Department of Electrical & Computer Engineering Technology (ECET)

Division of Engineering, Computer Programming, and Technology (ECPT)

EET 4950
Senior Design Project

Three-in-One Audio Signal

Processing Unit

Submitted by

Dustin Pegalee and Dustin Funk

Supervised by

Dr. Ejaz

November 26, 2023

 Abstract
In a live-sound environment, mobility is a key component when selecting equipment. For
professionals like wedding deejays or corporate event producers, there may be a need for
audio processing components to provide a quality experience when delivering audio via
microphone. A device that can provide a specific combination of these components in a
package that can fit into a bag along with other equipment could help to reduce the
overall weight and size of the total rig in need of transport and setup. This project aims to
provide three such needs in a single small-form factor package, with the warmth of
analog signal processing.
D2A’s “3IO-ASPU” (Three-In-One Audio Signal Processing Unit) is an audio
processing equipment module that is geared towards microphone use in live
environments. Its compact form fits in one hand, is easy to grab from a box or a bag, plug
in an XLR cable and power, and it is good to go. This will keep the equipment footprint
down when producing events.
The unit includes a microphone preamp, a filter module comprised of a selectable
high-pass and fixed low pass filter, an audio dynamic range compressor, and a parametric
equalizer. Analog circuits inside the unit amplify the low voltage supplied by the
microphone transducer, filter out unwanted frequencies, automatically reduce the
amplitude of any audio that is louder than a specific threshold set by the user, and finally
boost or cut a selectable band of frequencies specified by the user.

ii
 Acknowledgements
The authors would like to acknowledge Dr. Ejaz for his contribution and support with this
project with both technical guidance for the design and general project oversight and
feedback. His encouragement has been invaluable in ensuring a novel and sensible design
that challenges us and provides experience with concepts beyond our education level so
far.
We’d also like to acknowledge the soldering knowledge and PCB knowledge
from one of the lab assistants, Sabrina Hurst. In addition to that, we’d like to thank all the
lab techs and managers as well. Without her knowledge of these important concepts, we
wouldn’t have completed the troubleshooting or whole design in an organized and proper
manner.
Next, we’d like to thank and acknowledge the project evaluators that took their
time out of their busy and hectic schedules to come and provide their immeasurable
feedback.
Finally, and most importantly, we’d like to acknowledge Professor Raul Valery
and Professor Michael Young for their extensive knowledge and assistance in making this
project come to life. Their feedback and assistance with aspects of each module really
made this project evolve.

iii
 Table of Contents

Abstract .............................................................................................................................. 𝒊𝒊
Acknowledgements ..........................................................................................................𝒊𝒊𝒊
List of Figures ................................................................................................................... 𝒗𝒊
List of Tables ................................................................................................................. 𝒗𝒊𝒊𝒊
Chapter 1 Introduction...................................................................................................... 1
1.1 Introduction and Motivation....................................................................................... 2
1.2 Our Product ................................................................................................................ 3
1.3 Overview of the System ............................................................................................. 4
1.3.1 Engineering Requirements .................................................................................. 5
1.3.2 Block Diagram/Flow Chart ................................................................................. 5
1.3.3 Engineering Specifications .................................................................................. 8
1.4 Limitations ................................................................................................................. 9
1.5 Similar Products ......................................................................................................... 9
Chapter 2 Background Research ................................................................................... 10
2.1 Basic Audio Signal Processing Concepts ................................................................. 10
2.1.1 Preamplification ................................................................................................ 10
2.1.2 Filitration/Equalization ...................................................................................... 12
2.1.3 Compression ...................................................................................................... 13
2.2 Filter and Compression Techniques ........................................................................ 14
2.2.1 Filter Types ........................................................................................................ 14
2.2.2 Compresor Types............................................................................................... 17
2.3 Hardware of the Project ........................................................................................... 22
2.3.1 The OPAx134 Operational Amplifier ............................................................... 23
2.3.2 LM3915 LED Driver Chip ................................................................................ 23
2.3.3 XLR Connectors ................................................................................................ 25
2.3.4 Power Module ................................................................................................... 26
2.4 Power Budget .......................................................................................................... 27
Chapter 3 Contribution ................................................................................................... 28
3.1 Assembly and Contribution ...................................................................................... 29
3.2 Verification of Each Module .................................................................................... 29
3.2.1 Preamp Module ................................................................................................. 29
3.2.2 Filter Module ..................................................................................................... 31
3.2.3 Compressor Module .......................................................................................... 32
3.2.4 Parametric Equalizer Module ............................................................................ 35
3.3 Full Circuit Verification ........................................................................................... 39
3.4 PCB Design .............................................................................................................. 47
3.4.1 Soldering Work ................................................................................................. 48

iv
Chapter 4 Non-Technical Issues ..................................................................................... 50
4.1 Budget ...................................................................................................................... 51
4.1.1 Final Project Budget .......................................................................................... 51
4.2 Timeline ................................................................................................................... 53
4.3 Environmental Aspects ............................................................................................ 55
4.4 Health and Safety ..................................................................................................... 55
4.5 Ethical Aspects ......................................................................................................... 55
4.6 Social Aspects .......................................................................................................... 56
4.7 Sustainability ............................................................................................................ 56

Chapter 5 Conclusion ...................................................................................................... 57

5.1 Summary and Conclusion ........................................................................................ 58
5.2 Suggestions for Future Work ................................................................................... 58

References ......................................................................................................................... 60
Appendix A: Normalized Butterworth Filter Design ................................................... 62
Appendix B: Noninverting Amplifier Gain ................................................................... 68
Appendix C: Datasheet of Components ......................................................................... 70

v
 List of Figures
Figure 1.1 3D Render of Device ...................................................................................4
Figure 1.2 Final Block Diagram of 3IO-ASPU.............................................................6
Figure 1.3 Final Flowchart of 3IO-APU .......................................................................7
Figure 2.1 ART Pro Audio MP Studio V3 Preamp .....................................................11
Figure 2.2 Regions of the Different Filters .................................................................12
Figure 2.3 8 Band EQ by W2IHY ...............................................................................13
Figure 2.4 Sebatron SMAC Compressor .....................................................................14
Figure 2.5 Butterworth Filter.......................................................................................15
Figure 2.6 Bessel Filter ...............................................................................................15
Figure 2.7 Chebyshev Filter ........................................................................................16
Figure 2.8 Elliptic Filter ..............................................................................................17
Figure 2.9 General compressor operation ...................................................................18
Figure 2.10 Compressor Attack, release, and Threshold ..............................................18
Figure 2.11 SSL G-Master Buss Compressor ...............................................................19
Figure 2.12 Universal Audio 1176 Limiting Amplifier ................................................20
Figure 2.13 Teletronix LA-2A ......................................................................................20
Figure 2.14 Fairchild 670 Compressor ..........................................................................21
Figure 2.15 Pye Compressor .........................................................................................21
Figure 2.16 Peak detector circuit of compressor module ..............................................22
Figure 2.17 OPA4134 Pinout ........................................................................................23
Figure 2.18 LM3915 LED Driver Circuit .....................................................................24
Figure 2.19 10 Segment LED bar graph........................................................................24
Figure 2.20 a) Balanced Input Circuit, b) Balanced Output Circuit..............................25
Figure 2.21 XLR Wiring Diagram and Pin Location ....................................................26
Figure 2.22 Jameco Dual Power Supply Schematic......................................................26
Figure 3.1 AC Sweep Plot of 30 Hz HPF....................................................................30
Figure 3.2 Input and Output of Preamp .......................................................................30
Figure 3.3 AC Sweep of Filters ...................................................................................32
Figure 3.4 Compressor Circuit ....................................................................................32
Figure 3.5 Failed Compressor Output Signal Measured Using Oscilloscope .............33
Figure 3.6 Oscilloscope measurement window. Yellow: input. Blue: Output ............33

vi
Figure 3.7 Oscilloscope measurement window. Yellow: input. Blue: Output ............34
Figure 3.8 Compressor Output of Simulated Audio Input ..........................................34
Figure 3.9 First Attempt of Parametric EQ: (a)Circuit (b)Output ...............................35
Figure 3.10 Oscilloscope measurement window. Yellow: input. Blue: Output ............36
Figure 3.11 AC Sweep of Parametric EQ at (a)10%, (b)50%, (c) 100% ......................37
Figure 3.12 Continuity Between All Modules (a)Circuit, (b)Output ............................38
Figure 3.13 PCB measurement with oscilloscope .........................................................39
Figure 3.14 Oscilloscope measurement window Yellow: input. Blue: Output .............39
Figure 3.15 Oscilloscope measurements of full circuit. Yellow: input. Blue: output ...40
Figure 3.16 70Hz Option Verification ..........................................................................41
Figure 3.17 Oscilloscope window with channel 1: Input and channel 2: Output .........41
Figure 3.18 100 Hz Option Verification .......................................................................42
Figure 3.19 AC Sweeps for Parametric EQ Verification (a)10%, (b)50%, (c)100%....43
Figure 3.20 AC sweep chart showing Parametric EQ boost/cut potentiometer at
minimum, frequency sweep potentiometer at minimum ...................................................44
Figure 3.21 Yellow: Audio input signal. Blue: Compressed output signal ...................45
Figure 3.22 Oscilloscope measurement window. In yellow: Audio input signal. In blue:
Capacitor voltage ...............................................................................................................45
Figure 3.23 VU meter operational verification at (a)10%, (b)50%, (c) 100% ..............47
Figure 3.24 3D Render of PCB (a)top view, (b) side view, (c)No components ...........48
Figure 3.25 (a)Soldering Station and (b)Soldering Progress ........................................49
Figure 5.1 Power Phantom Circuit ..............................................................................59

vii
 List of Tables

Table 1.1 Final Engineering Requirements..................................................................5

Table 1.2 Final Engineering Specifications .................................................................8
Table 1.3 Similar Products ...........................................................................................9
Table 2.1 Proposed Power Budget .............................................................................27
Table 2.1 Proposed Power Budget .............................................................................27
Table 2.2 Final Power Budget....................................................................................27
Table 4.1 Proposed Project Budget ............................................................................51
Table 4.2 Final Project Budget ..................................................................................52
Table 4.3 Proposed Timeline Gantt Chart .................................................................53
Table 4.4 Detailed Proposed Timeline .......................................................................53
Table 4.5 Project Timeline Gantt Chart .....................................................................54
Table 4.6 Project Final detailed timeline by week .....................................................54

viii
ix
Chapter 1
Introduction

Summary
This chapter will introduce D2A’s 3IO ASPU functions and objectives. We will
address the problems faced with mobile DJs and music producers and how to
solve it. Additionally, we will explain the motivation to design this project.
Finally, our project requirements, specifications, block diagram and flow
chart, as well as the limitations of our product will be discussed. Our product
will be compared with similar products on the market to show its uniqueness.

1.1 Introduction and Motivation

1.2 Our Project
1.3 Overview of the System
1.4 Limitations
1.5 Similar Products

-1-
1.1. Introduction and Motivation
In the modern era of mobile DJs and musicians, there are various gigs, events, and festivals to
go to. They are not just worried about the music, but flights, hotels, rental space, and the most
important thing, gear. Some examples of the gear that they take are their laptops, cables,
instruments, DJ controllers, CD players, etc. Most importantly their microphones and the
things to make them sound good. This includes microphone preamps, filters and equalizers,
and compressors. Most of these devices are very bulky and are a necessity to make them
sound good and sounding good is the only thing the audience cares about. In the
musicians’/DJs’ lives, sounding good can be the difference between a million-dollar contract
or living in the back of your car for all your life.

Both group members have been in the music industry as a hobby; Dustin F has worked
in the live audio industry and is a wedding DJ, and Dustin P has worked in audio recording
studios and a bedroom music producer. For a typical DJ set up, we have our microphone
connected to a DJ mixer, and the DJ mixer would then connect to a power outlet. The mixer
audio output would connect to some speakers. With the DJ setup, there are no independent
controls on the mixer to control the dynamic range. The equalizer section is very minimal and
vague, and the only thing you can do is to control the volume and amplify the sound. If we
wanted to use compression or equalization, we would have to connect an external device or use
a software on our laptops to do such processing and requires more equipment.

As a bedroom music producer, one would plug a microphone into an audio interface
and if audio processing is required, one would have to go into the digital audio workstation
(DAW), navigate to the processing unit, and use a mouse to fine tune the parameter. Since the
effects are digital, one would have to save constantly to save knob settings. If the power goes
out, settings can be lost. One would have to load the program and change all the settings.
Furthermore, every instrument is different, so remembering each setting would be difficult.

That is where the motivation for this project came from. For mobile DJs, convenient
and on-the-fly signal processing without having to carry bulky external processing devices or
going through software on your laptop are important. For music producers, the most important
thing is workflow and convenience of having set parameters for every instrument without
having to fiddle with the software. As music producers, we can keep our focus on the sound
rather than the hardware itself or remembering settings.

-2-
 This device is intended to fill a potential need where the three functions described
(preamp, filtration, and compression) are better provided in a small form factor, handheld, and
are easily accessed and used. The device has a single input, a single output, and a power switch.
Typical audio compressors can be bulky, or not designed as on-the-go sound processing units
when considered for vocal productions or live events. This unit is intended to allow a user to
shape the sound of their vocal output and send this audio to external speakers or other
equipment.

1.2. Our Product

The 3IO-ASPU is a “Three-In-One Audio-signal Processing Unit” that processes audio input
from a microphone, and outputs the processed audio to another device, such as speakers or a
recording device. The unit is intended to be a compact device, easy to transport and store, and
simple to use. Four features of this device are a pre-amplifier, which boosts the low-level
microphone voltage at the input up to line level, a filtration module (LPF and HPF), for noise
reduction, an audio dynamic range compressor which reduces the amplitude of parts of the
incoming sound based on a threshold determined by a user, and a parametric equalizer, which
allows a user to define characteristics of the output sound.
The device consists of a single XLR input, a large gain adjustment knob for the pre-
amp module, selectable low-pass and high-pass frequencies, adjustment knobs for the
parametric EQ boost/cut, and center frequency, as well as knobs to adjust the attack, release,
threshold, and ratio of the audio compressor module. The back of the device will have a male
XLR output connection and a power connector. Ideally, the audio fidelity of the device will
be improved with the use of low-noise op-amps designed for audio use. The circuit is all-
analog as well, which should provide warmth to the sound that digital signal processing
typically does not. A 3D rendering of the product is shown in Figure 1.1. The actual product
will be discussed and shown in Chapter 3 of this report.

-3-
 Figure 1.1: 3D Render of Device

1.3. Overview of the System

The 3IO ASPU is a compact audio signal processing device that provides a simple easy-to-use
way to compress, amplify, and filter incoming audio signal from a microphone, and output
processed audio. The device also shows amplifier gain using a visual meter consisting of LED
lights. There are adjustable parameters for the pre-amp gain, compression ratio, compression
threshold, filter gain boost/cut, and filter centre frequency.
The 3IO ASPU is designed to be economical in terms of space and ease of use for the
end user. The unit has a female XLR input on one side, common for microphone connections,
and male XLR output on the other side. The output is a processed version of the audio signal
provided at the input.
The device accepts a mic-level voltage at its input, passes the signal through an
amplifier circuit with adjustable gain level, from which the output of that module provides
audio input to the filter module. The filter module has two selectable HPF filters (selected by
a toggle switch) and is then sent into a fixed LPF. The output of the filter module is connected
to the input of the compressor module. This module consists of a voltage-controlled amplifier,
or VCA, a peak detector, and a voltage subtractor circuit. Finally, the last module boosts or
cuts the gain of the audio signal along a curve defined by a selectable center frequency. The
final product will be shown in Chapter 3.

-4-
1.3.1. Engineering Requirements
There were a few iterations of the engineering requirements from the proposal phase to the
final project through discussions from the advisor meetings and further research. The following
table is our final design engineering requirements divided into high, medium, and low. Per the
request from the project evaluators, we added requirements that were quantifiable and
measurable to keep track of the progress of our project.

Table 1.1 – Final Engineering Requirements

Priority “The device shall have Verification
…”
Audio preamplification Input and Output
with a max of 50dB with waveforms on
a 30 Hz HPF oscilloscope
Noise-filtering low-
pass/high-pass filter with AC Sweep using
fixed 20kHz LPF and Spectrum Analyzer
selectable 70Hz/100Hz
HPF
Dynamic range
compression with a Real-time signal analysis
threshold up to 6V, ratio on oscilloscope
High from 1:1 to inf, and
attack time from 3ms to
25ms; release time from
95ms to 180ms.
Sound-shaping bandpass
filter with a ± 12dB AC Sweep using
adjustable gain with Spectrum Analyzer
center frequencies from
100 Hz to 1k Hz.
Medium Input and Output gain Visual inspection
VU meters
Low Compact footprint Can be held in one hand
and fit into a bag

1.3.2. Block Diagram/Flow Chart

Similar to the engineering requirements, the block diagram went through some changes. Since
there are active elements in the circuit, biasing voltages are required. This was reflected in the
modified block diagram.

-5-
 Figure 1.2: Final Block Diagram of 3IO-ASPU

The block diagram shows that the XLR audio input feeds the input signal to the pre-
amp module. This module includes adjustable gain of up to 50dB as well as a high-pass filter
to remove any low-level noise. The pre-amp output is then fed to the filter circuit, which has a
12dB boost/cut at an adjustable frequency. An additional high-pass or low-pass filter option is
available for users to activate in the case of noise above or below the cutoff frequencies. The
output of the filter module is fed to the compressor input module. This module is composed of
three sub-modules consisting of a voltage-controlled amplifier, a peak detector, and a voltage
subtractor. The output of the voltage subtractor subcircuit will feed the input of the VCA
triggering the gain reduction in a loop. The speed at which the VCA is triggered is called the
attack and the speed at which it returns to pass through is called release. The voltage threshold
of the compressor for activation is adjustable via potentiometer, as well as the compression
ratio itself. The flowchart shown in Figure 1.3 describes the signal flow of the device.

-6-
 Figure 1.3: Final Flowchart of 3IO-APU

The flow chart went through many iterations. Our signal processing unit has subjective
points in the system, but similar to the engineering requirements we tried to define
quantitative and measurable aspects of the device in our flowchart.

-7-
1.3.3. Engineering Specifications
The following table shows the engineering specifications for each module and their
corresponding components. The table includes the components used and what each module is
supposed to do.

Table 1.2 – Final Engineering Specifications

Block Type Components Engineering Justification Responsibility

Specifications

Power Module 12V Dual Power Converts This is the

Supply 120VAC to power source for
12VDC the circuit, Both
including
biasing for the
op-amps.

Preamplifier Op-amp, 70dB maximum Amplify low-

Module capacitors, amplification, level
resistors 30Hz HPF microphone
voltage to line
level Dustin P.

Filter Module Op-amps, SPDT LPF cutoff: Fixed LPF

switch, 20kHz cutoff at 20kHz
capacitors, and and selective
resistors HPF cutoffs: HPF cutoffs at
70Hz and 100Hz 70Hz and 100Hz Dustin P.

Compressor VCA, peak 6V maximum Audio

Module detector, threshold, op- compression
comparator amp based with adjustable Dustin F.
parameters

EQ Module 10k pot, 10k dual 12dB boost/cut Adjustable

gang pot, 3 op- amplifier gain filtering, boost
amps or cut Dustin F.

Display 10-segment LED 3 dB/step analog Audio gain

Module bar graph, audio display representation
driver chip for end user Dustin P.

-8-
1.4. Limitations
There are a couple of limitations with our project. The main limitation of our project is that it
is designed for vocals only. Another limitation of our project that it is designed for dynamic
microphones, meaning, that there is no phantom power built in for condenser microphones
(more will be explained in the next chapter). Next, the input of the compressor is designed to
take in a signal around 5V. Furthermore, there isn’t a makeup gain knob, we have a fixed gain
at the end of the compressor instead of a gain knob.

1.5. Similar Products

There are similar products on the market, however, not exactly similar to what we have built.
Most of the products in the market are missing features or not as fully featured as ours. Also,
most of them require software to unlock all the features that are not present on the device itself.
Table 1.3 shows two products that are compared against our product for a variety of
specifications.

Table 1.3 – Similar Products

-9-
Chapter 2
Background Research

Summary
In this chapter we present basic audio concepts, as well as different types of
filters and compressors. Rationale for the choice of filter topology and
compressor type will be discussed as well. Power budget is also discussed,
specifically the proposed power budget versus the actual power budget.

2.1 Basic Audio Signal Processing Concepts

2.2 Filter and Compression Techniques
2.3 Hardware of the Project
2.4 Power Module

- 10 -
2.1 Basic Audio Signal Processing Concepts
Sounding good is not as simple as plugging a microphone up to a speaker. There are three main
stages that an audio signal goes through to make a sound studio-quality: preamplification,
filtration/equalization, and compression.

2.1.1. Preamplification
Amplification is the process of increasing the voltage, current, or power level of a signal [1];
the device that does this is called an amplifier. The reason this process is called preamplification
is because the voltage level for a microphone is extremely low and it would be inaudible to use
or hear. The mic signal needs to be amplified to the line level before any processing can be
done. An example of a preamp is shown below:

Figure 2.1: ART Pro Audio MP Studio V3 Preamp [2]

The basic operation of the preamp is to increase the gain of low-voltage microphone
input up to the line-level voltage. This preamp has a tone control for different tones to create
a specific sound profile. Our device does not have a tone control. There is one gain knob and a
VU meter to track the incoming signal.

- 11 -
2.1.2. Filtration/Equalization
Filtration is the process of cutting out unwanted frequencies from a signal. There are two types
of filters: passive and active filters. The difference between active and passive filters is that
active filters contain active components, such as op-amps, and they need external power to
operate. Passive filters, on the other hand, use a combination of resistors, capacitors, and
inductors to perform filtration and do not need an external power supply to function.

Every filter has at least three regions that define the frequency response of a filter. These
regions are as follows: the passband, transition band, and stopband. The passband is a range of
frequencies (with full amplitude response) that can “pass” through a filter. Next, the transition
band is a range of frequencies that allows a transition between the passband and stopband of a
filter. This band is defined the cutoff frequency of the filter. The stopband is a range of
frequencies where signal cannot pass through. Figure 2.2 shows examples of the passbands,
transition bands, and stopbands of each type of filter.

Figure 2.2: Regions of the Different Filters [13]

There are four types of filter classes: low-pass, high-pass, band-pass, and band-reject [1].
Low-pass filters let the frequencies below the cutoff frequency pass and block the frequencies
above that. High-pass filters cut the frequencies below the cutoff and let everything else pass.
Band-pass filters let a range of frequencies pass through, blocking frequencies above and below
- 12 -
the two cutoff frequencies. Band-reject filters are opposite to band-pass filters; they cut a range
of frequencies and let everything else pass.

The device shown in Figure 2.3 is an example of a filter, more specifically, an equalizer.
This device boosts and cuts different bands of frequencies. In our product, we have a low-pass
and high-pass filter and a single-band adjustable equalizer.

Figure 2.3: 8 Band EQ by W2IHY [3]

2.1.3. Compression
In audio, compression is the process of controlling the dynamic range of a signal, which is the
difference between the loudest part and the softest part of the signal [4]. Four common
parameters of compressors are: threshold, ratio, attack, and release. The threshold parameter
determines when compression of the input signal will start. As the amplitude of an audio input
signal goes past the threshold, the compressor begins to reduce the overall gain. The attack
parameter determines how fast it takes for a compressor to begin compressing the signal. The
release setting determines the speed at which a compressor returns the signal to its original
uncompressed state [4]. The ratio parameter determines how much the compressor will reduce
the gain of the signal overall, when triggered by the threshold comparison. [4].

- 13 -
 Figure 2.4: Sebatron SMAC Compressor [5]

2.2. Filter and Compression Techniques

There are different types of filter and compression topologies or prototypes. For filters the
different topologies are Butterworth, Bessel, Chebyshev, and Elliptical. For compression, there
are five types of compression: VCA, FET, optical, tube, and PWM. In this section, different
topologies will be discussed, in general, but emphasis will be given to the discussion of filter
and compressor that were chosen for our project.

2.2.1. Filter Types

Butterworth

As mentioned above, there are four main filter topologies: Butterworth, Bessel, Chebyshev,
and Elliptic filters. The first filter is called Butterworth and is named after British Physicist,
Stephen Butterworth [6]. The filter is commonly known as the “maximally flat” option with
its passband response being flat and steepest roll-off without a ripple in the passband [6]. The
selectivity, or width of passband, is better than any of the filters, however, with the greater
selectivity, it creates more delay and phase linearity [6].

- 14 -
 Figure 2.5: Butterworth Filter [6]

Bessel

The next type of filter is the Bessel filter named after German mathematician, Friedrich
Bessel [6]. Known for its gentle response, the Bessel filter doesn’t have a sharp cutoff, can
only be used in applications where there is space between the passband and stopband such as
pulse applications [6]. The disadvantage of Bessel filters is that it requires the most
components and have very specific applications, however, the advantages that it has very low
phase shift and its sensitivity to component tolerance [6].

Figure 2.6: Bessel Filter [6]

- 15 -
Chebyshev

The Chebyshev filter is known for its rippling response and steep roll-off [6]. It requires
the least components to operate and has one of the best roll-offs out of any filter. The ripples
and steepness of the of the roll-off are directly proportional to each other; the greater the
steepness, the larger the ripples in the passband. The phase response is non-linear and can
cause problems with distortion in demodulation [6].

Figure 2.7: Chebyshev Filter [6]

Elliptic

The final filter type is called the Elliptic Filter, or Cauer Filter named after German
mathematician Wilhelm Cauer [6]. This is filter is characterized for its ripples in the passband
and stopband; unlike the Chebyshev, its selectivity is greatly improved. Despite its great
selectivity, the filter network is more complex and requires more components. The ripples in
the passband and stopband are independently controlled depending on application [6].

- 16 -
 Figure 2.8: Elliptic Filter [6]

Of all filter types, the filter chosen for this project was the Butterworth filter. Its flat
response type and steep roll-off provide a good choice for audio applications, as well as our
familiarity to the filter type from studies in the BSECET program. The goal of the filter module
is to reduce noise in the audio signal and block unwanted frequencies as well. Another reason
why we used the Butterworth filter is because it’s the most versatile and practical in all of the
applications we’ve studied in this program.

2.2.2. Compressor Types

Like filters, there are different types of audio compressors that use different techniques to
accomplish real-time and on-the-fly gain reduction of the audio signal. Compressors typically
operate as gain control for an audio signal, decreasing the signal gain when the average input
amplitude goes above a set threshold amplitude. This concept is shown in the figure below.

- 17 -
 Figure 2.9 General compressor operation. [16]

Typical characteristics of audio compressor operation includes attack, decay,

threshold, and ratio. Threshold is an amplitude level typically set by a potentiometer. This
level will be compared to the audio signal average peak amplitude to determine whether the
compressor operates or allows the signal to pass through. Attack is the speed at which the
compressor reaches full compression after the input signal passes the threshold. Release is the
speed at which the compressor operation returns to allowing the original signal to pass
through. Ratio determines the actual amount of gain modification that occurs. The figure
below shows the function of three of these characteristics in operation.

Figure 2.10 Compressor Attack, release, and Threshold. [4]

- 18 -
 The most common compression compressor type is known as a VCA compressor [7].
VCA stands for voltage-controlled amplifier. VCA compressors are known for their precise
control and predictable sound, and their style of compression is often emulated and perfected
by many engineers with precise parameters [7].

Figure 2.11: SSL G-Master Buss Compressor [7]

The second type of compressor are FET compressors, or Field Effect Transistor
compressors, they rely on transistors for gain reduction. They are known for their speed,
allowing for almost instant adjustments in attack and release times. However, this rapid
response can add some distortion and color, especially when pushed hard. While they may not
be the best choice for clean mix bus compression, they shine on individual instruments like
vocals, guitars, kick drums, and snares [7]. The iconic 1176 Limiting Amplifier is the most
famous FET compressor, prized for its ability to add a bright, present, and gritty character to
audio signals, especially in the "all buttons in" mode for intense parallel compression [7].

- 19 -
 Figure 2.12: Universal Audio 1176 Limiting Amplifier [7]

Next, optical compressors, often called 'opto,' use a light-dependent resistor and a light
source to control compression and gain reduction. The input signal activates the light source
inside the compressor, adjusting its brightness based on the input level. The light-sensitive
resistor then regulates compression [7]. Opto compressors are valued for their 'musical' and
'smooth' sound, offering a contrast to the quick and aggressive FET compressors or the clean
and punchy VCAs. A common vocal processing chain combines a FET compressor to capture
sharp transients and an opto compressor to 'glue' everything together. The Teletronix LA-2A
is an optical compressor, with simple controls and program-dependent attack and release times
based on input strength [7].

Figure 2.13: Teletronix LA-2A [7]

The fourth type of compressor are tube compressors, using vacuum tubes for gain
reduction, have a slower response to transients, making them suitable for instruments and mix
buses [7]. They don't compromise the vitality of mixes because their slower attack times

- 20 -
preserve important transients [7]. The Fairchild 670 is a well-known hardware tube
compressor.

Figure 2.14: Fairchild 670 Compressor [7]

The final compressor type is Pulse Width Modulation, or PWM, compressors, often
overlooked, are as powerful as other types. They use high-frequency pulse signals to control
signal amplitude over time by dividing it into discrete parts and muting sections. PWM
compressors operate at an incredibly fast pulse rate, avoiding artifacts or audible on/off
stuttering effects while offering transparency and lightning-quick attack and release times [7].
The Pye compressor from the 1960s is the iconic example of a PWM compressor.

Figure 2.15: Pye Compressor [7]

With several different compressor types to choose from, a VCA-based compressor was
chosen for this project due to familiarity with op-amp operation, consistency of components

- 21 -
such as the OPA4134 IC, and availability to informational material in reference to the operation
of components of VCA compressor operation.

The compressor module of this project consists of three internal subcircuit modules
operating in a feedback loop. A volume modulator controls the gain of the signal, a peak
detector determines the maximum amplitude of a given audio signal, and a voltage subtractor
compares the peak detector output to the threshold determined by the potentiometer. To verify
the attack and release characteristics of the compressor, the charge and discharge times of the
capacitor located at the output of the peak detector module must be measured.

The figure below shows the capacitor C16 that is measured to verify compressor attack
and release times.

Figure 2.16 Peak detector circuit of compressor module.

2.3. Hardware of the Project

The following sections describe the specific hardware that are used to satisfy the requirements
of this project. Since our project is purely analog, in this section we will focus on the specific
chips and op-amps that were used. Additionally, the power supply for the entire circuit will be
discussed.

- 22 -
2.3.1. The OPAx134 Audio Operational Amplifier
The primary operational component of our project is the OPAx134 Audio Operational
Amplifier. The OPAx134 comes in single, dual, and quad versions of the chip. For this project,
the 4-channel model OPA4134 IC is used to reduce the overall footprint of active components
in the circuit. The OPA4134 features ultra distortion at 0.00008%, high speed with a slew rate
at 20V/µs, a bandwidth of 8MHz, and has a wide supply range with ±2.5V to ±18V [8].

Some applications where this device are useful are professional audio and music, line
drivers, line receivers, multimedia audio, active filters, preamplifiers, integrators, and
crossover networks [8]. With all our fundamental circuits for each module, this chip will offer
more than enough power and capabilities for our project. Also, the dual and quad versions of
the chip have completely independent circuitry reducing crosstalk and freedom from
interaction even when chip is overloaded [8].

Figure 2.17: OPA4134 Pinout [8]

2.3.2. LM3915 LED Driver Chip

The LM3915 is a versatile integrated circuit that senses analog voltage levels and can drive up
to ten LEDs, LCDs, or vacuum fluorescent displays. It offers a logarithmic 3 dB/step analog
display. With just one pin change, you can switch between a bar graph and a moving dot
display. The LED current is regulated and can be programmed, removing the need for current
limiting resistors. This display system can function with a single supply voltage ranging from
3V to 25V [9].

The 3 dB/step display of the LM3915 IC is ideal for signals with wide dynamic ranges,
like audio level, power, light intensity, and vibration. In audio applications, it can be used for
average or peak level indicators, power meters, and RF signal strength meters [9]. In our case,
we will use a 10-LED bar graph to track the amplitude of the audio signal. The LEDs will
fluctuate with changing amplitude from audio signal. The LM3915 can be run in 2 modes: dot

- 23 -
mode (or single LED mode) or bar mode. We will be running it in bar mode. The following
is the circuit we will be making and is in the data sheet of the LM3915.

Figure 2.18: LM3915 LED Driver Circuit [9]

Figure 2.19: 10 Segment LED bar graph. [9]

- 24 -
2.3.3. XLR Connectors
XLR connectors, widely employed in professional audio and video equipment, boast a robust
design and balanced signal transmission capabilities. The term "XLR" stands for X-series,
Latch, Rubber. With three to five pins on each connector, including a locking mechanism for
secure connections, XLR connectors are found in microphones, amplifiers, and mixing
consoles within the professional audio-visual (AV) industry.
When talking about XLR connections, configuration comes in two forms: balanced and
unbalanced. Balanced XLR connections have three terminals: positive, negative, and ground
while the unbalanced connection only has a positive and negative terminal. Balanced
connections help reduce interference and noise while unbalanced connections are prone to
noise. In our project, we will be using balanced XLR inputs/outputs. The following figures
show the circuits for the conventional input/output.

(a)

(b)

Figure 2.20: a) Balanced Input Circuit, b) Balanced Output Circuit [10]

Similar to many connectors like power plugs, XLR connectors come as male and
female. Where you want to send a signal is dependent on the type of connector. Typically,
female connectors accept incoming signals while male connectors send the signal out. The
- 25 -
pins or wires of an XLR cable are color-coded to designate the proper connections. Pin 1 is
designated for ground (black wire), pin 2 is positive (red wire), and pin 3 is negative (white or
black). The following figures show the pin locations and designation between male and female.

Figure 2.21: XLR Wiring Diagram and Pin Location [11]

2.3.4. Power Module

To supply power to all op-amps, we are using the Jameco 20626 (JE215) Adjustable Dual
Power Supply Kit [12]. This dual power supply features independently adjustable positive and
negative output voltages, which would bias all operational amplifier IC to the appropriate
voltage. The voltage range of each terminal is from 1.2V to 15V DC. The output power of each
supply is as follows: 5V@500mA, 10V@750mA, 12V@500mA, and 15V@750mA. The
power supply has adjustable regulators with thermal overload protection and comes with a heat
sink to cool the regulators. All the circuitry and information were provided by the datasheet.
The following figure is the schematic of the power supply. Notice the two potentiometers, R3
and R4 that are used to change the voltage.

Figure 2.22: Jameco Dual Power Supply Schematic [12]

- 26 -
2.4. Power Budget
The power budget shows the overall output power our device consumes. All our voltages for
each op-amp are fixed; the only factor that would change our output power is the current and
the input audio signal. Table 2.1 shows the proposed total power budget for the project.

Table 2.1 – Proposed Power Budget

Input Bias Input Bias

Component Quantity Power (W)
Current Voltage

OPA4134
Quad- 5 100pA 12V 6nW
OpAmp

LM3915 IC 2 100nA 12V 2.4nW

LM3915
20 10mA 12V 2.4W
Driver

Total 2.4W

Since no major modifications were made to the project regarding power usage, the
actual power budget remains within the maximum operating conditions as calculated in the
proposed power budget of the project. The bias currents measured for the OPA4134 op-amps,
and LM3915 LED driver IC during operation were far below the maximum expected bias
currents, as well as the drivers for the LEDs of the VU meter module. The table below
represents the actual power budget of the practical devices of the project.

Table 2.2 – Final Power Budget

Max Input Input Bias

Component Quantity Power (W)
Bias Current Voltage
OPA4134
Quad- 5 100pA 12V 6nW
OpAmp
LM3915 IC 2 100nA 12V 2.4nW
LM3915
20 9.23mA 12V 2.21W
Driver
Total 2.21W

- 27 -
Chapter 3
Contribution

Summary
In this chapter, we present the process of building, testing, and troubleshooting of
our project. We will cover the PCB design and enclosure and verify our engineering
requirements and elaborate on our results.

3.1 Assembly and Contribution

3.2 Verification of Each Module
3.3 Full Circuit Verification
3.4 PCB Design

- 28 -
3.1 Assembly and Contribution
This project involves several analog circuit modules that perform functions indicated by the
engineering requirements of the project. The pre-amp module and filter modules were designed
and assembled by Dustin Pegalee, as well as the assembly of the XLR input circuit and VU
meter circuits. Dustin also researched PCB design for the project.
Dustin Funk assembled the compressor module, including the four modules within, and
the parametric EQ module. Since most modules are active circuits using biased op-amps, a dual
power supply is needed for the project. A kit was sourced and assembled to provide positive
and negative voltages to all modules.
We collaborated to combine all physical modules and ensure the audio signal flow from
the input to the output. We also coordinated on circuit connections, the prototype box
construction, and design.

3.2. Verification of Each Module

This section will focus on the verification and analysis of our circuits individually. After we
verify each module individually, we will then put all four circuits together and test everything
in conjunction with each other.

3.2.1. Preamp Module

The first requirement for the preamp module is a HPF of 30 Hz before the signal being
amplified. We changed the cutoff from 50 Hz to 30 Hz from proposal feedback and because
from further research, the rumble starts at 30 Hz and below; if we kept the 50 Hz, we’d lose
some resonance from the lower end.

When designing the 30 Hz HPF, we used a normalized unity gain Butterworth filter
design. Appendix A1 shows the step-by-step design procedures of the HPF. For verification,
we did an AC sweep through the oscilloscope and function generator. We started at a very low
frequency, recorded the voltage at that frequency, increased the frequency in even increments,
and plotted the voltage vs the frequency. The following figure was the plot of the results we
got from the AC sweep.

- 29 -
 Figure 3.1: AC Sweep Plot of 30 Hz HPF

The next requirement for the preamp is a gain of 70dB. We originally had the gain at 50dB,
but after the analysis of the audio coming the input of the microphone being really low, we
increased the gain of the preamp using Appendix B. The following figure shows the input and
output of the preamp after the change.

Figure 3.2: Input and Output of Preamp

Using the gain equation and decibel conversion in Appendix B, the gain of the preamp is at
60dB. Due to the noisy signal from the function generator and component noise, we were not
able to use a lower amplitude, however, we did get close to the desired value.
- 30 -
3.2.2. Filter Module
In total, there are three filters that were designed and tested. There are two HPFs (70Hz and
100Hz) and one LPF that should be around 18k Hz. The HPF’s were no issue in designing.
Where we had an issue was with the LPF. Somehow, depending on how much the
potentiometer was turned (how much gain there was in the preamp), the cut-off frequency
changed. Regardless, we ran an AC sweep of all three filters and the following are the results.

(a)

(b)
- 31 -
 (c)

Figure 3.3: AC Sweep of Filters

3.2.3. Compressor Module

The compressor module is constructed on the breadboard to test its functions and verify the
module is meeting engineering requirements. The circuit was assembled using a different brand
of 14 pin SOIC adapters than the ones we previously acquired. These required further soldering
to assemble, as well as soldering the IC chip to the adapter.

Figure 3.4: Compressor Circuit

After constructing the circuit, a test signal was sent to the input of the compressor. Using
the oscilloscope, the input and output waveforms are observed.

- 32 -
 Figure 3.5: Failed Compressor Output Signal Measured Using Oscilloscope

Unfortunately, the output signal was not a representation of the input signal. After
identifying a possible short on the adapter for the IC chip, a second signal measurement was
attempted:

Figure 3.6: Oscilloscope measurement window. Yellow: input. Blue: Output

Either there are further issues with the IC chip or adapter, or there are problems with
the potentiometers or diode string. The circuit will have to be reconstructed more cleanly with
thorough testing along the way.

After reconstructing the circuit, the expected output signal was observed, with slight
compression.

- 33 -
 Figure 3.7: Oscilloscope measurement window. Yellow: input. Blue: Output

The sine wave input from the function generator is set to 13.5Hz, to more closely
simulate the time scale of human speech, which the compressor operation is designed for. The
output waveform shows that when the voltage level goes above the threshold, the gain is
reduced. Changing the settings of the attack, decay, threshold, and ratio change the
characteristics of the output waveform as shown below:

Figure 3.8: Compressor Output of Simulated Audio Input

To observe the real-world operation of the compressor module, connection of the XLR
input module is required for an audio signal to pass through. The attack and release times of
the compressor can be determined by measuring the charge and discharge times of the capacitor
in the peak detector module.

- 34 -
3.2.4 Parametric Equalizer Module
After constructing the circuit on the breadboard, a preliminary analysis was performed using
the oscilloscope to see if there would be some type of output resembling the input, but the
output did not show the assumed signal that was present in the simulation analysis.

(a)

(b)

Figure 3.9: First Attempt of Parametric EQ: (a)Circuit (b)Output

A clean signal from the output of the reconstructed parametric EQ circuit after some
verification of individual components including potentiometers and all channels of the
OPA4134 IC chip. Through measurement, one of the dual gang potentiometers acquired for
this project showed a maximum resistance of 8.4kΩ. Another potentiometer was selected and

- 35 -
measured approximately 9.2kΩ. With both the boost/cut pot at 50% and the frequency sweep
pot at 50%, the output should be a direct representation of the input, as shown in the figure
below.

Figure 3.10: Oscilloscope measurement window. Yellow: input. Blue: Output

Time was then spent trying to identify the maximum center frequency of the frequency
sweep function by turning the boost pot to 100% and doing a manual AC sweep using the
function generator, looking for the approximate frequencies at which the maximum amplitude
begins to reduce, and finding the midpoint between those frequencies. The maximum
frequency appeared to be approximately 1kHz. The figures below are graphical representations
of three manual AC sweeps performed while the frequency potentiometer is set to 10%, 50%,
and 100% of its total range. The boost/cut potentiometer is set to maximum for each sweep to
show operation.

(a)
- 36 -
 (b)

(c)

Figure 3.11: AC Sweep of Parametric EQ at (a)10%, (b)50%, (c) 100%

Four modules of the project were combined in the lab to measure for continuity using an
oscilloscope. After some troubleshooting, the expected waveform at the output of the EQ
module was observed. The 1kHz 50mV peak waveform at the pre-amp input was amplified to
5V peak, set using the gain potentiometer, and successfully passed through each of the
modules, including preamp, high-pass filter, low-pass filter, compressor module, and
parametric equalizer module. The figures below show the connected modules, as well as the
oscilloscope reading showing continuity. The compressor module is set for zero compression
by setting the threshold above the maximum amplitude of the signal and reducing the ratio to
minimum. The parametric equalizer module is set to pass the signal through without any gain
increase or decrease.

- 37 -
 (a)

(b)

Figure 3.12: Continuity Between All Modules (a)Circuit, (b)Output

- 38 -
3.3. Full Circuit Verification
Once continuity is accomplished on the breadboard, and individual modules have been tested
to be functional, the PCB is soldered with the main circuit components required for testing
the engineering requirements. This includes the XLR input subcircuit, preamplifier module,
HPF circuits and switch, LPF, compressor module, and parametric equalizer module. The
PCB is then assembled with all potentiometers and input bias power supply. The figure below
shows the finished prototype.

Figure 3.13: PCB measurement with oscilloscope

The circuit is then tested for continuity using the XLR input with a Shure SM58
Dynamic microphone and XLR cable. The output of the parametric EQ circuit is probed, as
well as the input to the preamp module. The figure below shows the observed oscilloscope
window.

Figure 3.14: Oscilloscope measurement window Yellow: input. Blue: Output

- 39 -
 The input shows a maximum voltage of 34mV. This is typical of microphone
transducer outputs and is the reason that preamplifiers are necessary for most applications
involving these types of microphones. At the output of the parametric EQ circuit, the
observed voltage is a maximum of 3.44V. The parametric EQ module is not boosting or
cutting the gain of the internal signal, so the gain is primarily increased by the preamp
module and can be adjusted. A similar test was performed using the function generator input
to verify the gain operation of the preamplifier module. The figure below shows the
oscilloscope measurement of the input signal as well as the output measurement.

Figure 3.15: Oscilloscope measurements of full circuit. Yellow: input. Blue: output

After verifying signal continuity and the preamp operation, a series of AC sweeps
were performed to verify the operation of the both the 70Hz HPF and the 100Hz HPF
respectively. A function generator is set to 184mV sinewave output and connected to a node
on the circuit representing the preamp input of the circuit. The figure below is a plot
representing a manual AC sweep of the circuit with the filter switch set to the 70Hz filter
option.

- 40 -
 Figure 3.16: 70Hz Option Verification

The plot shows the maximum voltage to be 5V, with the cutoff voltage calculated to
be 70.7%, or 3.5V. This output voltage occurs at around 65Hz on the graph. The figure below
is a picture of the oscilloscope window showing input on channel one, and output on channel
two. The measurement shows a 5V maximum voltage at the output at 90.9Hz input
frequency.

Figure 3.17: Oscilloscope window with channel 1: Input and channel 2: Output

- 41 -
 Next, the switch position is changed to the 100Hz HPF circuit option, and an AC
sweep is performed to verify its operation. The figure below is a graphical representation of
the manual AC sweep.

Figure 3.18: 100 Hz Option Verification

The graph shows a maximum output voltage of 5V with the expected cutoff voltage of
3.5V. This voltage occurs at around 95Hz as shown on the graph. In both graphs, the output
shows the functionality of the 60Hz high-pass filter that is built into the preamp module of
the device. The 70Hz HPF specifically shows that the output signal at 60Hz has a max
voltage of approximate 3.5V which is the expected 70.7% of the maximum output voltage
through the sweep. This result and the previous outcome are within expectations of practical
device testing.

Once the filter section has been tested and verified. More AC sweeps are performed to
verify the operation of the parametric equalizer module. Each of three AC sweeps are done
with the “frequency sweep” potentiometer of the parametric equalizer set to the minimum
setting, the mid-point setting, and the maximum setting. With the “boost/cut” potentiometer
set to boost 10dB, the expected outcome of each sweep is a positive gain curve with peaks at
the center frequency.

The figures below each show a graph of the maximum voltage at the output versus the
input frequency of a sinusoidal waveform with peak magnitude of 184mV.

- 42 -
 (a)

(b)

(c)

Figure 3.19: AC Sweeps for Parametric EQ Verification (a)10%, (b)50%, (c)100%

- 43 -
 In Figure 3.19a the maximum gain occurs at approximately 130Hz. This is most likely
due to the 70Hz HPF causing the frequency response to drop off steeply around that
frequency. In Figure 3.19b, the peak gain occurs at approximately 280Hz with a much wider
roll-off on each side of the center frequency. This would need to be rectified with much
steeper roll-off above and below the center frequency to be an effective parametric equalizer.
In Figure 3.19c the peak gain occurs at 1.45kHz with a wide roll-off toward the upper
frequencies. The figures above show the gain boosting operation of the parametric EQ. The
figure below shows the gain cutting operation of the parametric EQ module.

Figure 3.20: AC sweep chart showing Parametric EQ boost/cut potentiometer at minimum,

frequency sweep potentiometer at minimum

After verifying the operation of the parametric equalizer module, the compressor
module verification is next addressed with a combination of microphone and function
generator inputs, due to the unique nature of the operation of audio compressors. First the
general operation of the compressor is verified using the microphone audio input. The figure
below shows the compression of the input signal.

- 44 -
 Figure 3.21: Yellow: Audio input signal. Blue: Compressed output signal

The compressor operation is verified using a threshold of 3V and the ratio

potentiometer set to approximately 50%. Next, to verify the attack and release, the charge and
discharge time of the peak detector module with the compressor module is measured.

The figure below is a picture of the oscilloscope window showing the audio input
signal, and measurement of the capacitor voltage.

Figure 3.22: Oscilloscope measurement window. In yellow: Audio input signal. In blue:
Capacitor voltage

The rise time of the capacitor voltage shows the charge time of the capacitor to be
approximately 20ms, and the discharge time of the capacitor to be approximately 90ms. This

- 45 -
falls within acceptable range as referenced by the engineering requirements of the project, with
attack times ranging from 3ms to 25ms, and release times from 95ms to 180ms.

The final component of the circuit to be verified as referenced in the engineering

requirements of the project is the VU meter. This component is verified on a breadboard due
to time constraints preventing inclusion of the LM3915 IC chip as well as the 10 segment LED
Display on the PCB containing other modules and components of the circuit.

Figure 3.23 shows the VU meter at minimum, mid-point, and maximum input voltage
levels, demonstrating visual verification of the module.

(a)

(b)

- 46 -
 (c)

Figure 3.23: VU meter operational verification at (a)10%, (b)50%, (c) 100%

3.4. PCB Design

Eagle CAD was used for the PCB design for this project. As mentioned, we acknowledge
Sabrina Hurst in assisting with the PCB design for this project. She gave us tips and tricks
about how to route and place components. The following figures are the 3D renders of the PCB
with and without the components.

(a)

(b)
- 47 -
 (c)

Figure 3.24: 3D Render of PCB (a)top view, (b) side view, (c)No components

3.4.1. Soldering Work

Both members of the group contributed with the soldering. The surface mounted soldering
took place at Dustin P’s jobsite. In order to solder the ICs on the board, a microscope was
used, and a fine tip was used for precision. The following was the soldering progress and
setup.

(a)

(b)
- 48 -
 (c)

Figure 3.25: (a)Soldering Station and (b)Soldering Progress

- 49 -
Chapter 4
Non-Technical Issues

Summary
In this chapter, the fiscal budget of the project will be discussed, as well
as environmental, safety, ethical, sustainability, and social aspects of the
project.

4.1 Budget
4.2 Timeline
4.3 Environmental Aspects
4.4 Health and safety
4.5 Ethical Aspects
4.6 Social Aspects
4.7 Sustainability
- 50 -
4.1 Budget
There were many changes to the budget and timeline from the initial proposal phase of this
project. The number of certain components that were initially assumed necessary was
determined to be less, although the number assumed for other components was assumed to be
underestimated. Also, the price of certain components was determined to be less than initial
assumptions, leading to a drastic reduction in total cost for the parts needed for the device. Still,
some equipment was purchased to facilitate the project's construction, such as a soldering
station with soldering iron and heat gun. The timeline drastically changed from the original
assumptions made during the proposal phase. This was due to a multitude of factors ranging
from extended delivery time of components to more troubleshooting and modification required
during the construction and testing process.

4.1.1 Final Project Budget

The initial proposed budget for the project included higher costs for PCB printing as well as an
estimate for the custom 3D printer project box. The table below shows the breakdown of
proposed components and costs for this project.

Table 4.1 – Proposed Project Budget

- 51 -
Table 4.2 below shows the actual breakdown of these components and costs for this project.

Table 4.2 – Final Project Budget

Item Unit Price (In USD) Quantity Total Cost (In USD)
Jameco 12V Dual
33.24 1 33.24
Power Supply Kit
Female XLR Input 3.79 1 3.79
Male XLR Input 2.91 1 2.91
20pcs Dual Gang
Stereo 6-pin 11.70 1 11.70
Potentiometers
1K-1M ohm
17.99 1 17.99
Potentiometer Kit
OPA4134 IC 5.11 10 56.05
5pcs 100k ohm
Logarithmic 18.93 1 18.93
Potentiometer kit
Diode pack 100pcs 4.99 1 4.99
10-Segment LED
7.99 2 15.98
Display
Todiys LM3915 IC 13.95 2 27.90
14-pin DOIC
11.15 3 33.45
adapters
Project box 15.17 1 15.17
Switches 3.00 3 9.00
Custom PCB 5.20 5 26.00

Total 277.10

- 52 -
4.2. Timeline
Table 4.3 below is a Gantt chart of the proposed timeline for this project. Table 4.4 is a
detailed proposed timeline by week.
Table 4.3 – Proposed Timeline Gantt Chart

Table 4.4 – Detailed Proposed Timeline

- 53 -
The tables below show the final project timeline presented in a Gantt Chart, as well as a
detailed timeline by week.

Table 4.5 – Project Timeline Gantt Chart

Table 4.6 – Project Final detailed timeline by week

- 54 -
The final project timeline shows that more bench testing and troubleshooting than initially was
assumed. Due to the nature of breadboard testing, there were many inconsistencies with
expected output signals throughout the testing process.

4.3 Environmental Aspects

The device is friendly to the environment and does not cause any harm since it is a low-voltage
electrical circuit contained within an enclosure.

4.4 Health and safety

The system developed is electrically safe and not harmful when used properly within standard
operating conditions. Because this project is a series of low-power components, the chances
of physical harm or accidents are low. During the prototyping phase, all wiring will be
shielded, and no electrical contact will be exposed. Any possibly means of risk to the end user
will be considered and prevented at all costs. The only thing that is a concern is the power
supply. If there were a power outage or surge, the power supply would be shorted first because
the device will use an external power supply. There is a slight chance that some component
will short out, but with skillful soldering and conductive/shielded solder, there will be less of a
chance of the device shorting out and not working.

4.5 Ethical Aspects

All references used for the completion of this project are properly cited using IEEE Citation
format. The IEEE Code of Ethics is also followed throughout this project [14]. Some codes
taken into specific consideration are:

• Code 3. To avoid real or perceived conflicts of interest whenever possible, and

to disclose them to affected parties when they do exist.
• Code 5: To seek, accept, and offer honest criticism of technical work, to
acknowledge and correct errors, to be honest and realistic in stating claims or
estimates based on available data, and to credit properly the contributions of
others.
• Code 6. To maintain and improve our technical competence and to undertake
technological tasks for others only if qualified by training or experience, or
after full disclosure of pertinent limitations.

- 55 -
These codes were most applicable to the project ethical standards based on the parameters of
the project itself and its scope.

4.6 Social Aspects

With the recent pandemic of the past two or so years, more people find other ways to make a
living, including music production and mobile DJing. Our device has a heavy impact on the
entry level DJ or music producer; it has all the features a beginner needs to get their foot in the
door of music and audio. It is portable enough for them to take anywhere and easy to navigate;
its stages are labeled, and the interface is easy to use. The most important social aspect we
would like to see our product in educational audio classes to teach students the basics of audio
processing.

4.7 Sustainability
Our product will sustain in the educational field and will expand to the professional audio scene
with iterations supporting all kinds of microphones and instruments. The intent of this device
is to be a handy tool if portability is the focus. The prospects of this device will be using better
and better op-amps, components, and build.

- 56 -
Chapter 5
Conclusion

Summary

In this chapter we will give our overall thoughts and projections for this project.
The future considerations and modifications will be stated and our future
prospects in the educational field will be stated.

5.1 Summary and Conclusion

5.2 Suggestions for Future Work

- 57 -
5.1. Summary and Conclusion
The D2A 3-in-1 Audio Signal Processing Unit allows the growing mobile DJ and music
producer to have all their necessary tools in the palm of their hands without having to carry all
this heavy equipment with them. They’ll be able to concentrate more on the music rather than
the bulk of the equipment.

Our main goal from the get-go was to make it easier for users of our product to have all
the tools they need to get going with all their audio needs. With further thought, our product
could be used for educational purposes. Classrooms all over the world could use our product
to teach the starving artists and producers all about audio processing and signal flow.

While working on this project, it opened our eyes to the numerous possibilities and
design choices that can help the user even further than what were presented in this project.
Comparing the proposed requirements to the final verified requirements, there were many
considerations we had to factor in, such as internal component noise, available components,
PCB design, as well as XLR input and output requirements. We realized that we had to make
some difficult decisions to meet the deadline. If time permitted, we would have been able to
use advanced testing methods to test the audio from beginning to end so we could prove our
product accomplished what was intended during the proposal phase.

5.2. Suggestions for Future Work

Throughout the course of this project through the proposal and project phase, we have received
feedback from our advisor, Dr. Ejaz, to different professors, such as Professor Young and
Professor Valery. Their feedback was the force that drove this project to its final result. We
came up with a list of things that would improve the project to its full potential.

One change that we’d like to make is to support all microphones by adding phantom
power in our preamp circuit. According to Shure, one of the leading audio manufacturers,
phantom power is a method delivering voltage to instruments or mics that have active
components in the instruments themselves such as condenser microphones. To achieve this,
there needs to be a circuit inserted in between the XLR input and microphone preamp like the
one below.

- 58 -
 Figure 5.1: Power Phantom Circuit [10]

Another future change we’d make is making the noise filter section user-defined by
adding a variable cutoff knob frequency. The goal of our device is to be used on all instruments
and depending on the instrument and sound you’re looking for; a variable cutoff noise filter
would be the most ideal option since the noise frequencies vary. For the parametric EQ module,
steeper roll-off above and below the center frequency would provide greater control of an audio
signal regarding either noise or desirable sound.

A final change we would make is the design to the compressor. Instead of a VCA-based
compressor, we would change the mode of operation to a JFET-based compressor (also known
as a dynamic range compressor). Throughout our research and implementation of the VCA, it
was designed for a particular voltage input level and could not go past a certain voltage, every
value of the components or stages were dependant on the previous module. Each stage was
only designed for a specific input or voltage level. There was no adaptability or flexibility
without changing the whole design. With FET compressors, they’re adaptable and can give a
good tone to the instruments such as drums; with VCA compressors, there is no tone
manipulation, so the output signal more closely represents the input signal.

- 59 -
 References

[1] W. D. Stanley, Operational Amplifiers with Linear Integrated Circuits. Upper

Saddle River, NJ: Prentice Hall, 2002.

[2] “ART Tube MP Studio V3 Tube Microphone Preamp | Reverb,” reverb.com.

https://reverb.com/p/art-tube-mp-studio-v3-tube-microphone-preamp-2010-2023

[3] “8 Band EQ by W2IHY - SPECIAL,” Cheapham.com by Hometek LLC.

https://www.cheapham.com/8-band-eq-by-w2ihy-special/

[4] “What Is Audio Compression? Everything You Need to Know,” iZotope.

https://www.izotope.com/en/learn/what-is-audio-compression.html

[5] musictoyshop.com, “Sebatron SMAC Compressor,” musictoyshop.com.

https://musictoyshop.com/shop/ols/products/sebatron-smac-compressor

[6] B. Technologies, “Filter Topology Face Off: A closer look at the top 4 filter types,”
blog.bliley.com, Aug. 02, 2016. https://blog.bliley.com/filter-typology-face-off-a-closer-look-
at-the-top-4-filter-types

[7] M. McAllister, “Types of Audio Compressors and Their Uses,” Produce Like A Pro,
Sep. 08, 2021. https://producelikeapro.com/blog/types-of-audio-
compressors/#:~:text=The%205%20types%20of%20audio

[8] Texas Instruments, “OPAx134 SoundPlus™ High Performance Audio Operational

Amplifiers,” OPAx134 datasheet, Dec. 1997 [Revised Oct. 2015].

[9] Texas Instrument, “LM3915 Dot/Bar Display Driver,” LM3915 datasheet, Jan. 2000
[Revised Mar. 2013].

[10] “Balanced I/O,” sound-au.com. https://sound-au.com/articles/balanced-io.htm#s2

[11] “Neutrik NA3FMX – Correct Phase Made Easy» Adventures in Hifi Audio,”
www.adventuresinhifiaudio.com. http://www.adventuresinhifiaudio.com/21/12/2018/neutrik-
na3fmx-correct-phase-made-easy/

[12] Jameco Electronics, “20626 (JE215) Adjustable Dual Power Supply Kit,” Kit Instructions.

[13] S. W. Smith, “Introduction to Digital Filters,” in The Scientist and Engineer’s Guide to
Digital Signal Processing, San Diego, CA: California Technical Pub., 1997

[14] IEEE Code of Ethics, https://www.ieee.org/about/corporate/governance/p7-8.html

- 60 -
[15] M. K. “Designing a simple audio compressor from scratch” Youtube. Jan 1, 2023 Video
File. Available: https://www.youtube.com/watch?v=Wag-yTyAxPA&t=908s [Accessed Jul.
18, 2023]

[16] M. Hicks, “Audio Compression Basics | Universal Audio,” Uaudio.com, 2019.

https://www.uaudio.com/blog/audio-compression-basics/

- 61 -
Appendix A
Normalized Butterworth Filter Design

Summary
In this appendix we present the Normalized Butterworth Filter Design for high-
pass and low-pass filters. We will go through a step-by-step process.

A1. HPF Filter Design

A2. LPF Filter Design

- 62 -
A1. HPF Filter Design
In our project, we used 4th and 5th order unity-gain filters. We will go through the design
process, tables, and equations.
The first step is defining the normalized cutoff frequency, 3-dB frequency, and final
capacitor value. Drawing a normalized model with the normalized resistor values on Table A1
is next. Figure A1 shows the normalized model. With 5th order filters, the three-pole section
goes before the two-pole section with the corresponding values. The following is the design
for a 5th order, 30 Hz HPF. The final capacitor value will be 100nF.
Table A1 – Resistance Values for HP Butterworth Active Filter Designs

- 63 -
 (a)

(b)
Figure A1.1: (a)5th Order and (b)4th Order Model

Next is calculating the frequency scaling factor by the using equation A1.1 and dividing the
capacitor values by the frequency scaling factor. 𝜔𝑟 is the cutoff frequency in radians and ̅̅̅̅
𝜔𝑟
is the normalized cutoff frequency. Equation A1.2 is the capacitance value when it is divided
by the frequency scaling factor.

𝜔𝑟 60𝜋
𝐾𝑓 = = = 60𝜋 (A1.1)
𝜔̅̅𝑟
̅̅ 1
1
𝐶′ = = 5.305𝑚𝐹 (A1.2)
𝐾𝑓

- 64 -
After finding the frequency scaling factor and dividing the capacitors by it, next. impedance
scaling factor will be done to have the all the capacitors equal to the final capacitor value,
multiply all the resistor values by the impedance scaling factor to get the final resistor values.
Equation A1.3 is for the impedance scaling factor and Figure A1.2 is final HPF design.
𝐶′ 5.3052𝑚𝐹
𝐾𝑟 = = = 53.0516 (A1.3)
𝐶𝑓𝑖𝑛𝑎𝑙 100𝑛𝐹
R1
R4
27.61k
V+
U1 16.4k
C1 C2 C3
V+
+ V+ C4 C5 U2
100n 100n 100n
R2 R3
OUT + V+
39.18k 125.9k 100n 100n
R5
- V- OUT
171.73k
0 0 OPAx134
V- - V-
0 OPAx134
V-

Figure A1.2: 30Hz HPF Final Design

- 65 -
A2. LPF Filter Design
LPF filter design goes through the same process as the HPF filter design: calculating the
frequency scaling factor, dividing the capacitors by frequency scaling factor, calculating the
impedance scaling factor, and dividing/multiplying capacitors and resistors, respectively, by
the impedance scaling factor. The only difference between the two are the poles and the
components switch places. HPFs have resistors as poles same capacitor values while LPFs
have capacitors as poles and same resistor values on each stage.
For our project, we will design a 4th order, 45k Hz LPF with C1 for both stages to be
10nF. Table A2 shows the capacitance values for each pole and Figure A2 shows the model
for the normalized Butterworth LPF filter design.
Table A2 – Capacitance Values for Low-Pass Butterworth Active Filter Designs

Figure A2: Model for a 5th Order LPF

- 66 -
Using the Equation A1.1, we find the frequency scaling factor:
𝜔𝑟 90000𝜋
𝐾𝑓 = = = 90000𝜋
̅𝜔̅̅̅𝑟 1

From Equation A1.2, the capacitors are divided by the frequency scaling factor:
[1.082 2.613 0.9241 0.3852]
𝐶′ = = [3.8268𝑛𝐹 9.2416𝑛𝐹 3.2683𝑛𝐹 1.3624𝑛𝐹]
90000𝜋

Given that both C1 of both stages had to equal 10nF, C1 from the first and second stage is
divided by 10nF to give the impedance scaling factor of each stage. This is similar to Equation
A1.3 except C1 from both stages are different:
𝐶1𝑠𝑡𝑎𝑔𝑒1 3.8268𝑛𝐹 𝐶1𝑠𝑡𝑎𝑔𝑒2 3.2683𝑛𝐹
𝐾𝑟1 = = = 382.68 𝐾𝑟2 = = = 326.83 (𝐴2.1)
𝐶𝑓𝑖𝑛𝑎𝑙 10𝑛𝐹 𝐶𝑓𝑖𝑛𝑎𝑙 10𝑛𝐹

Finally, the resistors were multiplied and capacitors by their respective impedance factors and
the final design is below:
C10

C12
10n
V+
U6 10n
R12 R13
V+
+ V+ R14 R15 U7
382.68 382.68
C11
OUT + V+
8.54n 924.16 924.16
C13
0 - V-
1.46n
OUT
OPAx134
V- 0 - V-
OPAx134
V-

- 67 -
Appendix B
Noninverting Amplifier Gain

The main configuration of amplifiers we are using in this project are noninverting amplifiers.
An example of a noninverting amplifier is in the following figure. In this configuration, the
input voltage goes through the positive terminal of the op-amp. On the negative input there’s
a voltage divider that goes between the input and feedback resistors; the input resistance
terminates to ground and the feedback resistor goes to the output.

Figure B.1: Noninverting Amplifier

It is called a noninverting amplifier because the output is in phase with input meaning
both input and output waveforms start and end the same way. No current flows through the
inverting terminal of the op-amp and the voltage divider is as follows:

𝑣𝑜 = 𝑣𝑖 + 𝑣𝑓 (B. 1)

𝑅𝑓 𝑅𝑓
𝑣𝑜 = 𝑣𝑖 + 𝑣𝑖 = (1 + ) 𝑣𝑖 (B. 2)
𝑅𝑖 𝑅𝑖

The gain of the amplifier is determined by the following:

- 68 -
 𝑅𝑓
𝑣𝑜 (1 + 𝑅𝑖 ) 𝑣𝑖 𝑅𝑓
𝐴𝐶𝐿 = = =1+ (B. 3)
𝑣𝑖 𝑣𝑖 𝑅𝑖

In audio, the gain is usually represented in decibels. To convert the gain factor to
decibels, the following equation is used:
𝐴𝐶𝐿 (𝑑𝐵) = 20 log(𝐴𝐶𝐿 ) (B. 4)

- 69 -
Appendix C
Datasheet of Components

Summary
In this appendix we present the portions of the datasheets that were useful to the
design of our project.

C1. OPAx134 Datasheet

C2. LM3915 Datasheet
C3. Jameco Power Supply Kit Instructions

- 70 -
- 71 -
- 72 -
- 73 -
- 74 -
- 75 -
Dustin Pegalee
▪ AS Degree in Audio Engineering
▪ Interned at Local Recording Studios
▪ Currently in BS in Electrical and Computer
Engineering Technology
▪ Work as a repair technician

Dustin Funk
▪ Certified Audio Engineer
▪ BSECET expected graduation December 2023
▪ Mobile DJ
▪ IT Professional

- 76 -