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Nov 24, 2013 (3 years and 7 months ago)

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2012

Christopher Lewis & Sean Brooks

Digital System Processing


Project VI

12/11/2012

Audio Equalizer

1


TABLE OF CONTENTS



BACKGROUND INFORMATI
ON


3


SIGNAL ACQUISTION

3


FILTERS & SIGNAL PRO
CESSING

4


SIGNAL OUTPUT &
DISPLAY

6


CONCLUSION

7




















2




The current report will describe and provide an informative breakdown of an audio equalizer created to
adjustable settings per any user preferences using LabVIEW. The final project for Project V, Digital
Signal
Processing System Design, was to apply b
asic theory and applications of modern digital signal processing, to learn
basic theory of real
-
time digital signal processing, and to develop ability to implement and simulate digital signal
processing algori
thms using MATLAB and on real
-
time DSP platform.
Within the project, we must demonstrate
that these course objectives have been learned:




Understand basic concepts of digital signal proce
ssing theories and techniques.



Develop basic understanding of real
-
t
ime digital signal processing.



Develop ability to implement digital signal processing alg
orithms in Matlab and
LabVIEW
.



Develop ability to implement digital signal processing algorithms on real
-
time DSP platform.

Background Information

An audio equalizer
is a device
commonly used in sound recording and reproduction to alter the frequency
response of an audio system using linear filters.
1

Equalization is can be broken into three parts to better
understand the process. The three parts are signal
acquisition

(input), signal processing, and the output of the
modified signal.
F
ilters are the main component to the equalization process.
These filters can be used for noise
suppression, signal enhancement
, removal or atten
uation of a specific frequ
en
cy.
The signal p
rocessing section of
the filters can be .
Audi
o equalizers are typically constructed

in a parallel
-
circuit manner, where the lowpass,
bandpass, and highpass

are connected in this configuration. The filters are set to sp
ecified frequencies based on
typical frequencies that can be heard by the average human ear.
Frequencies affected can be analyzed visually by
implementing a spectrum analyzer. This element allows the user to see and measure the effects as the controls are
moved to a particular preference.
Such equipment to replicate these specific frequencies can be relatively
expensive. This can be corrected by using software
, such as LabVIEW.

This can minimize any size restrictions and
makes it portable for easy transport
.


Signal Acquisition

The design of the audio equalizer allows a user to connect audio files via the 3.5 mm jack. This was chosen to allow
compatibility with most audio input devices. The myDAQ is used to acquire the input signal. The myDAQ

is a low
-
cost
data acquisition (DAQ) device that gives students the ability to measure and analyze live signals. National
Instruments myDAQ is compact and portable so students can extend hands
-
on learning outside of the lab
environment using industry
-
standard tools and
methods
.
2

Once the myDAQ is connected to LabVIEW, we had to
program the correct to allow the audio to feed into our project’s VI. We had to set the sample rate and voltage
specifications to allow the audio files to play without lag or error. For example, t
he voltage specifications must be
set to ±2V for the left and right channel for the input and output. I believe this is just a standard parameter for the
myDAQ because once the voltage parameters are exceed, an error dialog box will display and stop the VI

from
continuing.
In the diagrams below, we will show how the signal acquisition part of the VI looks:




1

Wikipedia

2

http://sine.ni.com/np/app/main/p/ap/academic/lang/en/pg/1/sn/n17:academic,n21:16781/fmid/6353/

3



Filters and Signal Processing

Three filters are cre
ated based on common usage in the audio market. These filters are the bass, mid, and treble.
Each filter is set to attenuate a specific frequency to enhance the signal to a user preference. The bass filter is
called a lowpass filter that removes
frequencies

above the 512 hertz. In this particular project, we used a
Butterworth

IIR filter. IIR filter stands for Infinite Impulse Response

which is used when filters are needed for
“continuous
-
time” processing.
T
he second filter is for the mid
-
tones
. A

bandpass is created to handle a range of
frequencies, instead
of having one cutoff frequency. The mid
-
tones are primarily utilized to enhance vocals and
certain intrinsic sounds that are not typically in the lower frequencies. These frequencies range betw
een 513 and
5000 hertz. Lastly, the treble is created to enhance the higher frequencies above 5k hertz. Each filter is connected
to a control element that allows user to control the amount of output. These control elements are multiplied to
have a large mu
ltiplied that is noticeable in the audio files while being played. After the each filter is altered, then
each adjustment is added together to form the final output that is heard though the left and right channels. An
additional feature that is apparent on

the front panel is the left and right balance. This allows the user to adjust the
output to play more to one side
. For example, if one is seating closer to the left speaker, he or she can shift the
toggle to have more audio to the left verses the right.


Settings are adjusted to
accommodate audio from the
3.5mm jack of the left and right
channels.

4



Filter Configurations in LabVIEW


Each filter is calculated separately and then summed
together to form the final output that is heard through
the speakers with the desired adjustments from the
user.

5



Lowpass Filter: Bass (Left),

Bandpass Filter
: Mid
-
tone

(Right)
, & Highpass Filter: Treble (Below)

Signal Output and Display

The signal output is displayed in three different methods, Sound, Spectral Analysis and Frequency Bands. The
myDAQ used again to play the altered audio signal t
hrough generating output signals for the left and right channel
speakers. The best way to enjoy music is through sound.

The settings to generate an output signal with the
myDAQ are
very similar to the input settings to acquire the signal. As you can see ab
ove in the block diagram, the
final blue data line is connected to myDAQ and the spectral measurement element. This allows one to visualize the
data as the sound is being played. The s
pectral measurement tool
performs FFT
-
based spectral measurements,
such
as the averaged magnitude spectrum, power spectrum, and phase spectrum on a signal. Thirdly, a frequency
band was created

to

visually display

the varying frequencies from the audio files. In order to develop the bands,
we have to covert the input data into

a sine wave. The sine wave is the scaled down into an array which called be
later converted into decibel units. The varying decibels from the audio files are grouped into 5 different arrays
depending of the frequency level. This is displayed on the front
panel as five columns consisting of Boolean
indicators. As each frequency range increases or decreases, the indicators match the variance from green, yellow,
and red.

6



Front Panel Display

Week
-
to
-
Week Task List

Week

Task(s)

1

Gather ideas and complete research about audio equalizers. Brainstorm to
determine best method to implement the researched data into LabVIEW.
Develop outline for report and power point presentation.

2

Build project within LabVIEW.

3

Troubleshot any prob
lems encountered during the build. Include
screenshots and begin typing the report for the project.

4

Practice for the presentation, possibly with notecards. Check that objectives
and goals set have been achieved. Make corrections related to format or
usa
ge in the report and presentation.


Conclusion

The audio equalizer is common in different real world applications. LabVIEW can be used to reduce cost provide
the same effects as you higher end audio equipment. Challenges did occur throughout the process.
Many hours
were dedicated to research to understand the elements required to create a frequency band within the software.
Also, we had difficulty developing the correct calculations to have the frequency vary as the audio files were
playing. The filters an
d controls for the filters had to be constantly changed to accommodate the small voltage
output of ±2V.
Overall, this was an interesting because it required us to use many engineering design standards
and learn the extensive elements within LabVIEW to achi
eve the completion of our final project.

7


References

I.

Welch, Thad, Wright, Cameron, Morrow, Michael.
Real
-
Time Digital Signal Processing from MATLAB
to C with the TMS3206x DSPs
. Boca Raton,FL: CRC Press.2012