hi-
the 'current' sketch of 'mine' is altered a bit.. (but reading through this thread, should tell you what needs to be 'changed back' to original settings..
my project is a 'custom' board/version and had custom DAC pins used, as well as no AREF pin broken out..
(leaving both of those as default should still get you a working sketch)
there is NO keypad
this sketch DOES require an SD card to be inserted/initialized (I dont believe Phill posted a version that was standalone, not needing the SD to be inserted)
this sketch should have a 'start.wav' file.. (this plays once the board boots/loads SD card... kinda telling you.. "I'm ready").
this sketch DOES NOT need any PI audio files on it.. (not related to the PI sketch at all)
"I" chose to just use the same breakout mic form the sketch..
I wasnt sure if a stand alone mic without any of the added components (caps, resistors) would be as good/effective (or more noise..etc)
Actually.. looking at my last post of some code..
I dont think there is much/anything to change except for reverting back to the old DAC pin values:
You are using a regular Arduino and a true Adafruit WaveShield..yes? (dont forget to read tut again.. and do not forget to connect AREF pin)
/*
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)
If using the WAV playback, you will also need:
- SD card
- Keypad, buttons or other sensor(s) for triggering sounds
Software requirements:
- WaveHC library for Arduino
- Demo WAV files on FAT-formatted SD card
This example sketch uses a 3x4 keypad for triggering sounds...but with
some changes could be adapted to use several discrete buttons, Hall effect
sensors, force-sensing resistors (FSRs), I2C keypads, etc. (or if you just
want the voice effect, no buttons at all).
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
- Matrix is wired to pins A2, A3, A4, A5 (rows) and 6, 7, 8 (columns)
- 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>
SdReader card; // This object holds the information for the card
FatVolume vol; // This holds the information for the partition on the card
FatReader root; // This holds the information for the volumes root directory
FatReader file; // This object represent the WAV file for a pi digit or period
WaveHC wave; // This is the only wave (audio) object, -- we only play one at a time
#define error(msg) error_P(PSTR(msg)) // Macro allows error messages in flash memory
#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; // Audio sample counters
uint8_t adc_save; // Default 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
const char *sound[] = {
"startup" };
//////////////////////////////////// SETUP
void setup() {
uint8_t i;
Serial.begin(9600);
// The WaveHC library normally initializes the DAC pins...but only after
// an SD card is detected and a valid file is passed. Need to init the
// pins manually here so that voice FX works even without a card.
pinMode(17, OUTPUT); // Chip select
digitalWrite(17, HIGH); // Set chip select high
pinMode(4, OUTPUT); // Serial clock
pinMode(7, OUTPUT); // Serial data
pinMode(8, OUTPUT); // Latch
//test for trunign on CS pin for SD card if needed
//pinMode(10, OUTPUT);
// Init SD library, show root directory. Note that errors are displayed
// but NOT regarded as fatal -- the program will continue with voice FX!
if(!card.init()) SerialPrint_P("Card init. failed!");
else if(!vol.init(card)) SerialPrint_P("No partition!");
else if(!root.openRoot(vol)) SerialPrint_P("Couldn't open dir");
else {
PgmPrintln("Files found:");
root.ls();
// Play startup sound (last file in array).
playfile(sizeof(sound) / sizeof(sound[0]) - 1);
}
// Optional, but may make sampling and playback a little smoother:
// Disable Timer0 interrupt. This means delay(), millis() etc. won't
// work. Comment this out if you really, really need those functions.
TIMSK0 = 0;
// Set up Analog-to-Digital converter:
analogReference(EXTERNAL); // 3.3V to AREF
adc_save = ADCSRA; // Save ADC setting for restore later
while(wave.isplaying); // Wait for startup sound to finish...
startPitchShift(); // and start the pitch-shift mode by default.
}
//////////////////////////////////// LOOP
// As written here, the loop function scans a keypad to triggers sounds
// (stopping and restarting the voice effect as needed). If all you need
// is a couple of buttons, it may be easier to tear this out and start
// over with some simple digitalRead() calls.
void loop() {
//
}
//////////////////////////////////// HELPERS
// Open and start playing a WAV file
void playfile(int idx) {
char filename[13];
(void)sprintf(filename,"%s.wav", sound[idx]);
Serial.print("File: ");
Serial.println(filename);
if(!file.open(root, filename)) {
PgmPrint("Couldn't open file ");
Serial.print(filename);
return;
}
if(!wave.create(file)) {
PgmPrintln("Not a valid WAV");
return;
}
wave.play();
}
//////////////////////////////////// PITCH-SHIFT CODE
void startPitchShift() {
// Read analog pitch setting before starting audio sampling:
//int pitch = analogRead(1);
int pitch = 512;
Serial.print("Pitch: ");
Serial.println(pitch);
// Right now the sketch just uses a fixed sound buffer length of
// 128 samples. It may be the case that the buffer length should
// vary with pitch for better results...further experimentation
// is required here.
nSamples = 128;
//nSamples = F_CPU / 3200 / OCR2A; // ???
//if(nSamples > MAX_SAMPLES) nSamples = MAX_SAMPLES;
//else if(nSamples < (XFADE * 2)) nSamples = XFADE * 2;
memset(buffer1, 0, nSamples + XFADE); // Clear sample buffers
memset(buffer2, 2, nSamples + XFADE); // (set all samples to 512)
// WaveHC library already defines a Timer1 interrupt handler. Since we
// want to use the stock library and not require a special fork, Timer2
// is used for a sample-playing interrupt here. As it's only an 8-bit
// timer, a sizeable prescaler is used (32:1) to generate intervals
// spanning the desired range (~4.8 KHz to ~19 KHz, or +/- 1 octave
// from the sampling frequency). This does limit the available number
// of speed 'steps' in between (about 79 total), but seems enough.
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
}
void stopPitchShift() {
ADCSRA = adc_save; // Disable ADC interrupt and allow normal use
TIMSK2 = 0; // Disable Timer2 Interrupt
}
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; // MUST read ADCL first!
buffer2[in] = ADCH;
if(++in >= nSamples) in = 0;
}
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); // Expand 10-bit data
lo = (buffer1[pos] << 2) | buffer2[pos]; // 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;
// will need adjustment if that changes.
w = in - out; // Weight of sample (1-n)
inv = XFADE - w; // Weight of xfade
pos = nSamples + ((inv + xf) % XFADE);
s = ((buffer2[out] << 8) | buffer1[out]) * w +
((buffer2[pos] << 8) | buffer1[pos]) * inv;
hi = s >> 10; // Shift 14 bit result
lo = s >> 2; // 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; // Shift 14 bit result
lo = s >> 2; // down to 12 bits
}
else {
// Input and output counters don't coincide -- just use sample directly.
hi = (buffer2[out] << 2) | (buffer1[out] >> 6); // Expand 10-bit data
lo = (buffer1[out] << 2) | buffer2[out]; // 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); // Unselect DAC
if(++out >= nSamples) out = 0;
}