#endif
#endif
diff --git a/src/main.C b/src/main.C
@@ -6,7 +6,7 @@
Author: Nasca Octavian Paul
This program is free software; you can redistribute it and/or modify
- it under the terms of version 2 of the GNU General Public License
+ it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
@@ -94,10 +94,11 @@ int Pexitprogram=0;//if the UI set this to 1, the program will exit
/*
* Try to get the realtime priority
*/
-void set_realtime(){
+void set_realtime()
+{
#ifdef OS_LINUX
sched_param sc;
-
+
sc.sched_priority=50;
//if you want get "sched_setscheduler undeclared" from compilation, you can safely remove the folowing line
@@ -110,28 +111,29 @@ void set_realtime(){
* Midi input thread
*/
#if !(defined(WINMIDIIN)||defined(VSTMIDIIN))
-void *thread1(void *arg){
+void *thread1(void *arg)
+{
MidiCmdType cmdtype=MidiNoteOFF;
unsigned char cmdchan=0,note=0,vel=0;
int cmdparams[MP_MAX_BYTES];
- for(int i=0;i<MP_MAX_BYTES;++i)
+ for (int i=0;i<MP_MAX_BYTES;++i)
cmdparams[i]=0;
set_realtime();
- while (Pexitprogram==0){
- Midi->getmidicmd(cmdtype,cmdchan,cmdparams);
+ while (Pexitprogram==0) {
+ Midi->getmidicmd(cmdtype,cmdchan,cmdparams);
note=cmdparams[0];
- vel=cmdparams[1];
-
- pthread_mutex_lock(&master->mutex);
+ vel=cmdparams[1];
- if ((cmdtype==MidiNoteON)&&(note!=0)) master->NoteOn(cmdchan,note,vel);
- if ((cmdtype==MidiNoteOFF)&&(note!=0)) master->NoteOff(cmdchan,note);
- if (cmdtype==MidiController) master->SetController(cmdchan,cmdparams[0],cmdparams[1]);
+ pthread_mutex_lock(&master->mutex);
- pthread_mutex_unlock(&master->mutex);
+ if ((cmdtype==MidiNoteON)&&(note!=0)) master->NoteOn(cmdchan,note,vel);
+ if ((cmdtype==MidiNoteOFF)&&(note!=0)) master->NoteOff(cmdchan,note);
+ if (cmdtype==MidiController) master->SetController(cmdchan,cmdparams[0],cmdparams[1]);
+
+ pthread_mutex_unlock(&master->mutex);
};
-
+
return(0);
};
#endif
@@ -142,39 +144,40 @@ void *thread1(void *arg){
#if defined(OSSAUDIOOUT)
//!(defined(JACKAUDIOOUT)||defined(JACK_RTAUDIOOUT)||defined(PAAUDIOOUT)||defined(VSTAUDIOOUT))
-void *thread2(void *arg){
+void *thread2(void *arg)
+{
REALTYPE outputl[SOUND_BUFFER_SIZE];
REALTYPE outputr[SOUND_BUFFER_SIZE];
set_realtime();
- while (Pexitprogram==0){
- pthread_mutex_lock(&master->mutex);
- master->AudioOut(outputl,outputr);
- pthread_mutex_unlock(&master->mutex);
+ while (Pexitprogram==0) {
+ pthread_mutex_lock(&master->mutex);
+ master->AudioOut(outputl,outputr);
+ pthread_mutex_unlock(&master->mutex);
#ifndef NONEAUDIOOUT
- audioout->OSSout(outputl,outputr);
-#endif
-
-/** / int i,x,x2;
- REALTYPE xx,xx2;
-
- short int xsmps[SOUND_BUFFER_SIZE*2];
- for (i=0;i<SOUND_BUFFER_SIZE;i++){//output to stdout
- xx=-outputl[i]*32767;
- xx2=-outputr[i]*32767;
- if (xx<-32768) xx=-32768;
- if (xx>32767) xx=32767;
- if (xx2<-32768) xx2=-32768;
- if (xx2>32767) xx2=32767;
- x=(short int) xx;
- x2=(short int) xx2;
- xsmps[i*2]=x;xsmps[i*2+1]=x2;
- };
- write(1,&xsmps,SOUND_BUFFER_SIZE*2*2);
-
- / * */
- };
+ audioout->OSSout(outputl,outputr);
+#endif
+
+ /** / int i,x,x2;
+ REALTYPE xx,xx2;
+
+ short int xsmps[SOUND_BUFFER_SIZE*2];
+ for (i=0;i<SOUND_BUFFER_SIZE;i++){//output to stdout
+ xx=-outputl[i]*32767;
+ xx2=-outputr[i]*32767;
+ if (xx<-32768) xx=-32768;
+ if (xx>32767) xx=32767;
+ if (xx2<-32768) xx2=-32768;
+ if (xx2>32767) xx2=32767;
+ x=(short int) xx;
+ x2=(short int) xx2;
+ xsmps[i*2]=x;xsmps[i*2+1]=x2;
+ };
+ write(1,&xsmps,SOUND_BUFFER_SIZE*2*2);
+
+ / * */
+ };
return(0);
};
#endif
@@ -182,30 +185,31 @@ void *thread2(void *arg){
/*
* User Interface thread
*/
-
-
-void *thread3(void *arg){
+
+
+void *thread3(void *arg)
+{
#ifndef DISABLE_GUI
ui->showUI();
while (Pexitprogram==0) {
#ifdef USE_LASH
- string filename;
- switch (lash->checkevents(filename)) {
- case LASHClient::Save:
- ui->do_save_master(filename);
- lash->confirmevent(LASHClient::Save);
- break;
- case LASHClient::Restore:
- ui->do_load_master(filename);
- lash->confirmevent(LASHClient::Restore);
- break;
- case LASHClient::Quit:
- Pexitprogram = 1;
- default:
- break;
- }
-#endif
- Fl::wait();
+ string filename;
+ switch (lash->checkevents(filename)) {
+ case LASHClient::Save:
+ ui->do_save_master(filename);
+ lash->confirmevent(LASHClient::Save);
+ break;
+ case LASHClient::Restore:
+ ui->do_load_master(filename);
+ lash->confirmevent(LASHClient::Restore);
+ break;
+ case LASHClient::Quit:
+ Pexitprogram = 1;
+ default:
+ break;
+ }
+#endif
+ Fl::wait();
}
#endif
return(0);
@@ -214,40 +218,41 @@ void *thread3(void *arg){
/*
* Sequencer thread (test)
*/
-void *thread4(void *arg){
- while (Pexitprogram==0){
- int type,par1,par2,again,midichan;
- for (int ntrack=0;ntrack<NUM_MIDI_TRACKS;ntrack++){
- if (master->seq.play==0) break;
- do{
- again=master->seq.getevent(ntrack,&midichan,&type,&par1,&par2);
+void *thread4(void *arg)
+{
+ while (Pexitprogram==0) {
+ int type,par1,par2,again,midichan;
+ for (int ntrack=0;ntrack<NUM_MIDI_TRACKS;ntrack++) {
+ if (master->seq.play==0) break;
+ do {
+ again=master->seq.getevent(ntrack,&midichan,&type,&par1,&par2);
// printf("ntrack=%d again=%d\n",ntrack,again);
- if (type>0) {
+ if (type>0) {
// printf("%d %d %d %d %d\n",type,midichan,chan,par1,par2);
// if (cmdtype==MidiController) master->SetController(cmdchan,cmdparams[0],cmdparams[1]);
-
-
- pthread_mutex_lock(&master->mutex);
- if (type==1){//note_on or note_off
- if (par2!=0) master->NoteOn(midichan,par1,par2);
- else master->NoteOff(midichan,par1);
- };
- pthread_mutex_unlock(&master->mutex);
- };
- } while (again>0);
-
- };
+
+
+ pthread_mutex_lock(&master->mutex);
+ if (type==1) {//note_on or note_off
+ if (par2!=0) master->NoteOn(midichan,par1,par2);
+ else master->NoteOff(midichan,par1);
+ };
+ pthread_mutex_unlock(&master->mutex);
+ };
+ } while (again>0);
+
+ };
//if (!realtime player) atunci fac asta
//!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
#ifdef OS_LINUX
- usleep(1000);
+ usleep(1000);
#elif OS_WINDOWS
- Sleep(1);
+ Sleep(1);
#endif
};
-
+
return(0);
};
@@ -256,7 +261,8 @@ void *thread4(void *arg){
*/
-void initprogram(){
+void initprogram()
+{
cerr.precision(1);
cerr << std::fixed;
#ifndef JACKAUDIOOUT
@@ -272,27 +278,27 @@ void initprogram(){
srand(time(NULL));
denormalkillbuf=new REALTYPE [SOUND_BUFFER_SIZE];
for (int i=0;i<SOUND_BUFFER_SIZE;i++) denormalkillbuf[i]=(RND-0.5)*1e-16;
-
+
OscilGen::tmpsmps=new REALTYPE[OSCIL_SIZE];
newFFTFREQS(&OscilGen::outoscilFFTfreqs,OSCIL_SIZE/2);
-
+
master=new Master();
master->swaplr=swaplr;
#if defined(JACKAUDIOOUT)
if (usejackit) {
- bool tmp=JACKaudiooutputinit(master);
+ bool tmp=JACKaudiooutputinit(master);
#if defined(OSSAUDIOOUT)
- if (!tmp) cout << "\nUsing OSS instead." << endl;
+ if (!tmp) cout << "\nUsing OSS instead." << endl;
#else
- if (!tmp) exit(1);
+ if (!tmp) exit(1);
#endif
- usejackit=tmp;
+ usejackit=tmp;
};
#endif
#if defined(OSSAUDIOOUT)
if (!usejackit) audioout=new OSSaudiooutput();
- else audioout=NULL;
+ else audioout=NULL;
#endif
#ifdef JACK_RTAUDIOOUT
@@ -300,7 +306,7 @@ void initprogram(){
#endif
#ifdef PAAUDIOOUT
PAaudiooutputinit(master);
-#endif
+#endif
#ifdef ALSAMIDIIN
Midi=new ALSAMidiIn();
@@ -319,7 +325,8 @@ void initprogram(){
/*
* Program exit
*/
-void exitprogram(){
+void exitprogram()
+{
pthread_mutex_lock(&master->mutex);
#ifdef OSSAUDIOOUT
delete(audioout);
@@ -338,7 +345,7 @@ void exitprogram(){
delete(ui);
#endif
delete(Midi);
- delete(master);
+ delete(master);
#ifdef USE_LASH
delete(lash);
@@ -353,30 +360,32 @@ void exitprogram(){
#ifdef OS_WINDOWS
#define ARGSIZE 100
- char winoptarguments[ARGSIZE];
- char getopt(int argc, char *argv[], const char *shortopts, int *index){
+char winoptarguments[ARGSIZE];
+char getopt(int argc, char *argv[], const char *shortopts, int *index)
+{
winoptarguments[0]=0;
char result=0;
-
+
if (*index>=argc) return(-1);
-
- if (strlen(argv[*index])==2)
- if (argv[*index][0]=='-') {
- result=argv[*index][1];
- if (*index+1<argc) {
- snprintf(winoptarguments,ARGSIZE,"%s",argv[*index+1]);
- };
- };
+
+ if (strlen(argv[*index])==2)
+ if (argv[*index][0]=='-') {
+ result=argv[*index][1];
+ if (*index+1<argc) {
+ snprintf(winoptarguments,ARGSIZE,"%s",argv[*index+1]);
+ };
+ };
(*index)++;
return(result);
- };
- int opterr=0;
+};
+int opterr=0;
#undef ARGSIZE
-
+
#endif
#ifndef VSTAUDIOOUT
-int main(int argc, char *argv[]){
+int main(int argc, char *argv[])
+{
#ifdef USE_LASH
lash = new LASHClient(&argc, &argv);
@@ -398,182 +407,194 @@ int main(int argc, char *argv[]){
if (argc==1) cerr << "Try 'zynaddsubfx -h' for command-line options.\n" << endl;
#endif
/* Get the settings from the Config*/
- SAMPLE_RATE=config.cfg.SampleRate;
+ SAMPLE_RATE=config.cfg.SampleRate;
SOUND_BUFFER_SIZE=config.cfg.SoundBufferSize;
OSCIL_SIZE=config.cfg.OscilSize;
swaplr=config.cfg.SwapStereo;
-
+
/* Parse command-line options */
#ifdef OS_LINUX
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'},
- {"not-use-jack",2,NULL,'A'},
- {"dummy",2,NULL,'Y'},
- {"help",2,NULL,'h'},
- {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'},
+ {"not-use-jack",2,NULL,'A'},
+ {"dummy",2,NULL,'Y'},
+ {"help",2,NULL,'h'},
+ {0,0,0,0}
};
#endif
opterr=0;
int option_index=0,opt,exitwithhelp=0;
-
- char loadfile[1001];ZERO(loadfile,1001);
- char loadinstrument[1001];ZERO(loadinstrument,1001);
-
- while (1){
+
+ char loadfile[1001];
+ ZERO(loadfile,1001);
+ char loadinstrument[1001];
+ ZERO(loadinstrument,1001);
+
+ while (1) {
/**\todo check this process for a small memory leak*/
#ifdef OS_LINUX
- opt=getopt_long(argc,argv,"l:L:r:b:o:hSDUAY",opts,&option_index);
- char *optarguments=optarg;
+ opt=getopt_long(argc,argv,"l:L:r:b:o:hSDUAY",opts,&option_index);
+ char *optarguments=optarg;
#else
- opt=getopt(argc,argv,"l:L:r:b:o:hSDUAY",&option_index);
- char *optarguments=&winoptarguments[0];
-#endif
-
- if (opt==-1) break;
-
- int tmp;
- switch(opt){
- 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
+ opt=getopt(argc,argv,"l:L:r:b:o:hSDUAY",&option_index);
+ char *optarguments=&winoptarguments[0];
+#endif
+
+ if (opt==-1) break;
+
+ int tmp;
+ switch (opt) {
+ 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 'A':
+ break;
+ case 'U':
+ noui=1;
+ break;
+ case 'A':
#ifdef JACKAUDIOOUT
#ifdef OSSAUDIOOUT
- usejackit=false;
-#endif
-#endif
- break;
- case 'l':tmp=0;
- if (optarguments!=NULL) {
- snprintf(loadfile,1000,"%s",optarguments);
- };
- break;
- case 'L':tmp=0;
- if (optarguments!=NULL) {
- snprintf(loadinstrument,1000,"%s",optarguments);
- };
- break;
- case 'r':tmp=0;
- if (optarguments!=NULL) tmp=atoi(optarguments);
- if (tmp>=4000) {
- SAMPLE_RATE=tmp;
- } else {
- cerr << "ERROR:Incorrect sample rate: " << optarguments << endl;
- exit(1);
- };
- break;
- case 'b':tmp=0;
- if (optarguments!=NULL) tmp=atoi(optarguments);
- if (tmp>=2) {
- SOUND_BUFFER_SIZE=tmp;
- } else {
- cerr << "ERROR:Incorrect buffer size: " << optarguments << endl;
- exit(1);
- };
- break;
- case 'o':tmp=0;
- if (optarguments!=NULL) tmp=atoi(optarguments);
- OSCIL_SIZE=tmp;
- if (OSCIL_SIZE<MAX_AD_HARMONICS*2) OSCIL_SIZE=MAX_AD_HARMONICS*2;
- OSCIL_SIZE=(int) pow(2,ceil(log (OSCIL_SIZE-1.0)/log(2.0)));
- if (tmp!=OSCIL_SIZE){
- cerr << "\nOSCIL_SIZE is wrong (must be 2^n) or too small. Adjusting to ";
- cerr << OSCIL_SIZE << "." << endl;
- }
- break;
- case 'S':swaplr=1;
- break;
- case 'D':dump.startnow();
- break;
- case '?':cerr << "ERROR:Bad option or parameter.\n" << endl;
- exitwithhelp=1;
- break;
- };
+ usejackit=false;
+#endif
+#endif
+ break;
+ case 'l':
+ tmp=0;
+ if (optarguments!=NULL) {
+ snprintf(loadfile,1000,"%s",optarguments);
+ };
+ break;
+ case 'L':
+ tmp=0;
+ if (optarguments!=NULL) {
+ snprintf(loadinstrument,1000,"%s",optarguments);
+ };
+ break;
+ case 'r':
+ tmp=0;
+ if (optarguments!=NULL) tmp=atoi(optarguments);
+ if (tmp>=4000) {
+ SAMPLE_RATE=tmp;
+ } else {
+ cerr << "ERROR:Incorrect sample rate: " << optarguments << endl;
+ exit(1);
+ };
+ break;
+ case 'b':
+ tmp=0;
+ if (optarguments!=NULL) tmp=atoi(optarguments);
+ if (tmp>=2) {
+ SOUND_BUFFER_SIZE=tmp;
+ } else {
+ cerr << "ERROR:Incorrect buffer size: " << optarguments << endl;
+ exit(1);
+ };
+ break;
+ case 'o':
+ tmp=0;
+ if (optarguments!=NULL) tmp=atoi(optarguments);
+ OSCIL_SIZE=tmp;
+ if (OSCIL_SIZE<MAX_AD_HARMONICS*2) OSCIL_SIZE=MAX_AD_HARMONICS*2;
+ OSCIL_SIZE=(int) pow(2,ceil(log (OSCIL_SIZE-1.0)/log(2.0)));
+ if (tmp!=OSCIL_SIZE) {
+ cerr << "\nOSCIL_SIZE is wrong (must be 2^n) or too small. Adjusting to ";
+ cerr << OSCIL_SIZE << "." << endl;
+ }
+ break;
+ case 'S':
+ swaplr=1;
+ break;
+ case 'D':
+ dump.startnow();
+ break;
+ case '?':
+ cerr << "ERROR:Bad option or parameter.\n" << endl;
+ exitwithhelp=1;
+ break;
+ };
};
-
+
if (exitwithhelp!=0) {
- cout << "Usage: zynaddsubfx [OPTION]\n" << endl;
- cout << " -h , --help \t\t\t\t display command-line help and exit" << endl;
- cout << " -l file, --load=FILE\t\t\t loads a .xmz file" << endl;
- cout << " -L file, --load-instrument=FILE\t\t loads a .xiz file" << endl;
- cout << " -r SR, --sample-rate=SR\t\t set the sample rate SR" << endl;
- cout << " -b BS, --buffer-size=SR\t\t set the buffer size (granularity)" << endl;
- cout << " -o OS, --oscil-size=OS\t\t set the ADsynth oscil. size" << endl;
- cout << " -S , --swap\t\t\t\t swap Left <--> Right" << endl;
- cout << " -D , --dump\t\t\t\t Dumps midi note ON/OFF commands" << endl;
- cout << " -U , --no-gui\t\t\t\t Run ZynAddSubFX without user interface" << endl;
+ cout << "Usage: zynaddsubfx [OPTION]\n" << endl;
+ cout << " -h , --help \t\t\t\t display command-line help and exit" << endl;
+ cout << " -l file, --load=FILE\t\t\t loads a .xmz file" << endl;
+ cout << " -L file, --load-instrument=FILE\t\t loads a .xiz file" << endl;
+ cout << " -r SR, --sample-rate=SR\t\t set the sample rate SR" << endl;
+ cout << " -b BS, --buffer-size=SR\t\t set the buffer size (granularity)" << endl;
+ cout << " -o OS, --oscil-size=OS\t\t set the ADsynth oscil. size" << endl;
+ cout << " -S , --swap\t\t\t\t swap Left <--> Right" << endl;
+ cout << " -D , --dump\t\t\t\t Dumps midi note ON/OFF commands" << endl;
+ cout << " -U , --no-gui\t\t\t\t Run ZynAddSubFX without user interface" << endl;
#ifdef JACKAUDIOOUT
#ifdef OSSAUDIOOUT
- cout << " -A , --not-use-jack\t\t\t Use OSS/ALSA instead of JACK" << endl;
+ cout << " -A , --not-use-jack\t\t\t Use OSS/ALSA instead of JACK" << endl;
#endif
#endif
#ifdef OS_WINDOWS
- cout << "\nWARNING: On Windows systems, only short comandline parameters works." << endl;
- cout << " eg. instead '--buffer-size=512' use '-b 512'" << endl;
+ cout << "\nWARNING: On Windows systems, only short comandline parameters works." << endl;
+ cout << " eg. instead '--buffer-size=512' use '-b 512'" << endl;
#endif
- cout << '\n' << endl;
- return(0);
+ cout << '\n' << endl;
+ return(0);
};
-
+
//---------
-
+
initprogram();
#ifdef USE_LASH
#ifdef ALSAMIDIIN
ALSAMidiIn* alsamidi = dynamic_cast<ALSAMidiIn*>(Midi);
if (alsamidi)
- lash->setalsaid(alsamidi->getalsaid());
+ lash->setalsaid(alsamidi->getalsaid());
#endif
#ifdef JACKAUDIOOUT
lash->setjackname(JACKgetname());
#endif
#endif
- if (strlen(loadfile)>1){
+ if (strlen(loadfile)>1) {
int tmp=master->loadXML(loadfile);
- if (tmp<0) {
- fprintf(stderr,"ERROR:Could not load master file %s .\n",loadfile);
- exit(1);
- } else {
- master->applyparameters();
+ if (tmp<0) {
+ fprintf(stderr,"ERROR:Could not load master file %s .\n",loadfile);
+ exit(1);
+ } else {
+ master->applyparameters();
#ifndef DISABLE_GUI
- if (noui==0) ui->refresh_master_ui();
+ if (noui==0) ui->refresh_master_ui();
#endif
- cout << "Master file loaded." << endl;
- };
+ cout << "Master file loaded." << endl;
+ };
};
- if (strlen(loadinstrument)>1){
- int loadtopart=0;
+ if (strlen(loadinstrument)>1) {
+ int loadtopart=0;
int tmp=master->part[loadtopart]->loadXMLinstrument(loadinstrument);
- if (tmp<0) {
- cerr << "ERROR:Could not load instrument file " << loadinstrument << '.' << endl;
- exit(1);
- } else {
- master->part[loadtopart]->applyparameters();
+ if (tmp<0) {
+ cerr << "ERROR:Could not load instrument file " << loadinstrument << '.' << endl;
+ exit(1);
+ } else {
+ master->part[loadtopart]->applyparameters();
#ifndef DISABLE_GUI
- if (noui==0) ui->refresh_master_ui();
+ if (noui==0) ui->refresh_master_ui();
#endif
- cout << "Instrument file loaded." << endl;
- };
+ cout << "Instrument file loaded." << endl;
+ };
};
-
+
#if !(defined(NONEMIDIIN)||defined(WINMIDIIN)||defined(VSTMIDIIN))
pthread_create(&thr1,NULL,thread1,NULL);
@@ -584,32 +605,32 @@ int main(int argc, char *argv[]){
if (!usejackit) pthread_create(&thr2,NULL,thread2,NULL);
#endif
-/*It is not working and I don't know why
-//drop the suid-root permisions
-#if !(defined(JACKAUDIOOUT)||defined(PAAUDIOOUT)||defined(VSTAUDIOOUT)|| (defined (WINMIDIIN)) )
- setuid(getuid());
- seteuid(getuid());
-// setreuid(getuid(),getuid());
-// setregid(getuid(),getuid());
-#endif
-*/
+ /*It is not working and I don't know why
+ //drop the suid-root permisions
+ #if !(defined(JACKAUDIOOUT)||defined(PAAUDIOOUT)||defined(VSTAUDIOOUT)|| (defined (WINMIDIIN)) )
+ setuid(getuid());
+ seteuid(getuid());
+ // setreuid(getuid(),getuid());
+ // setregid(getuid(),getuid());
+ #endif
+ */
if (noui==0) pthread_create(&thr3,NULL,thread3,NULL);
pthread_create(&thr4,NULL,thread4,NULL);
#ifdef WINMIDIIN
- InitWinMidi(master);
+ InitWinMidi(master);
#endif
- while (Pexitprogram==0){
+ while (Pexitprogram==0) {
#ifdef OS_LINUX
- usleep(100000);
+ usleep(100000);
#elif OS_WINDOWS
- Sleep(100);
+ Sleep(100);
#endif
- };
-
+ };
+
#ifdef WINMIDIIN
- StopWinMidi();
+ StopWinMidi();
#endif
exitprogram();
@@ -617,7 +638,7 @@ int main(int argc, char *argv[]){
};
-#else
+#else
#include "Output/VSTaudiooutput.h"
@@ -626,16 +647,17 @@ extern "C" __declspec(dllexport) AEffect *main_plugin(audioMasterCallback audioM
int instances=-1;
-AEffect *main (audioMasterCallback audioMaster){
+AEffect *main (audioMasterCallback audioMaster)
+{
// if (audioMaster(0,audioMasterVersion,0,0,0,0)!=0) {
// return(0);
// };
- if (instances==-1){
- Midi=new NULLMidiIn();
- denormalkillbuf=new REALTYPE [SOUND_BUFFER_SIZE];
- for (int i=0;i<SOUND_BUFFER_SIZE;i++) denormalkillbuf[i]=(RND-0.5)*1e-16;
- instances=0;
+ if (instances==-1) {
+ Midi=new NULLMidiIn();
+ denormalkillbuf=new REALTYPE [SOUND_BUFFER_SIZE];
+ for (int i=0;i<SOUND_BUFFER_SIZE;i++) denormalkillbuf[i]=(RND-0.5)*1e-16;
+ instances=0;
};
if (instances!=0) return(0);//don't allow multiple instances
@@ -646,76 +668,79 @@ AEffect *main (audioMasterCallback audioMaster){
};
void* hInstance;
-BOOL WINAPI DllMain (HINSTANCE hInst,DWORD dwReason,LPVOID lpvReserved){
+BOOL WINAPI DllMain (HINSTANCE hInst,DWORD dwReason,LPVOID lpvReserved)
+{
hInstance=hInst;
return(1);
};
-void *thread(void *arg){
+void *thread(void *arg)
+{
VSTSynth *vs=(VSTSynth *) arg;
-/* FILE *a=fopen("aaaa1","a");
- fprintf(a,"%lx %lx %lx -i=%d\n",vs,0,vs->vmaster,instances);
- fflush(a);fclose(a);
-*/
+ /* FILE *a=fopen("aaaa1","a");
+ fprintf(a,"%lx %lx %lx -i=%d\n",vs,0,vs->vmaster,instances);
+ fflush(a);fclose(a);
+ */
vs->ui=new MasterUI(vs->vmaster,&vs->Pexitprogram);
-/* a=fopen("aaaa1","a");
- fprintf(a,"%lx %lx %lx\n",vs,vs->ui->master,vs->vmaster);
- fflush(a);fclose(a);
-*/
+ /* a=fopen("aaaa1","a");
+ fprintf(a,"%lx %lx %lx\n",vs,vs->ui->master,vs->vmaster);
+ fflush(a);fclose(a);
+ */
vs->ui->showUI();
-/* a=fopen("aaaa1","a");
- fprintf(a,"%lx %lx %lx\n",vs,vs->ui,vs->vmaster);
- fflush(a);fclose(a);
-*/
+ /* a=fopen("aaaa1","a");
+ fprintf(a,"%lx %lx %lx\n",vs,vs->ui,vs->vmaster);
+ fflush(a);fclose(a);
+ */
while (vs->Pexitprogram==0) Fl::wait(0.01);
delete(vs->ui);
Fl::wait(0.01);
-/* a=fopen("aaaa1","a");
- fprintf(a,"EXIT\n");
- fflush(a);fclose(a);
-*/
+ /* a=fopen("aaaa1","a");
+ fprintf(a,"EXIT\n");
+ fflush(a);fclose(a);
+ */
+
-
pthread_exit(0);
return(0);
};
//Parts of the VSTSynth class
-VSTSynth::VSTSynth (audioMasterCallback audioMaster):AudioEffectX(audioMaster,1,0){
+VSTSynth::VSTSynth (audioMasterCallback audioMaster):AudioEffectX(audioMaster,1,0)
+{
instances++;
- if (audioMaster){
- setNumInputs(0);
- setNumOutputs(2);
- setUniqueID('ZASF');
- canProcessReplacing();
+ if (audioMaster) {
+ setNumInputs(0);
+ setNumOutputs(2);
+ setUniqueID('ZASF');
+ canProcessReplacing();
// hasVu(false);
// hasClip(false);
- isSynth(true);
+ isSynth(true);
+
+ programsAreChunks(true);
- programsAreChunks(true);
-
- };
+ };
- SAMPLE_RATE=config.cfg.SampleRate;
+ SAMPLE_RATE=config.cfg.SampleRate;
SOUND_BUFFER_SIZE=config.cfg.SoundBufferSize;
OSCIL_SIZE=config.cfg.OscilSize;
swaplr=config.cfg.SwapStereo;
this->Pexitprogram=0;
-
+
this->vmaster=new Master();
this->vmaster->swaplr=swaplr;
-
+
// FILE *a=fopen("aaaa0","a");
// fprintf(a,"%lx %lx %lx\n",this,this->ui,this->ui->masterwindow);
@@ -729,61 +754,69 @@ VSTSynth::VSTSynth (audioMasterCallback audioMaster):AudioEffectX(audioMaster,1,
-VSTSynth::~VSTSynth(){
+VSTSynth::~VSTSynth()
+{
this->Pexitprogram=1;
-
+
Sleep(200);//wait the thread to finish
-
+
// pthread_mutex_lock(&vmaster->mutex);
- delete(this->vmaster);
+ delete(this->vmaster);
instances--;
};
-long VSTSynth::processEvents(VstEvents *events){
- for (int i=0;i<events->numEvents;i++){
+long VSTSynth::processEvents(VstEvents *events)
+{
+ for (int i=0;i<events->numEvents;i++) {
- //debug stuff
+ //debug stuff
// FILE *a=fopen("events","a");
// fprintf(a,"%lx\n",events->events[i]->type);
// fflush(a);fclose(a);
- if ((events->events[i])->type != kVstMidiType) continue;
- VstMidiEvent *ev= (VstMidiEvent*) events->events[i];
- unsigned char *data= (unsigned char *)ev->midiData;
- int status=data[0]/16;
- int cmdchan=data[0]&0x0f;
- int cntl;
-
- pthread_mutex_lock(&vmaster->mutex);
- switch(status){
- case 0x8:vmaster->NoteOff(cmdchan,data[1]&0x7f);
- break;
- case 0x9:if (data[2]==0) vmaster->NoteOff(cmdchan,data[1]&0x7f);
- else vmaster->NoteOn(cmdchan,data[1]&0x7f,data[2]&0x7f);
- break;
- case 0xB: cntl=Midi->getcontroller(data[1]&0x7f);
- vmaster->SetController(cmdchan,cntl,data[2]&0x7f);
- break;
- case 0xE: vmaster->SetController(cmdchan,C_pitchwheel,data[1]+data[2]*(long int) 128-8192);
- break;
- };
- pthread_mutex_unlock(&vmaster->mutex);
-
+ if ((events->events[i])->type != kVstMidiType) continue;
+ VstMidiEvent *ev= (VstMidiEvent*) events->events[i];
+ unsigned char *data= (unsigned char *)ev->midiData;
+ int status=data[0]/16;
+ int cmdchan=data[0]&0x0f;
+ int cntl;
+
+ pthread_mutex_lock(&vmaster->mutex);
+ switch (status) {
+ case 0x8:
+ vmaster->NoteOff(cmdchan,data[1]&0x7f);
+ break;
+ case 0x9:
+ if (data[2]==0) vmaster->NoteOff(cmdchan,data[1]&0x7f);
+ else vmaster->NoteOn(cmdchan,data[1]&0x7f,data[2]&0x7f);
+ break;
+ case 0xB:
+ cntl=Midi->getcontroller(data[1]&0x7f);
+ vmaster->SetController(cmdchan,cntl,data[2]&0x7f);
+ break;
+ case 0xE:
+ vmaster->SetController(cmdchan,C_pitchwheel,data[1]+data[2]*(long int) 128-8192);
+ break;
+ };
+ pthread_mutex_unlock(&vmaster->mutex);
+
};
-return(1);
+ return(1);
};
-long VSTSynth::getChunk(void** data,bool isPreset){
+long VSTSynth::getChunk(void** data,bool isPreset)
+{
int size=0;
size=vmaster->getalldata((char **)data);
return((long)size);
};
-long VSTSynth::setChunk(void *data,long size,bool isPreset){
+long VSTSynth::setChunk(void *data,long size,bool isPreset)
+{
vmaster->putalldata((char*)data,size);
return(0);
};
