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 Post subject: Voice Changer
PostPosted: Sun May 24, 2020 5:58 am 
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Service Number: A05/TQ2.0.32141E1
Country: United States
I will still have to play with things some but the voice changer will allow any number of voice effects. I am still trying to hash out the code to get a static Gou'ld or Tok'ra sounding voice, but the adjustable pot allows for many more uses as well. This will also do a decent darth vader voice as well.

This just takes Adafruit's wave shield, max4466 microphoe, adjustable potentiometer, arduino uno. and a bit of soldering.

This is based off of this tutorial:
https://learn.adafruit.com/wave-shield-voice-changer

I am trying to streamline the code a little bit more.

Code:
/* ADAVOICE is an Arduino-based voice pitch changer plus WAV playback.
   Fun for Halloween costumes, comic convention getups and other shenanigans!
Hardware requirements:
 - Arduino Uno, Duemilanove or Diecimila (not Mega or Leonardo compatible)
 - Adafruit Wave Shield, Speaker attached to Wave Shield output, Battery for portable use
If using the voice pitch changer, you will also need:
 - Adafruit Microphone Breakout, 10K potentiometer for setting pitch (or hardcode in sketch)
Software requirements: WaveHC library for Arduino, Demo WAV files on FAT-formatted SD card
Connections:
 - 3.3V to mic amp+, 1 leg of potentiometer and Arduino AREF pin
 - GND to mic amp-, opposite leg of potentiometer
 - Analog pin 0 to mic amp output
 - Analog pin 1 to center tap of potentiometer
 - Wave Shield output to speaker or amplifier
 - Wave shield is assumed wired as in product tutorial
Potentiometer sets playback pitch.  Pitch adjustment does NOT work in
realtime -- audio sampling requires 100% of the ADC.  Pitch setting is
read at startup (or reset) and after a WAV finishes playing.  POINT SPEAKER AWAY FROM MIC to avoid feedback.
Written by Adafruit industries, with portions adapted from the 'PiSpeakHC' sketch included with WaveHC library.
*/
#include <WaveHC.h>
#include <WaveUtil.h>
WaveHC    wave;  // This is the only wave (audio) object, -- we only play one at a time
#define ADC_CHANNEL 0 // Microphone on Analog pin 0. Wave shield DAC: digital pins 2, 3, 4, 5
#define DAC_CS_PORT    PORTD
#define DAC_CS         PORTD2
#define DAC_CLK_PORT   PORTD
#define DAC_CLK        PORTD3
#define DAC_DI_PORT    PORTD
#define DAC_DI         PORTD4
#define DAC_LATCH_PORT PORTD
#define DAC_LATCH      PORTD5
uint16_t in = 0, out = 0, xf = 0, nSamples; uint8_t  adc_save;// Audio sample counters, ADC Mode
// WaveHC didn't declare it's working buffers private or static,
// so we can be sneaky and borrow the same RAM for audio sampling!
extern uint8_t
  buffer1[PLAYBUFFLEN],                   // Audio sample LSB
  buffer2[PLAYBUFFLEN];                   // Audio sample MSB
#define XFADE     16                      // Number of samples for cross-fade
#define MAX_SAMPLES (PLAYBUFFLEN - XFADE) // Remaining available audio samples
#define DEBOUNCE 10            // Number of iterations before button 'takes'
//////////////////////////////////// SETUP
void setup() {
  uint8_t i; Serial.begin(9600); pinMode(2, OUTPUT); pinMode(3, OUTPUT); pinMode(4, OUTPUT); pinMode(5, OUTPUT); digitalWrite(2, HIGH);   // Chip select, Serial clock, Serial data, Latch, Set chip select high
 TIMSK0 = 0; analogReference(EXTERNAL); adc_save = ADCSRA; startPitchShift();// 3.3V to AREF, Save ADC setting, Start pitch-shift
}
void loop() { if(!wave.isplaying && !(TIMSK2 & _BV(TOIE2))) startPitchShift(); }
//////////////////////////////////// PITCH-SHIFT CODE
void startPitchShift() { // Read analog pitch setting before starting audio sampling:
  int pitch = analogRead(1); Serial.print("Pitch: "); Serial.println(pitch); nSamples = 128;
  memset(buffer1, 0, nSamples + XFADE); memset(buffer2, 2, nSamples + XFADE); // (set all samples to 512)
  TCCR2A = _BV(WGM21) | _BV(WGM20); // Mode 7 (fast PWM), OC2 disconnected
  TCCR2B = _BV(WGM22) | _BV(CS21) | _BV(CS20);  // 32:1 prescale
  OCR2A  = map(pitch, 0, 1023,
    F_CPU / 32 / (9615 / 2),  // Lowest pitch  = -1 octave
    F_CPU / 32 / (9615 * 2)); // Highest pitch = +1 octave
  // Start up ADC in free-run mode for audio sampling:
  DIDR0 |= _BV(ADC0D);  // Disable digital input buffer on ADC0
  ADMUX  = ADC_CHANNEL; // Channel sel, right-adj, AREF to 3.3V regulator
  ADCSRB = 0;           // Free-run mode
  ADCSRA = _BV(ADEN) |  // Enable ADC
    _BV(ADSC)  |        // Start conversions
    _BV(ADATE) |        // Auto-trigger enable
    _BV(ADIE)  |        // Interrupt enable
    _BV(ADPS2) |        // 128:1 prescale...
    _BV(ADPS1) |        //  ...yields 125 KHz ADC clock...
    _BV(ADPS0);         //  ...13 cycles/conversion = ~9615 Hz
  TIMSK2 |= _BV(TOIE2); // Enable Timer2 overflow interrupt
  sei();                // Enable interrupts
}
ISR(ADC_vect, ISR_BLOCK) { // ADC conversion complete
  // Save old sample from 'in' position to xfade buffer:
  buffer1[nSamples + xf] = buffer1[in]; buffer2[nSamples + xf] = buffer2[in]; if(++xf >= XFADE) xf = 0;
  // Store new value in sample buffers:
  buffer1[in] = ADCL; buffer2[in] = ADCH; if(++in >= nSamples) in = 0; } // MUST read ADCL first!
ISR(TIMER2_OVF_vect) { // Playback interrupt
  uint16_t s; uint8_t  w, inv, hi, lo, bit; int      o2, i2, pos;
  // Cross fade around circular buffer 'seam'.
  if((o2 = (int)out) == (i2 = (int)in)) {
    // Sample positions coincide.  Use cross-fade buffer data directly.
    pos = nSamples + xf; hi = (buffer2[pos] << 2) | (buffer1[pos] >> 6); lo = (buffer1[pos] << 2) |  buffer2[pos]; } //Expand 10-bit data to 12 bits
  if((o2 < i2) && (o2 > (i2 - XFADE))) {
    // Output sample is close to end of input samples.  Cross-fade to avoid click.  The shift operations here assume that XFADE is 16;
    w   = in - out; inv = XFADE - w; // Weight of sample (1-n)+ xfade
    pos = nSamples + ((inv + xf) % XFADE); s   = ((buffer2[out] << 8) | buffer1[out]) * w + ((buffer2[pos] << 8) | buffer1[pos]) * inv;
    hi = s >> 10; lo = s >> 2;// Shift 14 bit result down to 12 bits
  } else if (o2 > (i2 + nSamples - XFADE)) { // More cross-fade condition
    w   = in + nSamples - out; inv = XFADE - w; pos = nSamples + ((inv + xf) % XFADE); s   = ((buffer2[out] << 8) | buffer1[out]) * w +
          ((buffer2[pos] << 8) | buffer1[pos]) * inv; hi = s >> 10; lo = s >> 2;// Shift 14 bit result down to 12 bits   
  } else { // Input and output counters don't coincide -- just use sample directly.
    hi = (buffer2[out] << 2) | (buffer1[out] >> 6); lo = (buffer1[out] << 2) |  buffer2[out]; } // Expand 10-bit data to 12 bits
  // Might be possible to tweak 'hi' and 'lo' at this point to achieve different voice modulations -- robot effect, etc.?
  DAC_CS_PORT &= ~_BV(DAC_CS); // Select DAC Clock out 4 bits DAC config (not in loop because it's constant)
  DAC_DI_PORT  &= ~_BV(DAC_DI); // 0 = Select DAC A, unbuffered
  DAC_CLK_PORT |=  _BV(DAC_CLK); DAC_CLK_PORT &= ~_BV(DAC_CLK); DAC_CLK_PORT |=  _BV(DAC_CLK); DAC_CLK_PORT &= ~_BV(DAC_CLK);
  DAC_DI_PORT  |=  _BV(DAC_DI); // 1X gain, enable = 1
  DAC_CLK_PORT |=  _BV(DAC_CLK); DAC_CLK_PORT &= ~_BV(DAC_CLK); DAC_CLK_PORT |=  _BV(DAC_CLK); DAC_CLK_PORT &= ~_BV(DAC_CLK);
  for(bit=0x08; bit; bit>>=1) { // Clock out first 4 bits of data
    if(hi & bit) DAC_DI_PORT |=  _BV(DAC_DI);
    else         DAC_DI_PORT &= ~_BV(DAC_DI); DAC_CLK_PORT |=  _BV(DAC_CLK); DAC_CLK_PORT &= ~_BV(DAC_CLK); }
  for(bit=0x80; bit; bit>>=1) { // Clock out last 8 bits of data
    if(lo & bit) DAC_DI_PORT |=  _BV(DAC_DI);
    else         DAC_DI_PORT &= ~_BV(DAC_DI); DAC_CLK_PORT |=  _BV(DAC_CLK); DAC_CLK_PORT &= ~_BV(DAC_CLK); }
  DAC_CS_PORT    |=  _BV(DAC_CS); if(++out >= nSamples) out = 0;   // Unselect DAC
  }

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