robert@372: /*
robert@372:  ____  _____ _        _    
robert@372: | __ )| ____| |      / \   
robert@372: |  _ \|  _| | |     / _ \  
robert@372: | |_) | |___| |___ / ___ \ 
robert@372: |____/|_____|_____/_/   \_\.io
robert@372: 
robert@372:  */
robert@372: 
andrewm@57: /*
andrewm@57:  * render.cpp
andrewm@57:  *
andrewm@57:  *  Created on: Oct 24, 2014
andrewm@57:  *      Author: parallels
andrewm@57:  */
andrewm@57: 
robert@372: /**
robert@372: \example 3_analog_input
robert@372: 
robert@372: Connecting potentiometers
robert@372: -------------------------
robert@372: 
robert@372: This sketch produces a sine tone, the frequency and amplitude of which are 
robert@372: affected by data received on the analog pins. Before looping through each audio 
robert@372: frame, we declare a value for the frequency and amplitude of our sine tone 
robert@372: (line 55); we adjust these values by taking in data from analog sensors 
robert@372: (for example potentiometers) with `analogRead()`.
robert@372: 
robert@372: - connect a 10K pot to 3.3V and GND on its 1st and 3rd pins.
robert@372: - connect the 2nd middle pin of the pot to analogIn 0.
robert@372: - connect another 10K pot in the same way but with the middle pin connected to analogIn 1.
robert@372: 
robert@372: The important thing to notice is that audio is sampled twice as often as analog 
robert@372: data. The audio sampling rate is 44.1kHz (44100 frames per second) and the 
robert@372: analog sampling rate is 22.05kHz (22050 frames per second). On line 62 you might 
robert@372: notice that we are processing the analog data and updating frequency and 
robert@372: amplitude only on every second audio sample, since the analog sampling rate is 
robert@372: half that of the audio.
robert@372: 
robert@372: Note that the pin numbers are stored in the variables `gAnalogInputFrequency` and 
robert@372: `gAnalogInputAmplitude`. These are declared in the main.cpp file; if you look in 
robert@372: that file you will see that they have the values of 0 and 1. Bear in mind that 
robert@372: these are analog input pins which is a specific header!
robert@372: */
andrewm@57: 
giuliomoro@301: #include <Bela.h>
andrewm@57: #include <rtdk.h>
andrewm@57: #include <cmath>
andrewm@57: 
andrewm@57: float gPhase;
andrewm@57: float gInverseSampleRate;
andrewm@57: int gAudioFramesPerAnalogFrame;
andrewm@57: 
andrewm@57: // These settings are carried over from main.cpp
andrewm@57: // Setting global variables is an alternative approach
andrewm@57: // to passing a structure to userData in setup()
andrewm@57: 
andrewm@57: extern int gSensorInputFrequency;
andrewm@57: extern int gSensorInputAmplitude;
andrewm@57: 
andrewm@57: // setup() is called once before the audio rendering starts.
andrewm@57: // Use it to perform any initialisation and allocation which is dependent
andrewm@57: // on the period size or sample rate.
andrewm@57: //
andrewm@57: // userData holds an opaque pointer to a data structure that was passed
andrewm@57: // in from the call to initAudio().
andrewm@57: //
andrewm@57: // Return true on success; returning false halts the program.
andrewm@57: 
giuliomoro@301: bool setup(BelaContext *context, void *userData)
andrewm@57: {
andrewm@57: 	if(context->analogFrames == 0 || context->analogFrames > context->audioFrames) {
andrewm@57: 		rt_printf("Error: this example needs analog enabled, with 4 or 8 channels\n");
andrewm@57: 		return false;
andrewm@57: 	}
andrewm@57: 
andrewm@57: 	gAudioFramesPerAnalogFrame = context->audioFrames / context->analogFrames;
andrewm@57: 	gInverseSampleRate = 1.0 / context->audioSampleRate;
andrewm@57: 	gPhase = 0.0;
andrewm@57: 
andrewm@57: 	return true;
andrewm@57: }
andrewm@57: 
andrewm@57: // render() is called regularly at the highest priority by the audio engine.
andrewm@57: // Input and output are given from the audio hardware and the other
andrewm@57: // ADCs and DACs (if available). If only audio is available, numMatrixFrames
andrewm@57: // will be 0.
andrewm@57: 
giuliomoro@301: void render(BelaContext *context, void *userData)
andrewm@57: {
andrewm@57: 	float frequency = 440.0;
andrewm@57: 	float amplitude = 0.8;
andrewm@57: 
andrewm@57: 	// There are twice as many audio frames as matrix frames since audio sample rate
andrewm@57: 	// is twice as high
andrewm@57: 
andrewm@57: 	for(unsigned int n = 0; n < context->audioFrames; n++) {
andrewm@57: 		if(!(n % gAudioFramesPerAnalogFrame)) {
andrewm@57: 			// Even audio samples: update frequency and amplitude from the matrix
andrewm@308: 			frequency = map(analogRead(context, n/gAudioFramesPerAnalogFrame, gSensorInputFrequency), 0, 1, 100, 1000);
andrewm@308: 			amplitude = analogRead(context, n/gAudioFramesPerAnalogFrame, gSensorInputAmplitude);
andrewm@57: 		}
andrewm@57: 
andrewm@57: 		float out = amplitude * sinf(gPhase);
andrewm@57: 
andrewm@57: 		for(unsigned int channel = 0; channel < context->audioChannels; channel++)
andrewm@57: 			context->audioOut[n * context->audioChannels + channel] = out;
andrewm@57: 
andrewm@57: 		gPhase += 2.0 * M_PI * frequency * gInverseSampleRate;
andrewm@57: 		if(gPhase > 2.0 * M_PI)
andrewm@57: 			gPhase -= 2.0 * M_PI;
andrewm@57: 	}
andrewm@57: }
andrewm@57: 
andrewm@57: // cleanup() is called once at the end, after the audio has stopped.
andrewm@57: // Release any resources that were allocated in setup().
andrewm@57: 
giuliomoro@301: void cleanup(BelaContext *context, void *userData)
andrewm@57: {
andrewm@57: 
andrewm@57: }