Support for loading a specific channel from a stereo wav file as an argument to the load function. Default is channel 0

1 files changed, 9 insertions, 62 deletions

diff --git a/main.cpp b/main.cpp
index 5b1a457..f1d13b2 100755
--- a/main.cpp
+++ b/main.cpp
@@ -1,74 +1,21 @@
 #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];
-maxiEnvelope ADSR[6];
-
-//These are the control values for the envelope
-
-double adsrEnv[8]={1,5,0.125,100,0.125,200,0,1000};
-
-//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];
-
+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/mickgrierson/Documents/workspace/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) {
-	
-	mix=0;//we're adding up the samples each update and it makes sense to clear them each time first.
+void play(double *output) {//this is where the magic happens. Very slow magic.
 	
-	//so this first bit is just a basic metronome so we can hear what we're doing.
+	*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.
 	
-	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(0, adsrEnv[0]);//trigger the envelope from the start
-		pitch[voice]=voice+1;
-		voice++;
-
-		lastCount=0;
-
-	}
-
-	//and this is where we build the synth
-	
-	for (int i=0; i<6; i++) {
-		
-	
-	ADSRout[i]=ADSR[i].line(8,adsrEnv);//our ADSR env has 8 value/time pairs.
-	
-	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;
-	output[1]=mix*0.5;
-
 }
+