Merge branch 'master' of ssh://git.xpub.nl:2501/XPUB/special-issue-x

.gitignore

@@ -0,0 +1 @@
+.DS_Store

sketches/delicious hands/README.md

@@ -0,0 +1,15 @@
+STRINGS1
+======
+
+* Knob 1: Loop start
+* Knob 2: Loop length
+* Knob 3: Pitch
+* Input 3: Gate trigger
+
+All code ripped from Framen who ripped and modified these fine people's work:
+
+* http://www.ginkosynthese.com/product/grains/
+* http://playground.arduino.cc/Code/PCMAudio
+* michael@hurts.ca
+* https://www.facebook.com/HRTLmuzik
+* https://github.com/thatpixguy/hrtl-amen

sketches/delicious hands/delicious hands.ino

@@ -0,0 +1,137 @@
+#include <stdint.h>
+#include <avr/interrupt.h>
+#include <avr/io.h>
+#include <avr/pgmspace.h>
+
+#include "sample.h"
+
+#define LED_PIN     13
+#define SPEAKER_PIN 11 
+
+#define KNOB_1  (0)
+#define KNOB_2  (1)  
+#define KNOB_3  (2)   
+#define INPUT_3 (3)   
+
+volatile uint16_t sample;
+volatile uint16_t loop_start;
+volatile uint16_t loop_length;
+volatile uint16_t index_bounds;
+volatile uint16_t loop_overflow;
+
+volatile boolean gate;
+volatile boolean gate_prev;
+
+byte lastSample;
+
+void startPlayback()
+{
+    pinMode(SPEAKER_PIN, OUTPUT);
+
+    // Set up Timer 2 to do pulse width modulation on the speaker pin.
+    // Use internal clock (datasheet p.160)
+    ASSR &= ~(_BV(EXCLK) | _BV(AS2));
+
+    // Set fast PWM mode  (p.157)
+    TCCR2A |= _BV(WGM21) | _BV(WGM20);
+    TCCR2B &= ~_BV(WGM22);
+
+    // Do non-inverting PWM on pin OC2A (p.155)
+    // On the Arduino this is pin 11.
+    TCCR2A = (TCCR2A | _BV(COM2A1)) & ~_BV(COM2A0);
+    TCCR2A &= ~(_BV(COM2B1) | _BV(COM2B0));
+    // No prescaler (p.158)
+    TCCR2B = (TCCR2B & ~(_BV(CS12) | _BV(CS11))) | _BV(CS10);
+
+    // Set initial pulse width to the first sample.
+    OCR2A = pgm_read_byte(&sound_data[0]);
+
+    // Set up Timer 1 to send a sample every interrupt.
+    cli();
+
+    // Set CTC mode (Clear Timer on Compare Match) (p.133)
+    // Have to set OCR1A *after*, otherwise it gets reset to 0!
+    TCCR1B = (TCCR1B & ~_BV(WGM13)) | _BV(WGM12);
+    TCCR1A = TCCR1A & ~(_BV(WGM11) | _BV(WGM10));
+
+    // No prescaler (p.134)
+    TCCR1B = (TCCR1B & ~(_BV(CS12) | _BV(CS11))) | _BV(CS10);
+
+    // Set the compare register (OCR1A).
+    // OCR1A is a 16-bit register, so we have to do this with
+    // interrupts disabled to be safe.
+    OCR1A = F_CPU / SAMPLE_RATE; // 16e6 / 8000 = 2000
+
+    // Enable interrupt when TCNT1 == OCR1A (p.136)
+    TIMSK1 |= _BV(OCIE1A);
+
+    lastSample = pgm_read_byte(&sound_data[sound_length - 1]);
+    sample = 0;
+    sei();
+}
+
+void stopPlayback()
+{
+    TIMSK1 &= ~_BV(OCIE1A); // Disable playback per-sample interrupt.
+    TCCR1B &= ~_BV(CS10);   // Disable the per-sample timer completely.
+    TCCR2B &= ~_BV(CS10);   // Disable the PWM timer.
+    digitalWrite(SPEAKER_PIN, LOW);
+}
+
+void setup()
+{
+    pinMode(LED_PIN, OUTPUT);
+    digitalWrite(LED_PIN, HIGH);
+
+    startPlayback();
+
+    loop_start  = 0; 
+    loop_length = sound_length;
+    gate = false;
+    gate_prev = false;
+}
+
+// This is called at 8000 Hz to load the next sample.
+ISR(TIMER1_COMPA_vect)
+{
+    if(sample >= index_bounds)
+    {
+        sample = loop_start;
+    }
+    else if((sample < loop_start) &&
+            (sample >= loop_overflow))
+    {
+        sample = loop_start;
+    }
+    else if((gate == true) &&
+            (gate_prev == false))
+    {
+        sample = loop_start;
+    }
+    else
+    {
+        OCR2A = pgm_read_byte(&sound_data[sample % sound_length]);
+    }
+    gate_prev = gate;
+    sample++;
+}
+
+void loop()
+{
+    loop_start = analogRead(KNOB_1) / 1024.0 * sound_length;
+    loop_length = (analogRead(KNOB_2) + 1) / 1024.0 * sound_length;
+    OCR1A = (512.0 / (analogRead(KNOB_3) + 1)) * (F_CPU / SAMPLE_RATE);
+    gate = analogRead(3) >> 9;  // 10 bits in. gate < 512 == off, gate >= 512 == on
+
+//  can be up to 2x sound length. the more you know.
+    index_bounds = loop_start + loop_length;
+//  this will set the overflow length. take the loop overflow into account when checking the loop boundaries
+    if(index_bounds > sound_length)
+    {
+        loop_overflow = index_bounds - sound_length;
+    }
+    else
+    {
+        loop_overflow = 0;
+    }
+}