Commit 7669964b authored by Daniel Clarke's avatar Daniel Clarke

ignoring cpp examples now

parent 343505a4
......@@ -3,4 +3,5 @@ src/cpp/Maximilian-master/
Recordings/
maxiLib/maximilian_examples_web/load_sample_test.html
maxiLib/sed_addText.sh
maxiLib/sed_text_stuff/
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maxiLib/sed_text_stuff/
maximilian_examples_cpp/
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#include "maximilian.h"
//This shows how the fundamental building block of digital audio - the sine wave.
//
maxiOsc mySine;//One oscillator - can be called anything. Can be any of the available waveforms.
void setup() {//some inits
//nothing to go here this time
}
void play(double *output) {
output[0]=mySine.sinewave(440);
output[1]=output[0];
}
// Here is an example of a Maximilian filter being used.
// There are a number of filters in Maximilian, including low and high pass filters.
// There are also resonant filters and a state variable filter.
#include "maximilian.h"
maxiOsc myCounter,mySwitchableOsc;//
int CurrentCount;//
double myOscOutput,myCurrentVolume, myFilteredOutput;//
maxiEnv myEnvelope;
maxiFilter myFilter;
void setup() {//some inits
//Timing is in ms
myEnvelope.setAttack(0);
myEnvelope.setDecay(1); // Needs to be at least 1
myEnvelope.setSustain(1);
myEnvelope.setRelease(1000);
}
void play(double *output) {
myCurrentVolume=myEnvelope.adsr(1.,myEnvelope.trigger);
CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value.
// You'll notice that these 'if' statements don't require curly braces "{}".
// This is because there is only one outcome if the statement is true.
if (CurrentCount==1) myEnvelope.trigger=1; //trigger the envelope
else myEnvelope.trigger=0;//release the envelope to make it fade out only if it's been triggered
myOscOutput=mySwitchableOsc.sawn(100);
// Below, the oscilator signals are being passed through a low pass filter.
// The middle input is the filter cutoff. It is being controlled by the envelope.
// Notice that the envelope is being amplified so that it scales between 0 and 1000.
// The last input is the resonance.
myFilteredOutput=myFilter.lores(myOscOutput,myCurrentVolume*1000,10);
output[0]=myFilteredOutput;//left speaker
output[1]=output[0];
}
#include "maximilian.h"
maxiOsc myOsc,myAutoPanner;//
double myStereoOutput[2];
maxiMix myOutputs;//this is the stereo mixer channel.
void setup() {//some inits
}
void play(double *output) {
myOutputs.stereo(myOsc.noise(),myStereoOutput,(myAutoPanner.sinewave(1)+1)/2);//Stereo, Quad or 8 Channel. Specify the input to be mixed, the output[numberofchannels], and the pan (0-1,equal power).
output[0]=myStereoOutput[0];//When working with mixing, you need to specify the outputs explicitly
output[1]=myStereoOutput[1];//
}
#include "maximilian.h"
maxiSample beats; //We give our sample a name. It's called beats this time. We could have loads of them, but they have to have different names.
void setup() {//some inits
beats.load("/Users/username/somewhere/Maximilian/beat2.wav");//load in your samples. Provide the full path to a wav file.
printf("Summary:\n%s", beats.getSummary());//get info on samples if you like.
}
void play(double *output) {//this is where the magic happens. Very slow magic.
*output=beats.play();//just play the file. Looping is default for all play functions.
// *output=beats.play(0.69);//play the file with a speed setting. 1. is normal speed.
// *output=beats.play(0.5,0,44100);//linear interpolationplay with a frequency input, start point and end point. Useful for syncing.
// *output=beats.play4(0.5,0,44100);//cubic interpolation play with a frequency input, start point and end point. Useful for syncing.
}
//Using BPF equation from http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
//Example contributed by Rebecca Fiebrink
#include "maximilian.h"
float xs[3], ys[3];
float a0, a1, a2, b0, b1, b2;
float f0 = 400; //THE FREQUENCY
float Q = 1.0;
maxiOsc mySwitchableOsc;
void setup() {//some inits
double w0 = 2*PI*f0/44100;
double alpha = sin(w0)/(2*Q);
//Band-pass reson:
// b0 = alpha;
// b1 = 0;
// b2 = -1 * alpha;
// a0 = 1 + alpha;
// a1 = -2*cos(w0);
// a2 = 1 - alpha;
//Notch:
b0 = 1;
b1 = -2*cos(w0);
b2 = 1;
a0 = 1 + alpha;
a1 = -2*cos(w0);
a2 = 1 - alpha;
//LPF:
// b0 = (1 - cos(w0))/2;
// b1 = 1 - cos(w0);
// b2 = (1 - cos(w0))/2;
// a0 = 1 + alpha;
// a1 = -2*cos(w0);
// a2 = 1 - alpha;
}
void play(double *output) {
xs[0] = mySwitchableOsc.sawn(400);
ys[0] = (b0/a0)*xs[0] + (b1/a0)*xs[1] + (b2/a0)*xs[2]
- (a1/a0)*ys[1] - (a2/a0)*ys[2];
*output = ys[0];
ys[2] = ys[1]; ys[1] = ys[0];
xs[2] = xs[1]; xs[1] = xs[0];
}
#include "maximilian.h"
//This shows how to use maximilian to build a monophonic synth
//These are the synthesiser bits
maxiOsc VCO1,VCO2,LFO1,LFO2;
maxiFilter VCF;
maxiEnv ADSR;
//This is a bunch of control signals so that we can hear something
maxiOsc timer;//this is the metronome
int currentCount,lastCount;//these values are used to check if we have a new beat this sample
//and these are some variables we can use to pass stuff around
double VCO1out,VCO2out,LFO1out,LFO2out,VCFout,ADSRout;
void setup() {//some inits
ADSR.setAttack(1000);
ADSR.setDecay(1);
ADSR.setSustain(1);
ADSR.setRelease(1000);
}
void play(double *output) {
//so this first bit is just a basic metronome so we can hear what we're doing.
currentCount=(int)timer.phasor(0.5);//this sets up a metronome that ticks every 2 seconds
if (lastCount!=currentCount) {//if we have a new timer int this sample, play the sound
ADSR.trigger=1;
cout << "tick\n";//the clock ticks
lastCount=0;//set lastCount to 0
}
//and this is where we build the synth
ADSRout=ADSR.adsr(1.0,ADSR.trigger);
LFO1out=LFO1.sinebuf(0.2);//this lfo is a sinewave at 0.2 hz
VCO1out=VCO1.pulse(55,0.6);//here's VCO1. it's a pulse wave at 55 hz, with a pulse width of 0.6
VCO2out=VCO2.pulse(110+LFO1out,0.2);//here's VCO2. it's a pulse wave at 110hz with LFO modulation on the frequency, and width of 0.2
VCFout=VCF.lores((VCO1out+VCO2out)*0.5, ADSRout*10000, 10);//now we stick the VCO's into the VCF, using the ADSR as the filter cutoff
double finalSound=VCFout*ADSRout;//finally we add the ADSR as an amplitude modulator
ADSR.trigger=0;
output[0]=finalSound;
output[1]=finalSound;
}
#include "maximilian.h"
//This shows how to use maximilian to build a polyphonic synth.
//These are the synthesiser bits
maxiOsc VCO1[6],VCO2[6],LFO1[6],LFO2[6];
maxiFilter VCF[6];
maxiEnv ADSR[6];
//This is a bunch of control signals so that we can hear something
maxiOsc timer;//this is the metronome
int currentCount,lastCount,voice=0;//these values are used to check if we have a new beat this sample
//and these are some variables we can use to pass stuff around
double VCO1out[6],VCO2out[6],LFO1out[6],LFO2out[6],VCFout[6],ADSRout[6],mix,pitch[6];
void setup() {//some inits
for (int i=0;i<6;i++) {
ADSR[i].setAttack(0);
ADSR[i].setDecay(200);
ADSR[i].setSustain(0.2);
ADSR[i].setRelease(2000);
}
}
void play(double *output) {
mix=0;//we're adding up the samples each update and it makes sense to clear them each time first.
//so this first bit is just a basic metronome so we can hear what we're doing.
currentCount=(int)timer.phasor(8);//this sets up a metronome that ticks 8 times a second
if (lastCount!=currentCount) {//if we have a new timer int this sample, play the sound
if (voice==6) {
voice=0;
}
ADSR[voice].trigger=1;//trigger the envelope from the start
pitch[voice]=voice+1;
voice++;
}
//and this is where we build the synth
for (int i=0; i<6; i++) {
ADSRout[i]=ADSR[i].adsr(1.,ADSR[i].trigger);//our ADSR env is passed a constant signal of 1 to generate the transient.
LFO1out[i]=LFO1[i].sinebuf(0.2);//this lfo is a sinewave at 0.2 hz
VCO1out[i]=VCO1[i].pulse(55*pitch[i],0.6);//here's VCO1. it's a pulse wave at 55 hz, with a pulse width of 0.6
VCO2out[i]=VCO2[i].pulse((110*pitch[i])+LFO1out[i],0.2);//here's VCO2. it's a pulse wave at 110hz with LFO modulation on the frequency, and width of 0.2
VCFout[i]=VCF[i].lores((VCO1out[i]+VCO2out[i])*0.5, 250+((pitch[i]+LFO1out[i])*1000), 10);//now we stick the VCO's into the VCF, using the ADSR as the filter cutoff
mix+=VCFout[i]*ADSRout[i]/6;//finally we add the ADSR as an amplitude modulator
}
output[0]=mix*0.5;//left channel
output[1]=mix*0.5;//right channel
// This just sends note-off messages.
for (int i=0; i<6; i++) {
ADSR[i].trigger=0;
}
}
#include "maximilian.h"
//Bizarelly, this sounds a little bit like Kraftwerk's 'Metropolis', although it isn't. Funny that.
maxiOsc sound,bass,timer,mod,lead,lead2,leadmod;//here are the synth bits
maxiEnv envelope, leadenvelope;//some envelopes
maxiFilter filter, filter2;//some filters
maxiDelayline delay;//a delay
convert mtof;//a method for converting midi notes to frequency
double bassout,leadout, delayout;//some variables to hold the data and pass it around
int trigger, trigger2, newnote;//some control variables
int currentCount,lastCount,playHead=0, currentChord=0;//some other control variables
int pitch[8]={57,57,59,60};//the bassline for the arpeggio
int chord[8]={0,0,7,2,5,5,0,0};//the root chords for the arpeggio
float currentPitch,leadPitch;//the final pitch variables
//here's the lead line trigger array, followed by the pitches
int leadLineTrigger[256]={1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,0,0,0,1,0,0,0,1,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0};
int leadLinePitch[15]={69,67,65,64,67,66,64,62,65,64,62,57,55,60,57};
void setup() {//some inits
}
void play(double *output) {//this is where the magic happens. Very slow magic.
currentCount=(int)timer.phasor(9);//this sets up a metronome that ticks every so often
if (lastCount!=currentCount) {//if we have a new timer int this sample, play the sound
trigger=1;//play the arpeggiator line
trigger2=leadLineTrigger[playHead%256];//play the lead line
if (trigger2==1) {//if we are going to play a note
leadPitch=mtof.mtof(leadLinePitch[newnote]);//get the next pitch val
newnote++;//and iterate
if (newnote>14) {
newnote=0;//make sure we don't go over the edge of the array
}
}
currentPitch=mtof.mtof(pitch[(playHead%4)]+chord[currentChord%8]);//write the frequency val into currentPitch
playHead++;//iterate the playhead
if (playHead%32==0) {//wrap every 4 bars
currentChord++;//change the chord
}
//cout << "tick\n";//the clock ticks
lastCount=0;//set lastCount to 0
}
bassout=filter2.lores(envelope.adsr(bass.saw(currentPitch*0.5)+sound.pulse(currentPitch*0.5,mod.phasor(1)),1,0.9995, 0.25, 0.9995, 1, trigger),9250,2);//new, simple ADSR.
leadout=filter.lores(leadenvelope.ar(lead2.saw(leadPitch*4)+lead.pulse(leadPitch+(leadmod.sinebuf(1.9)*1.5), 0.6), 0.00005, 0.999975, 50000, trigger2),5900,10);//leadline
delayout=(leadout+(delay.dl(leadout, 14000, 0.8)*0.5))/2;//add some delay
if(trigger!=0)trigger=0;//set the trigger to off if you want it to trigger immediately next time.
output[0]=(bassout)/2;//sum output
output[1]=(bassout)/2;
}
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#include "maximilian.h"
maxiSample beats; //We give our sample a name. It's called beats this time. We could have loads of them, but they have to have different names.
maxiDyn compressor; //this is a compressor
double out;
void setup() {//some inits
beats.load("/Users/michaelgrierson/Documents/workspace/Maximilian/ofxMaxim/examples/OSX/ofMaximExample007OSX_Granular/bin/data/beat2.wav");//load in your samples. Provide the full path to a wav file.
printf("Summary:\n%s", beats.getSummary());//get info on samples if you like.
compressor.setAttack(100);
compressor.setRelease(300);
compressor.setThreshold(0.25);
compressor.setRatio(5);
//you can set these any time you like.
}
void play(double *output) {//this is where the magic happens. Very slow magic.
//here, we're just compressing the file in real-time
//arguments are input,ratio,threshold,attack,release
out=compressor.compress(beats.play());
output[0]=out;
output[1]=out;
}
#include "maximilian.h"
maxiSample kick,snare; //we've got two sampleplayers
maxiOsc timer; //and a timer
int currentCount,lastCount,playHead,hit[16]={1,0,0,1,0,0,1,0,0,1,0,0,1,0,0,1}; //This is the sequence for the kick
int snarehit[16]={0,0,0,0,1,0,0,0,0,0,0,0,1,1,0,0};//This is the sequence for the snare
int kicktrigger,snaretrigger;
double sampleOut;
void setup() {//some inits
//YOU HAVE TO PROVIDE THE SAMPLES....
kick.load("/Users/yourusername/somewhere/kick.wav");//load in your samples. Provide the full path to a wav file.
snare.load("/Users/yourusername/somewhere/snare.wav");//load in your samples. Provide the full path to a wav file.
printf("Summary:\n%s", kick.getSummary());//get info on samples if you like.
//beats.getLength();
}
void play(double *output) {//this is where the magic happens. Very slow magic.
currentCount=(int)timer.phasor(8);//this sets up a metronome that ticks 8 times a second
if (lastCount!=currentCount) {//if we have a new timer int this sample, play the sound
kicktrigger=hit[playHead%16];//get the value out of the array for the kick
snaretrigger=snarehit[playHead%16];//same for the snare
playHead++;//iterate the playhead
lastCount=0;//reset the metrotest
}
if (kicktrigger==1) {//if the sequence has a 1 in it
kick.trigger();//reset the playback position of the sample to 0 (the beginning)
}
if (snaretrigger==1) {
snare.trigger();//likewise for the snare
}
sampleOut=kick.playOnce()+snare.playOnce();//just play the file. No looping.
output[0]=sampleOut;//left channel
output[1]=sampleOut;//right channel
kicktrigger = 0;//set trigger to 0 at the end of each sample to guarantee retriggering.
snaretrigger = 0;
}
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#include "maximilian.h"
//this tutorial explains how to use the maxiEnv
maxiSample sound1;
maxiOsc timer,snarePhase; //and a timer
maxiEnv envelope;//this is going to be an envelope
int currentCount,lastCount,playHead,
sequence[16]={1,0,0,0,1,0,0,0,1,0,0,0,1,0,0,0}; //This is the sequence for the kick
int sampleTrigger;
double sampleOut;
void setup() {//some inits
//YOU HAVE TO PROVIDE THE SAMPLES....
sound1.load("/Users/mickgrierson/Documents/audio/68373__juskiddink__Cello_open_string_bowed.wav");//load in your samples. Provide the full path to a wav file.
printf("Summary:\n%s", sound1.getSummary());//get info on samples if you like.
//beats.getLength();
}
void play(double *output) {//this is where the magic happens. Very slow magic.
currentCount=(int)timer.phasor(8);//this sets up a metronome that ticks 8 times a second
if (lastCount!=currentCount) {//if we have a new timer int this sample, play the sound
sampleTrigger=sequence[playHead%16];
playHead++;//iterate the playhead
lastCount=0;//reset the metrotest
}
//the envelope we're using here is an AR envelope.
//It has an input (which in this case is a sound)
//It has an attack coefficient, a hold val (in samples)
//and a release coefficient. Finally, it has a trigger input.
//If you stick a 1 in the trigger input, it retriggers the envelope
sampleOut=envelope.ar(sound1.play(1.), 0.1, 0.9999, 1, sampleTrigger); //
output[0]=sampleOut;//left channel
output[1]=sampleOut;//right channel
sampleTrigger = 0;//set trigger to 0 at the end of each sample to guarantee retriggering.
}
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//This examples shows another fundamental building block of digital audio - adding two sine waves together. When you add waves together they create a new wave whose amplitude at any time is computed by adding the current amplitudes of each wave together. So, if one wave has an amplitude of 1, and the other has an amplitude of 1, the new wave will be equal to 2 at that point in time. Whereas, later, if one wave has an amplitude of -1, and the other has an amplitude of 1, the new wave - the one you hear - will equal 0. This can create some interesting effects, including 'beating', when the waves interact to create a single wave that fades up and down based on the frequencies of the two interacting waves. The frequency of the 'beating' i.e. the fading in and out, is equal to the difference in frequency between the two waves.
#include "maximilian.h"
maxiOsc mySine,myOtherSine;//Two oscillators with names.
void setup() {//some inits
//nothing to go here this time
}
void play(double *output) {//this is where the magic happens. Very slow magic.
//output[0] is the left output. output[1] is the right output
output[0]=mySine.sinewave(440)+myOtherSine.sinewave(441);//these two sines will beat together. They're now a bit too loud though..
output[1]=output[0];
}
#include "maximilian.h"
//This shows how to use maximilian to do basic amplitude modulation. Amplitude modulation is when you multiply waves together. In maximilian you just use the * inbetween the two waveforms.
maxiOsc mySine,myOtherSine;//Two oscillators. They can be called anything. They can be any of the available waveforms. These ones will be sinewaves
void setup() {//some inits
//nothing to go here this time
}
void play(double *output) {
// This form of amplitude modulation is straightforward multiplication of two waveforms.
// Notice that the maths is different to when you add waves.
// The waves aren't 'beating'. Instead, the amplitude of one is modulating the amplitude of the other
// Remember that the sine wave has positive and negative sections as it oscillates.
// When you multiply something by -1, its phase is inverted but it retains its amplitude.
// So you hear 2 waves per second, not 1, even though the frequency is 1.
output[0]=mySine.sinewave(440)*myOtherSine.sinewave(1);
output[1]=output[0];
}
#include "maximilian.h"
//This shows how to use maximilian to do basic amplitude modulation.
//It also shows what happens when you modulate waves with waves that have frequencies over 20 hz.
//You start to get interesting effects.
maxiOsc mySine,myOtherSine,myPhasor;//Three oscillators. They can be called anything. They can be any of the available waveforms.
void setup() {//some inits
//nothing to go here this time
}
void play(double *output) {
//Using the phasor we can create a ramp, and use this ramp to set the frequency of one of the waves.
//When the frequency of the lower waveform passes over the threshold of 20hz, we start to hear two new waveforms.
//The frequency of the first new wave is the sum of the two original waves.
//The frequency of the second new wave is the difference of the two original waves.
//So you hear two new waves, one going up, one going down.
output[0]=mySine.sinewave(440)*myOtherSine.sinewave(myPhasor.phasor(0.01,0,440));
output[1]=output[0];
}
// One way of thinking about FM synthesis is to see it as vibrato.
// You make a pitch, then vary it up and down at some rate.
// You can change the speed of the pitch variation (modulation frequency), and also the amount of variation (modulation index).
// In FM, usually only one of the waveforms - the carrier that provides the initial pitch - is sent to the output.
// The frequency of the the carrier wave is continually adjusted at a rate equal to the frequency of the second wave (the modulator).
// So at any given point in time, the frequency of the carrier can increase by an amount equal to the current amp of the modulator.
// This has some interesting effects.
#include "maximilian.h"
maxiOsc mySine,myOtherSine;//Two oscillators
void setup() {//some inits
//nothing to go here this time
}
void play(double *output) {
// In this example, the 'myOtherSine.sinewave' is at an amplitude of 1, it's original amplitude.
// This is pretty simple and not too useful.
//output[0]=mySine.sinewave(440*myOtherSine.sinewave(1));
// Perhaps you should comment out the above line and uncomment the below one instead
// It shows how the frequency of the carrier is altered by ADDING a second waveform to its frequency value.
// The carrier frequency is 440, and the modulation frequency is 1.
// It also shows how the modulation index works. In this case the modulation index is 100
// Try adjusting the modolation index. Also, try altering the modulation frequency.
output[0]=mySine.sinewave(440+(myOtherSine.sinewave(1)*100));
output[1]=output[0];
}
// In complex FM systems you can have lots of modulators stacked together in interesting ways, and theoretically this can make any sound.
// John Chowning is the guy you probably want to talk to about that.
// Nothing much to say about this other than I like it.
#include "maximilian.h"
maxiOsc mySine,myOtherSine,myLastSine,myPhasor;//Three oscillators
void setup() {//some inits
//nothing to go here this time
}
void play(double *output) {
output[0]=mySine.sinewave(myOtherSine.sinewave(myLastSine.sinewave(0.1)*30)*440);//awesome bassline
output[1]=output[0];
}
#include "maximilian.h"
maxiOsc mySine; // This is the oscillator we will use to generate the test tone
maxiClock myClock; // This will allow us to generate a clock signal and do things at specific times
double freq; // This is a variable that we will use to hold and set the current frequency of the oscillator
void setup() {
myClock.setTicksPerBeat(1);//This sets the number of ticks per beat
myClock.setTempo(120);// This sets the tempo in Beats Per Minute
freq=20; // Here we initialise the variable
}
void play(double *output) {
myClock.ticker(); // This makes the clock object count at the current samplerate
//This is a 'conditional'. It does a test and then does something if the test is true
if (myClock.tick) { // If there is an actual tick at this time, this will be true.
freq+=100; // DO SOMETHING
} // The curly braces close the conditional
//output[0] is the left output. output[1] is the right output
output[0]=mySine.sinewave(freq);//simple as that!
output[1]=output[0];
}
// This example shows how you can create a basic counter with a phasor.
// A phasor oscillator can create a ramp between any two values.
// It takes three inputs - frequency, start value and stop value.
// These are all double precision floats, so it's a continuous slide.
// If you write it into an integer, it will round it off for you.
// This creates a bunch of steps.
#include "maximilian.h"
maxiOsc myCounter,mySquare;//these oscillators will help us count and play sound
int CurrentCount;//we're going to put the current count in this variable so that we can use it more easily.
void setup() {//some inits
//nothing to go here this time
}
void play(double *output) {
// Here you can see that CurrentCount is an int. It's taking the continuous output of the phasor and convering it.
// You don't need to explicityly 'cast' (i.e. change) the value from a float to an int.
// It happens automagically in these cases.
// Once every second, CurrentCount counts from 1 until it gets to 9, then resets itself.
// When it reaches 9 it resets, so the values you get are 1-8.
CurrentCount=myCounter.phasor(1, 1, 9);