Instructables

Arduino Audio Output

Featured
Picture of Arduino Audio Output
Generate sound or output analog voltages with an Arduino.  This Instructable will show you how to set up a really basic digital to analog converter so you can start generating analog waves of all shapes and sizes from a few digital pins on an Arduino.  (This article is a companion to another Instructable I've written about sending audio into an Arduino, find that here)

Some ideas that come to mind:

sample based instrument- store samples on the Arduino or on an SD card and trigger playback with buttons or other types of controls.  Check out my Arduino drum sampler for an idea of how to get started.
digital synthesizer- make saw, sine, triangle, pulse, or arbitrary waveshapes- check out my waveform generator to get started
MIDI to control voltage module/ MIDI synthesizer- receive MIDI messages and translate them into a voltage so you can control an analog synthesizer with MIDI, or use the MIDI data to output audio of a certain frequency
analog output- you may find yourself needing to generate analog voltages from your Arduino at some point, maybe to communicate with an analog device
effects box/digital signal processing- in combination with a microphone/audio input you can perform all kinds of digital signal manipulations and send the processed audio out to speakers.  Check out my vocal effects box for an example. 
audio playback device- make your own ipod.  With the addition of an SD shield you could create your own Arduino mp3 player (check out the wave shield documentation for an idea of how to get started with the code).  The circuits and code provided here are compatible with SD shields that communicate via SPI.

Feel free to use any of the info here to put together an amazing project for the DIY Audio Contest!  We're giving away an HDTV, some DSLR cameras, and tons of other great stuff!  The contest closes Nov 26.

Parts List:

(x9) 1/4 Watt 20kOhm Resistors Digikey 0KQBK-ND
(x7) 1/4 Watt 10kOhm Resistors Digiikey CF14JT10K0CT-ND
(x2) TS922IN Digikey 497-3049-5-ND I like these because they can be powered off the Arduino's 5V supply (one 924 works too, but they don't seem to be available on digikey at the moment)
(x1) 10kOhm potentiometer linear Digikey 987-1308-ND
(x1) 0.01uF capacitor Digikey 445-5252-ND
(x1) 220uF capacitor Digikey P5183-ND
(x1) 0.1uF capacitor Digikey 445-5303-ND
(x1) 1/4 Watt 3kOhm Resistor Digikey CF14JT3K00CT-ND
(x1) 1/4 Watt 10Ohm Resistor Digikey CF14JT10R0CT-ND
(x1) Arduino Uno Sparkfun DEV-09950

Additional Materials:
22 Gauge Wire
solder
 
Remove these adsRemove these ads by Signing Up

Step 1: Digital to Analog Converter

Picture of Digital to Analog Converter
r2r.jpg
schematic audio out arduino.jpg

DAC stands for "digital to analog converter." Since the Arduino does not have analog out capabilities, we need to use a DAC to convert digital data (numbers/ints/bytes) to an analog waveform (oscillating voltage). A simple, easy to program, and cheap way to do this is to use something called an R2R resistor ladder. Essentially, it takes incoming digital bits (0V and 5V from Arduino), weights them, and sums them to produce a voltage between 0 and 5 volts (see the schematic in fig 2, taken from the Wikipedia resistor ladder page). You can think of a resistor ladder as a multi-leveled voltage divider.

The resistor ladder I'll be demonstrating in this tutorial is an 8-bit DAC, this means it can produce 256 (2^8) different voltage levels between 0 and 5v. I connected each of digital pins 0-7 to each of the 8 junctions in my 8 bit DAC (shown in figs 1 and 3).

I like using these resistor ladder DACs because I always have the materials around, they're cheap, and I think they're kind of fun, but they will not give you the highest quality audio. You can buy a chip that works in the exact same was as an R2R DAC (and will work with all the code in this instructable), but has internal, highly-matched resistors for better audio quality, I like this one bc it runs off a single 5V supply (you can even do stereo audio with it), but there are many more available, look for "parallel input, 8 bit, dac ic".

Alternatively, there are chips that take in serial data to perform digital to analog conversion. These chips are generally higher fidelity (definitely better quality that the resistor ladder DAC) and they only use two or three of the Arduino's output pins (as opposed to 8). Downsides are they are a little more challenging to program, more expensive, and will not work with the code in this Instructable, though I'm sure there are some other tutorials available. After a quick search on digikey, these looked good, for Arduino, try to find something that will run off a single 5V supply.

One more note - there seems to be kind of a misconception abut 8 bit audio- that it always has to sound like the sounds effects from a Mario game- but 8bit audio with this really basic DAC can actually replicate the sounds of people's voices and instruments really well, I'm always amazed at the quality of sound that can come from a bunch of resistors.

Step 2: Set up DAC and Test

IMG_0066 copy.jpg
IMG_0065 copy.jpg
IMG_0048 copy.jpg
IMG_0047 copy.jpg
IMG_0049 copy.jpg
I constructed my DAC on a breadboard (figs 1-3).  The schematic is given in fig 8.  Below are a few pieces of sample code that generate the waveforms shown in figs 4-7.  In the following pieces of code I send a value between 0 and 255 to "PORTD" when I want to send data to the DAC, it looks like this:

PORTD = 125;//send data to DAC

This is called addressing the port directly.  On the Arduino, digital pins 0-7 are all on port d of the Atmel328 chip.  The PORTD command lets us tells pins 0-7 to go HIGH or LOW in one line (instead of having to use digitalWrite() eight times).  Not only is this easier to code, it's much faster for the Arduino to process and it causes the pins to all change simultaneously instead of one by one (you can only talk to one pin at a time with digitalWrite()).  Since port d has eight pins on it (digital pins 0-7) we can send it one of 2^8 = 256 possible values (0-255) to control the pins.  For example, if we wrote the following line:

PORTD = 0;

it would set pins 0-7 LOW.  With the DAC set up on pins 0-7 this will output 0V.  if we sent the following:

PORTD = 255;

it would set pins 0-7 HIGH.  This will cause the DAC to output 5V.  We can also send combinations of LOW and HIGH states to output a voltage between 0 and 5V from the DAC.   For example:

PORTD = 125;
125 = 01111101 in binary.  This sets pin 7 low (the msb is 0), pins 6-2 high (the next five bits are 1), pin 1 low (the next bit is 0), and pin 0 high (the lsb is 1).  You can read more about how this works here.  To calculate the voltage that this will output from the DAC, we use the following equation:

voltage output from DAC = [ (value sent to PORTD) / 255 ] * 5V
so for PORTD = 125:
voltage output from DAC = ( 125 / 255 ) * 5V = 2.45V

The code below sends out several voltages between 0 and 5V and holds each for a short time to demonstrate the concepts I've described above.  In the main loop() function I've written:

  PORTD = 0;//send (0/255)*5 = 0V out DAC
  delay(1);//wait 1ms
  PORTD = 127;//send (127/255)*5 = 2.5V out DAC
  delay(2);//wait 2ms
  PORTD = 51;//send (51/255)*5 = 1V out DAC
  delay(1);//wait 1ms
  PORTD = 255;//send (255/255)*5 = 5V out DAC
  delay(3);//wait 3ms


The output is shown on an oscilloscope in fig 4.  The center horizontal line across the oscilloscope represents 0V and each horizontal line represents a voltage increase/decrease of 2V.  The image notes on fig 4 show the output of each of the lines of code above, click on the image to view the image notes.


The code below outputs a ramp from 0 to 5V.  In the loop() function, the variable "a" is incremented from 0 to 255.  Each time it is incremented, the value of "a" is sent to PORTD.  This value is held for 50us before a new value of "a" is sent.  Once "a" reaches 255, it gets reset back to 0.  The time for each cycle of this ramp (also called the period) takes:

period = (duration of each step) * (number of steps)
period = 50us * 256 = 12800us = 0.0128s


so the frequency is:
frequency of ramp = 1/0.0128s = 78Hz

The output from the DAC on an oscilloscope can be seen in fig 5.

The code below outputs a sine wave centered around 2.5V, oscillating up to a max of 5V and a min of 0V.  In the loop() function, the variable "t" is incremented from 0 to 100.  Each time it is incremented, the expression:
127+127*sin(2*3.14*t/100)
is sent to PORTD.  This value is held for 50us before "t" is incremented again and a new value is sent out to PORTD.  Once "t" reaches 100, it gets reset back to 0.  The period of this sine wave should be:

period = (duration of each step) * (number of steps)
period = 50us * 100 = 5000us = 0.005s


so the frequency should be:
frequency of ramp = 1/0.005s = 200Hz



But this is not the case, the output from the DAC is shown in fig 6.  As indicated in the image notes, it does not have a frequency of 200hz, its frequency is more like 45hz.  This is because the line:

PORTD = 127+127*sin(2*3.14*t/100);
takes a very long time to calculate.  In general multiplication/division with decimal numbers and the sin() function take the Arduino a lot of time to perform.

One solution is to calculate the values of sine ahead of time and store them in the Arduino's memory.  Then when the Arduino sketch is running all the Arduino will have to do is recall these values from memory (a very easy and quick task for the Arduino).  I ran a simple Python script (below) to generate 100 values of 127+127*sin(2*3.14*t/100):

import math
for x in range(0, 100):
print str(int(127+127*math.sin(2*math.pi*x*0.01)),)+str(","),


I stored these values in an array called "sine" in the Arduino sketch below.  Then in my loop, for each value of "t" I sent an element of sine[] to PORTD:

PORTD = sine[t];

The output from this DAC for this sketch is shown in fig 7.  You can see that it outputs a sine wave of 200hz, as expected.

Step 3: DAC Buffer

Picture of DAC Buffer
IMG_0082 copy.jpg
IMG_0083 copy.jpg
IMG_0077 copy.jpg
IMG_0084 copy.jpg
schematic audio out arduino.jpg
Now that we have a good signal coming out Arduino, we need to protect it.  The R2R DAC is very sensitive to any loads put on it, so trying to drive speakers directly from the DAC will distort the signal heavily.  Before doing anything with the signal you need to set up some kind of buffer circuit.  I set up one of the op amps in the TS922 dual op amp package as a voltage follower to buffer my DAC from the rest of my circuit (see schematic in fig 6, be sure to power the op amp with 5V and ground).

Once this was set up I wired an LED and 220ohm resistor in series between the output of the op amp and ground.  The sketch below outputs a slow ramp out the DAC so you can actually see the LED get brighter as the ramp increases in voltage.  The period of the ramp is:

period = (duration of each step) * (number of steps)
period = 5ms * 256 = 1280ms = 1.28s

so the LED takes 1.28 seconds to ramp up from off to full brightness.

Step 4: Low Pass Filter

Picture of Low Pass Filter
IMG_0103 copy.jpg
IMG_0104 copy.jpg
IMG_0150 copy.jpg
schematic audio out arduino.jpg
The purpose of a low pass filter is to smooth out the output of the DAC in order to reduce noise.  By using a low pass filter on the signal, you can smooth out the "steps" in your waveform while keeping the overall shape of the waveform intact (see fig 4).  I used a simple RC flow pass filter to achieve this: a resistor and a capacitor in series to ground.  Connect the resistor to the incoming signal and the capacitor to ground, the signal coming from the junction between these two components will be low pass filtered.  I sent this filtered signal into another buffer circuit (I wired an op amp in a voltage follower configuration) to protect the filtered signal from any loads further down in the circuit.  See the schematic in fig 5 for more info.

You can calculate the values of the capacitor and resistor you need for a low pass filter according to the following equation:

cutoff frequency = 1/ (2*pi*R*C)

Nyquist's Theroum states that for a signal with a sampling rate of x Hz, the highest frequency that can be produced is x/2 Hz.  You should set your cutoff frequency to x/2Hz (or maybe slightly lower depending on what you like).  So if you have a sampling rate of 40kHz (standard for most audio), then the maximum frequency you can reproduce is 20kHz (the upper limit of the audible spectrum), and the cutoff frequency of your low pass filter should be around 20kHz.

For a cutoff frequency of 20,000Hz and 1kOhm resistor:
20000=1/(2*3.14*1000*C)
C =~ 8nF


since 8nF capacitors are hard to come by I rounded up to 0.01uF.  This gives a cutoff frequency of about 16kHz.  You can mess around with different values and see what you like best, I tend to like heavier filtering because it removes more unwanted noise.
 

Step 5: Signal Amplitude

Picture of Signal Amplitude
IMG_0098 copy.jpg
IMG_0099 copy.jpg
Next I added a potentiometer to control the amplitude of my signal.  To do this I wired the output from the 2nd voltage follower to one side of a 10k potentiometer.  The I wired the other side of the pot to ground.  The signal coming out from the middle of the pot has an adjustable amplitude (between 0 and 2.5V) depending on where the pot is turned.  See the schematic (fig 7) for more info.  You can see the output of the signal before the pot and after the pot (when turned to halfway point) in fig 6.

Step 6: Amplifier

Picture of Amplifier
IMG_0113 copy.jpg
IMG_0114 copy.jpg
IMG_0116 copy.jpg
schematic audio out arduino.jpg
Many times when we talk about amplifiers we think about circuits which increase the amplitude of a signal.  In this case I'm talking about increasing the current of the signal so that it can drive a load (like a speaker).  In this stage of the circuit I set up both op amps on one TS922 package as parallel voltage followers.  What this means is I sent the output from the amplitude pot to the non-inverting input of both op amps.  Then I wired both op amps as voltage followers and connected their outputs to each other.  Since each op amp can source 80mA of current, combined they can source 160mA of current.

Step 7: DC Offset

Picture of DC Offset
IMG_0144 copy.jpg
IMG_0142 copy.jpg
schematic audio out arduino.jpg
Before sending a signal to speakers, you want to make sure it is oscillating around 0V (typical of audio signals).  So far, the Arduino DAC output we've been dealing with is oscillating around 2.5V.  To fix this we can use a big capacitor.  As indicated in the schematic, I used a 220uF capacitor to DC offset my signal so that it oscillates around 0V.  The output of the DC offset signal (blue) and un-offset signal (yellow) for two different amplitudes can be found in figs 2 and 3.

Step 8: Output

Picture of Output
IMG_0140 copy.jpg
IMG_0141 copy.jpg
IMG_0010 copy.jpg
IMG_0009 copy.jpg
schematic audio out arduino.jpg
Finally, I wired up a 1/4" mono jack with two wires.  I connected the ground lead to the Arduino's ground and the signal lead to the negative lead of the 220uF capacitor.  The ground pin is usually the larger pin on the jack, test for continuity with the threaded portion of the jack to make sure that you have located the ground pin correctly (see fig 5).  The signal pin will be continuous with the clip that extends out from the jack (fig 5).  See the schematic for more info.

Step 9: 40kHz Sampling Rate

For those of you who are interested in producing audio at 40kHz sampling rate, here is some code that uses timer interrupts to let you do that.  Arduino timer interrupts allow you to pause what you are doing in your main loop() function and jump to a special function called an "interrupt routine."  Once this routine is done you come back to where you left off in the loop().  You set up and specify the frequency of these interrupts in the setup() part of your code.  You can learn the specifics of setting up interrupts here, but if you are only interested in 40kHz interrupts, then you can just copy parts of the code below.

To set up the interrupt you need to copy the following lines into your setup() function:
 
  cli();//disable interrupts
  //set timer0 interrupt at 40kHz
  TCCR0A = 0;// set entire TCCR0A register to 0
  TCCR0B = 0;// same for TCCR0B
  TCNT0  = 0;//initialize counter value to 0
  // set compare match register for 40khz increments
  OCR0A = 49;// = (16*10^6) / (40000*8) - 1 (must be <256)
  // turn on CTC mode
  TCCR0A |= (1 << WGM01);
  // Set CS11 bit for 8 prescaler
  TCCR0B |= (1 << CS11);
  // enable timer compare interrupt
  TIMSK0 |= (1 << OCIE0A);
  sei();//enable interrupts


the contents of the interrupt routine are encapsulated in the following function:

ISR(TIMER0_COMPA_vect){ //40kHz interrupt routine
}


You want to keep the interrupt routine as short as possible, only the necessities.  You can do all of your other tasks (checking on buttons, turning on leds, etc) in the loop().  Also keep in mind that setting up interrupts may affect other Arduino functions such as analogWrite and delay.

In the code below, I use the interrupt function to send a new value of sine[] to PORTD at a rate of 40kHz and increment the variable "t."  Figs 1 and 2 show the (unfiltered) output of the code on an oscilloscope.  We can calculate the expected frequency as follows:

frequency = (sampling frequency) / (steps per cycle)
frequency = 40,000 / 100 = 400hz


at a sampling frequency of 40kHz we expect the duration of each step to be:

duration of each sample step = 1/(sampling frequency)
duration of each sample step = 1/40,000 = 25us

Step 10: Extra Tips

This DAC uses quite a bit of the Arduino's available digital pins, including some that are normally used for serial communications and PWM, so here are a few tips that will help you deal with pin conflicts.

If you want to do serial communication: Software Serial is an Arduino library that allows you to turn any of the Arduino's pins into serial pins.  Usually when you are doing an Arduino project that requires serial communication, you avoid using digital pins 0 and 1 because they need to be free to send serial data.  I like to use them for the 8 bit DAC because pins 0-7 are all part of PORTD on the Arduino's Atmel328 chip, this allows me to address all of them in a single line of code.  PORTB only has 6 pins (digital pins 8-13) and PORTC only has 6 pins (analog pins 0-5), so you cannot construct an 8 bit DAC with these ports alone.

If you need to use the PWM pins, or otherwise need to use different pins as the DAC: If you must use the PWM pins you can use bit manipulation to free up pins 3, 5, and 6 and replace them with pins 8, 12, and 13.  Say you want to send the number 36 to PORTD.  You can use the following lines:

//define variables:
boolean bit3state;
boolean bit5state;
boolean bit6state;


//in your main loop():

bit3state = (36 & B00001000)>>3;//get the third bit of 36
bit5state = (36 & B00100000)>>5;//get the fifth bit of 36
bit6state = (36 & B01000000)>>6;//get the sixth bit of 36


//send data to portd w/o disrupting pins 3, 5, and 6
PORTD |= (36&B10010111);//set high pins high using the number 36 with zeros replacing bits 3, 5, and 6
PORTD &= (36|B01101000);//set low pins low using the number 36 with ones replacing bits 3, 5, and 6

//send data to portb w/o disrupting pins 9, 10, and 11
PORTB |= 0 | (bit3state) | (bit5state<<4) | (bit6state<<5);//set high pins
PORTB &= 255 & ~(1-bit3state) & ~((1-bit5state)<<4) & ~((1-bit6state)<<5);//set low pins


be sure to keep these PORTD and PORTB lines right next to each other in your code, you want the pins on port d and port b to switch at as close to the same time as possible.

Here is the code from the previous step, edited so that it does not use any PWM pins.  As you see in fig 1, the unfiltered output from the DAC has many discontinuities caused by the lag between sending data to port d and port b, as well as splitting up the commands for setting pins high and low.  You can get rid of most of these discontinuities with the low pass filter (fig 2).  If you wanted to use this technique you might consider increasing the cutoff frequency of your low pass filter.  If you wanted to make this really good, you could send your 5 most significant bits to port d and your 3 least significant bits to port b.  This would decrease the amplitude of some of the discontinuities, reducing the magnitude of the noise.  I'll let you figure that one out on your own.

If you run out of digital pins and need more:  Remember you can always use your analog pins as Digital I/O.  Try out the following functions, they work just like you are dealing with a regular digital pin.

digitalWrite(A0,HIGH);//set pin A0 high
digitalWrite(A0,LOW);//set pin A0 low
digitalRead(A0);//read digital data from pin A0

Otherwise, try using a multiplexer.  If you need more digital outputs, the 74HC595 allows you to turn three of the Arduino's digital pins into 8 outputs.  You can even daisy chain multiple 595's together to create many more outputs pins.  You could set up your whole DAC on one of these chips if you wanted (though it would take a few lines of code to address it and might slow you down too much for higher sampling rates).  The Arduino website is a good place to start learning about how to use the 595.

If you need more digital inputs, the 74HC165 or CD4021B let you turn three of the Arduino's digital pins into 8 inputs.  Again, the Arduino website is a good place to start learning how to use these chips.

If you want to use the info in this Instructable with the Mega or other boards:  In this Instructable I talked exclusively about the Arduino Uno with Atmel328.  The same code will run fine on any board with an Atmel328 or Atmel168 chip on it.  You can also use the same ideas with a Mega.  You should try to attach your DAC to any port that has 8 available pins, that way you can address your DAC with one line of code ("PORTD =" )  On the Uno, the only port that has 8 available pins is port d.   This picture indicates that the Mega has several ports with 8 pins: ports a, b, c, and l are the obvious choices.  If you don't care about wasting analog pins you could also use ports f or k.
Reddyco1 month ago

Hey Amanda! Great job on this instructable, I'm using it to embase my work on a eletronic drum sound generator with Arduino (hopefully, one day I'll post here how to do it).

I'm writing to ask abuot that 0.01 uF and 10ohm resistor in parallel with the speaker and the DC offset capacitor. What are they used for?

Thanks in advance!

amandaghassaei (author)  Reddyco24 days ago

thanks! they're just threre to reduce noise, not a big deal if you don't have them

flowirin1 month ago

what's the upper limit on the sampling frequency for the arudino? can i get it up to 80kHz?

amandaghassaei (author)  flowirin24 days ago

definitely, you could get up to a few hundred kHz with no problem.

One concern is that the resistor ladder dac that I show in this ible might not respond fast enough as you increase the sampling rate - this might end up applying a low pass filter on your output. I updated step 1 with a little more info about alternative DACs, you might check out the R2R DAC IC I demonstrated in this Instructable (you can wire it up to only use one channel if you need), it has much better quality control then just throwing a bunch of resistors together on a breadboard and I think it will give you better results. It says the settling time for that DAC is 100ns, which should work fine for 80kHz sampling rate.

you will also have to change the frequency of the interrupt. For 40kZ I used this line:

OCR0A = 49;// = (16*10^6) / (40000*8) - 1

try this instead:

OCR0A = 24;// = (16*10^6) / (40000*8) - 1

ibirnam6 months ago
Just wanted to confirm this: since the TS922IN is now obsolete, would this be a sufficient replacement? Thank you!
amandaghassaei (author)  ibirnam6 months ago
that's going to be really hard to work w bc it's surface mount. Just get the lm386 chip and a couple of resistors and capacitors and wire it up like this:
http://www.hobby-hour.com/electronics/lm386-20.gif
it may need a 9v supply instead of 5v, I can't remember.

I'm working my way through this trying to substitute the lm386 at Step 3. The gif you linked looks like the replacement for the amplifier. I can see the part that replaces the low pass filter (thanks to your excellent explanation of what that is) but I don't see anything that I recognize as being the DAC buffer. Is the buffer unnecessary with the lm386, or is it there and I don't recognize it?
Thanks!
PS I'm a software engineer, so use small words. :-)

amandaghassaei (author)  heymarky1 month ago

actually, a resistor and capacitor only act as a low pass filter when the output signal is connected to the junction between them, here is a pic. You can see that switching the order of the components will turn it into a high pass filter. The lm386 circuit is not wired up the same way, so it won't act as a low pass filter. So here's what I would do:

arduino - dac - lm386 - low pass filter - output

you could also use a tl1072 or tl082 to replace both ts922's, but these require a +/- 9v supply, which is annoying.

Awesome, thanks for the help!

Ibirnam Is LM386 working instead of using TS922IN ?

joshuaphua12 months ago

near the end of the circuit, what's the 0.1uF capacitor and 10Ohm resistor for? another kind of filter?

amandaghassaei (author)  joshuaphua12 months ago

not a filter, just helps make the DC offset more stable.

joshuaphua13 months ago
Can you explain the DC offset more? I understand how a +2.5VDC offset works but am confused about this one. Thanks!
amandaghassaei (author)  joshuaphua13 months ago

I always think of it like this: the signal going into one side of the capacitor causes an alternating excess of positive or negative charge on one side of the cap. The other side of the cap reacts by accumulating opposite charge - this causes an alternating voltage on the opposite side of the cap. Since no current (or a negligible amount) actually gets passed across the cap, the DC voltage on one side does not transfer over to the other side, so the alternating voltage is centered around 0.

thanks!!

punk12904 months ago
I had to replace the TS922 with an LM386. I am trying to get step 3 working. Unfortunately I don't have an oscillator to verify I did things right in step 2. My issue is that when I connect my Adruino to my breadboard with this circuit the power LED dims and my computer no longer sees the Arduino. I tried unhooking the breadboard and then loading the program to my Arduino. That worked fine. When I connected everything back up, the LED came on immediately without ramping up. Any thoughts or ideas for me?
amandaghassaei (author)  punk12903 months ago

sounds like you're shorting out one of the arduino's power pins (the strip of pins near the analog inputs). Double check those.

Ploopy4 months ago
How did you write your code in those squares that you scroll?
  
amandaghassaei (author)  Ploopy3 months ago

it's a feature that's actually no longer supported, sorry! we're looking at new ways to make code easier to embed in the editor, it will happen.

avionics28 months ago
Thank you so much for making it clear now I understand everything very clearly and thank you for making such a concise tutorial. My hat off to you.

take care and please give us more tutorials.
amandaghassaei (author)  avionics26 months ago
thanks!
These chips come in handy if you do this alot. And they can be dasychained to increase the resolution.
http://www.digikey.com/product-detail/en/4116R-R2R-253LF/4116R-R2R-253LF-ND/2565168
I made a 32 bit one for my propeller, but frankly couldn't tell much of a difference.
elbiot7 months ago
Doesn't the Arduino UNO have PWM out? Is there a reason you use an external DAC instead of the PWM?
amandaghassaei (author)  elbiot6 months ago
yes it does, but the PWM only does 0 and 5V output, I wanted to output analog voltages so I could make any waveform shape.
um, you mean you want waveforms at ultrasonic frequencies? The PWM is for putting out any analog waveform, but it seems its a bit slower than the PCM that you use. The digital outs in your Pulse Code Modulation r2r setup only put out 0 or 5 volts, but this leads to an analog voltage of 0-5v. Pulse Width Modulation out just needs a capacitor and resistor (low pass filter) to do the same.
elbiot elbiot6 months ago
Ah, I see its not that simple for good audio. Lots of examples used 8kHz sample rates and things like that. This clever person used dual PWM to get 16 bit depth at a good sample rate. Hardware is still super simple. http://forum.arduino.cc/index.php?topic=142824.0
AntonDan8 months ago
Hi! That is a really nice guide :). I've got a question though.. The output is 8 bit right? I want to make a vocal effect (mostly distortion) pedal.. Will the 8 bit output reduce the quality of the sound (or something like that)? Thanks :)
amandaghassaei (author)  AntonDan8 months ago
yes, 8-bit (especially out an r2r dac like this) is going to sound noticeably distorted/noisy, but it is still a pretty good approximation of the original signal. here is an example:
http://www.instructables.com/id/Arduino-Vocal-Effects-Box/
avionics28 months ago
Great Amanda ! that made it clear now here it comes the second question hope you don't mind. in OCRA1 you put 49 , can you explain how you reached to number 49?

thanks for the good article.
amandaghassaei (author)  avionics28 months ago
my comment on that line has an error, it should say
(16*10^6) / (40000*8) - 1
which means the arduino has a clock speed of 16Mhz (16*10^6), and I set the prescaler to 8 (divide by 8) and the frequency I want is 40000 (divide by 40000) and all numbers are zero indexed (-1)
(16*10^6) / (40000*8) - 1 = 49
avionics28 months ago
Amanda can you explain where you get the numbers in your sine[] ?
amandaghassaei (author)  avionics28 months ago
I wrote a little python script to generate them, nothing fancy, just a bunch of values of 127+127*sin(x)
qwertyfinger9 months ago
Amazing tutorial! Thank you so much, I'm using an arduino uno to make a musical sequencer and hopefully this will help me a lot!
also, i'm assuming that if I want to use 16 bit sound i just increase the length of the ladder? i'm using a shift register for serial output because i don't have rapidly changing sound and if i weren't i'd need something like 42 digital I/O pins in total!
amandaghassaei (author)  qwertyfinger9 months ago
theoretically yes, but after 8 bit the noise caused by adding more resistors in your ladder makes the increased resolution kind of pointless. I'd recommend getting an r2r dac ic for this they might even make 16 bit versions. The resistors in those dacs are calibrated to reduce noise.
hpan9 months ago
thank you so much! you're pretty awesome
hpan9 months ago
hola. Amanda.
when you create a sine wave, why do you use
"PORTD = 127+127*sin(2*3.14*t/100)" and let t runs from 0 to 100? I'm a little confused about that. please help. thank you
amandaghassaei (author)  hpan9 months ago
it's not 0 to 100, it's 0 to 255, where 255 is a 5V output and 0 is a 0V output (127 is 2.5V and so on). the sin function always returns something between -1 and 1, so the equation will only fall between 127+127 =~256 and 127 - 127 = 0
faziefazie1 year ago
can I using this mono jack instead of yours?
mono_jacks.jpg
amandaghassaei (author)  faziefazie1 year ago
sure
Pro

Get More Out of Instructables

Already have an Account?

close

PDF Downloads
As a Pro member, you will gain access to download any Instructable in the PDF format. You also have the ability to customize your PDF download.

Upgrade to Pro today!