How to play audio from a microcontroller

COLLEGE OF ENGINEERING TRIVANDRUM

Mahadev G

+919496370662

+919632920773

[email protected]

HOW TO PLAY AUDIO FROM A

MICROCONTROLLER

INTRODUCTION TO SOUND

ROBOCET , COLLEGE OF ENGINEERING

TRIVANDRUM

• In physics, sound is a vibration that propagates as a typically audible mechanical wave of

pressure and displacement, through a medium such as air or water.[1]

• So sound is a collection of

compression and

rarefaction of pressure of a

single or varying frequencies

ELECTRONIC SOUND - SPEAKER


TRIVANDRUM

• A loudspeaker (or loud-speaker or speaker) is an electroacoustic transducer; a device

which converts an electrical audio signal into a corresponding sound.[2]

• The sound as an electrical signal is a collection of waves of different frequencies, but

audible sound(for humans) is an electrical signal having frequencies in [20Hz,20kHz] ; but

peak audible frequencies lie between [2kHz,5kHz]

http://en.wikipedia.org/wiki/Loudspeaker#cite_note-Talbot-Smith-2


TRIVANDRUM

• So sound is pretty much an analog signal; which is an infinite data set. For the signal to be

understood by microcontrollers it needs to be digital and finite. To make it finite we must

sample the sound with an interval so that the information of the analog signal is not lost.

• We’ll go to Nyquist theorem for this, so for a signal not be aliased (corrupted) the minimum

sampling frequency should be 2*Fbw, where Fbw is the maximum frequency component present

in the signal.

• So lets analyse a song.

SPECTRUM OF “IN THE END”– LINKIN PARK


TRIVANDRUM

• Spectrum Curtsy Audacity


TRIVANDRUM

• So looking at the spectrum of the song, there is an attenuation of -30dB at 4kHz and a -37dB

attenuation at 8kHz, after that the attenuation just increases. The more the attenuation the

less we hear.

From this we can infer that a song has less of high frequency components that we actually

hear.

• Most of the sound we hear are under the 8kHz band.

• So using our good old Nyquist sampling theorem a sampling frequency of 8kHz*2 would be

good enough to reproduce the song.[or any audible sound for that matter]

SAMPLED AT 16KHZ


TRIVANDRUM

• So looking at the spectrum of the

16kHz sampled song, you find that

the core sound of the song is

retained without loss of information

HOW SOUND IS SAVED..


TRIVANDRUM

• Now that the sound has become finite, we

need to save it. Each sample has an analog

value/amplitude associated to it. This has to

be captured, this is done by ADC’s and

saved as 8,16 or 32 bit data depending upon

the quality of the sound.

• So putting it all together, we have for each

second of a song we have 16000 8/16/32 bit

data. This is how a .wav file is represented.

This kind of modulation is called pulse code

modulation.

HOW TO RECONSTRUCT THE ANALOG

SIGNAL FROM THE DIGITAL AUDIO SAMPLES


TRIVANDRUM

• Since each sample stored represents the amplitude of the signal at that particular instant,

we will use Pulse Width modulation to recreate the signal amplitude.

• Why pulse width modulation??

• It’s the easiest way to produce an analog voltage from a digital signal.

• How?

• Explained better in the link http://arduino.cc/en/Tutorial/PWM

INTERPOLATING..


TRIVANDRUM

• Now you have at t=1/16000 having amplitude x1 and t=2/16000 having amplitude x2, to

make the signal continuous the space between these two times should have something !!

• So you flood it with a PWM signal having an average of x1 from time time t=1/16000 to

t=2/16000, so that the average between these two sample times is x1, and similarly

between t=2/16000 to t=3/16000 with a PWM signal of average x2.

• For that your PWM frequency should be higher than 16kHz. Choose a frequency 10 times

greater so that at least 10 waves can fit in between two adjacent samples.

• Choosing a higher frequency for the PWM signal can greatly improve the noise

characteristics of your audio since the more the frequency of the PWM signal the more

the noise is being pushed out of the frequency band we are using. [ How?? See last slide]

THE NUTSHELL..


TRIVANDRUM

DO WE NEED A LOW PASS FILTER AT THE END?


TRIVANDRUM

• Actually the low pass filter is not required as if

you look at the equivalent circuit of the standard

loud speaker, you see that the R1 and C1 forms

an LPF

• But an Amplifier might be required if you are

trying to drive powerful loudspeakers directly

using a microcontroller output pin.

• For a standard headphone the microcontroller

pin drive strength is good enough to reproduce

sound with good volume.

HOW TO GET THE AUDIO DATA FROM A SONG..


TRIVANDRUM

1. Install the open source app “Audacity”

http://audacity.sourceforge.net/

2. Open any song in it, and export it as

.wav file (screen shot).

3. Reopen the .wav file

4. Goto Analyse menu-> Sample data

Export

You have a lot of handles from there to

export, get the signal amplitude in numbers

or in dB scale; export to csv or txt file format

http://audacity.sourceforge.net/

THE ALGORITHM IN THE

MICROCONTROLLER..


TRIVANDRUM

LIMITING FACTORS


TRIVANDRUM

• The major limiting factor is definitely the code memory of the microcontroller, Each

sample takes a minimum of 8bit of space, and since we have 16k samples in a second we

have 16k * 8 bits of space for a second. Which is sort of the max limit for most

microcontrollers available under 3$.

• Using a sampling frequency of 8k would double the amount of sound that you could play

without compromising much on quality. But if your sounds are very low frequency even 6k

would work, but would not recommend using sub 8k frequencies for this application.

• So if you want to play a whole song interface an SRAM or SPI-Flash.

THANKS AND COURTESY


TRIVANDRUM

• Hugely helping in understanding sound, modulation, reconstruction and noise is the

application note by Audio Precision.

http://users.ece.utexas.edu/~bevans/courses/rtdsp/lectures/10_Data_Conversion/AP_Underst

anding_PDM_Digital_Audio.pdf

• Please go through this as it will be an eye opener.

• Audacity

• Wikipedia for all definitions [1] [2] and google image search for images ;)

http://users.ece.utexas.edu/~bevans/courses/rtdsp/lectures/10_Data_Conversion/AP_Understanding_PDM_Digital_Audio.pdf

Engineering

How to play audio from a microcontroller