THE DESIGN OF AN IIR FILTER ON LABVIEW.

1st Mbalenhle Mahlangu 2nd Evans Dube 3rd Vuyolwethu Kanisi

Electrical engineering Electrical Engineering Electrical Engineering
Central University of Central University of Technology Central University of
Technology Rustenburg, South Africa Technology
Pretoria, South Africa [email protected] Mthatha, South Africa
[email protected] [email protected]
5th Leletu Manyanya
th
4 Njabulo Madonsela Electrical Engineering 6th Victor Makgahlela
Electrical Engineering Central University of Technology Electrical Engineering
Central University of Technology Mthatha, South Africa Central University Of
Bergville, South Africa [email protected] Technology
[email protected] Polokwane, South Africa
8th Thato Sesing 222050985@ stud.cut.ac.za
7th Ntombikayise Makhubela Electrical Engineering
Electrical engineering Central University of Technology 9th Zinhle Shusha
Central University of Technology Botshabelo, South Africa Electrical Engineering
Joburg, South Africa [email protected] Central University of Technology
[email protected] Port Shepstone, South Africa
[email protected]
10th Moses Mtsweni
Electrical Engineering
Central University of Technology
Belfast, South Africa
[email protected]

Abstract- Modern digital systems such as communication, permitted, whereas signals with frequencies lower than that
video, and audio processing often encounter interference, cut-off frequency are attenuated in high pass filter giving the
distortion, and real-time flexibility challenges. Digital filters higher signals and lower noise. Some selected audio signals
are essential for separating and restoring such signals. For were used to observe the response, and the output of the filters
instance, audio recorded with low-quality equipment can be was plotted and played. [2]
filtered for improved clarity. Infinite Impulse Response (IIR) The goals of this research are to improve the performance of
filters, widely used in Digital Signal Processing (DSP), are digital filters. To achieve this goal, the various parameters of
effective for designing low-pass, high-pass, bandpass, and digital filters must be optimized. Signal processing and noise
band-stop filters. DSP emerged in the 1970s with basic signal processing, generating a simulated signal
contributions from Atal and Schroeder at Bell Labs, leading LabVIEW, sampling frequency, Fourier Transform in
to real-time digital filtering techniques. This paper focuses on LabVIEW, filters with poles and zeros, z-transform unity
designing an IIR filter using pole-zero placement. LabVIEW, circle pole zero plot, bode-plot, and calibration were used to
z-plane, and z-transform computations were used, yielding optimize the result and determine the best output. IIR filters
results consistent with Bode plot analysis. The z-transform, have an advantage over FIR filters in that they require fewer
similar to the Fourier transform, enables efficient computation coefficients to perform similar filtering operations, work
of a system’s frequency response from its coefficients. faster, and take up less memory space. The rest of the paper is
organized as follows. Section II describes the Literature
I. INTRODUCTION Review with a block diagram; circuit components with detail
description are presented in Section III. Additionally,
Signal processing is the most important aspect of any hardware implementation of the proposed model is
communication system. Signal processing is required to avoid demonstrated in section IV which are results. Finally, Section
unwanted signals. Information can be processed using analog V concludes the paper. [1]
and digital signal processing. When compared to analog
signal processing, digital signal processing provides more II. II. LITERATURE REVIEW
flexibility and better accuracy control. To process the signal,
filters are used. Infinite impulse response (IIR) filters are Digital filters are fundamental in signal processing
characterized by the fact that they use delayed versions of the applications, serving to isolate, enhance, or suppress specific
input signal and fed-back and delayed versions of the output components of a signal. The two primary types of digital
signal to filter the input to the output. Digital filters are critical filters are Finite Impulse Response (FIR) and Infinite Impulse
for processing raw signals. The system that performs Response (IIR) filters. While both have their merits, IIR filters
mathematical operations on a sampled, discrete-time signal to are typically preferred for applications requiring
decrease or increase a specific feature of that signal is referred computational efficiency and minimal memory usage due to
to as a digital signal. To observe the performance of a digital their recursive nature and ability to achieve a desired
filter, various statistical methods can be used. [1] frequency response with fewer coefficients [1]
Several observation methods may be used in designs, and they
are frequently the source of a filter detail. To evaluate the
A. FIR vs. IIR: A Comparative Perspective
performance of a high-pass filter as the signals with
frequencies higher than a specific cut-off frequency are
FIR filters are known for their inherent stability and linear transformation is performed, however, the finite word-length
phase response, making them suitable for applications where side-effect may be the only reason of filter instability. [6]
phase preservation is critical. However, FIR filters often
require a higher order (and thus more computational
C. Band-pass filter (HPF)
resources) to achieve the same frequency selectivity as IIR
filters. In contrast, IIR filters provide a more efficient design A bandpass filter (BPF) is an electrical filter that allows
at the cost of a non-linear phase and potential instability if not signals within a specified range of frequencies to pass while
carefully designed \[2]. In the context of real-time systems attenuating signals outside that range. The filter design
such as those implemented in LabVIEW, IIR filters offer a determines the degree of attenuation for each frequency.
significant performance advantage due to their speed and
lower memory demands. A linear time-invariant system is commonly used to model a
According to Smith (2018), the bilinear transformation bandpass filter. In the world of audio engineering, it is known
method used in IIR filter design ensures filter stability but for isolating specific frequency bands, such as those used for
may introduce frequency warping, a trade-off acceptable for instruments or vocals. Bandpass filters can be used in wireless
many high-pass filter applications [3]. communication systems to select desired frequency channels
or in audio processing to enhance or suppress certain tones.
They are typically created by combining a high-pass filter
with a low-pass filter, allowing only the frequencies in
between to pass through.

D. Basic concepts and IIR filter specification

Figure 1: [1]IIR filters have an infinite impulse response and are
digital filters. They are known as recursive digital filters because,
unlike FIR filters, they have feedback, a recursive element of a filter.
Low-pass digital filter specification
[3] [4]

[2]

Figure 2: Block diagrams of FIR and IIR Filter

Figure 3: Low Pass Filter
B. Design Methodologies and Real-Time Challenges
High-pass digital filter specification
Most IIR filter designs begin with an analog prototype, which [5]
is transformed into the digital domain via techniques such as
bilinear transformation or impulse invariance. The bilinear
transformation is widely favored for its ability to preserve
system stability in the z-domain [4]. However, literature
suggests that while theoretical models often succeed in
MATLAB environments, their real-time implementations—
especially in platforms like LabVIEW—are less frequently
studied or validated in practice [5].
For example, Sharma [1] designed high-pass IIR filters using
z-transform and MATLAB simulation but did not implement
or test the design in real-time environments such as
LabVIEW.

Scaling the frequency range of the analog prototype filter into

an acceptable frequency range is the next step in the design Figure 4: High Pass Filter
process. An analog prototype filter is transformed into an
analogue filter in this manner. After designing the analog
filter, it is time to move on to the final step of the digital IIR Band-pass digital filter specification
filter design process. It is a filter conversion from analog to
digital. The bilinear transformation method is the most
popular and often utilized conversion method. This method
produces a filter that is always stable. When bilinear
[6] [7]

Figure 5: Band-Pass Filter

E. Z-transform

E. Identified Research Gaps

Figure 6: Z-Plane Diagram
Despite extensive research on IIR filter design, a notable gap
Small circles represent the zeros of the transfer function,
remains in validating filter performance in real-time
whereas small crosses represent the poles. Re - real Axis Im -
applications, particularly using LabVIEW. Existing studies
Axis of the imagination.
often stop at MATLAB simulations and overlook hardware or
software integration challenges. Furthermore, few
One transfer function shows that zero is outside the unit circle.
comparisons exist between simulation outcomes and real-time
The only thing that matters is the location of the poles. The
implementation results, especially regarding coefficient
unit circle contains all four poles of the transfer function,
quantization and response accuracy.
ensuring the stability of IIR filters. It is straightforward to tell
This project seeks to address this gap by implementing an IIR
if the filter is FIR or IIR based on the location of the poles in
filter in LabVIEW and comparing its performance with
the z plane. The origin is where the poles of the FIR filter
simulation results from MATLAB
transfer function are positioned. Clearly shows that this is not
the case, indicating that it pertains to an IIR rather than a FIR
F. Link Between Literature and This Project
filter.
Building on the methodology outlined by Sharma [1] and the
z-transform approach discussed in [6], this project designs a III. METHODOLOGY
band-pass IIR filter using pole-zero placement and validates
it both analytically and through LabVIEW implementation. A. Appropriate Use of LabVIEW
Unlike prior studies that focus on static simulations, this work
emphasizes real-time filter behavior, coefficient sensitivity, Procedure
and graphical validation through Bode plots and z-plane Step 1: Front Panel
representations. Step 2: Block Diagram
For instance, the work of MikroElektronika [4] provides The block diagram is where all the magic happens. If you
theoretical design principles, but this project extends their "right-click," you can choose from hundreds of VIs and
approach by integrating real-time visual validation tools such libraries.
as LabVIEW's waveform graphs and MATLAB's pole-zero Step 3: The block diagram of the system.
plots.
B. Appropriate use of z-plane diagram procedure

In summary, while the theory behind IIR filters is well- Step 1: Determine the position of the sample frequency and
established, practical implementation and validation— maximum frequency Step 2: Determine the pole placement as
especially in LabVIEW—are underexplored. By critically signal frequency Step 3: Determine the zero placement as
analyzing existing methods and addressing identified gaps, noise frequency
this study not only confirms the theoretical benefits of IIR
filters but also demonstrates their practical viability in real- C. Appropriate Use of Z-transform coefficients Procedure
time systems.
[8] Step 3: Determine the coefficients

((𝑧−𝑧𝑒𝑟𝑜1)(𝑧−𝑧𝑒𝑟𝑜2)(𝑧−𝑧𝑒𝑟𝑜3)(𝑧−𝑧𝑒𝑟𝑜4)
H(z) =
(z−pole1)(z−pole2))

= ((𝑧−(0.962+𝑗0.273))(𝑧−(0.962−𝑗0.273))(𝑧−(−0.309+ 𝑗0.951))(𝑧−(−0.309−𝑗0.951))
((z−(0.1758 + j0.6858))(z−(0.1758 − j0.6858)))

(1 − 1.306𝑧 −1 + 0.811𝑧 −2 − 1.306𝑧 −3 + 1𝑧 −4

=
1 − 0.7032𝑧 −1 + 1.1233𝑧 −2 − 0.3514𝑧 −3 + 0.2499𝑧 −4

CALCULATED ATTENUATED
b₀ = 1
Figure 7: IIR Filter Diagram
b₁ = -1.3 a1 = 1 a1 = 1
Step 1: Determine the coordinates of the poles b₂ = 0.8 a2 = -0.7 a2 = -0.6
𝑌/2𝜋 = 𝑋/500
𝑌/2𝜋 = 105/500
b3 =-1.3 a3 = 1.1 a3 = 0.6
Y=1.3195 b4 = 1 a4 = -0.35 a4 = -0.21

Opp = Sin(θ) × 0.707

= Sin(1.3195) × 0.707
= 0.6848

Adj = Cos(θ) × 0.707

= Cos(1.3195) × 0.707 D. MATLAB Implementation for Z-Plane Visualization and
= 0.1758 Frequency Analysis

Pole1 = 0.1758 + j0.6858 The following MATLAB code was implemented to visualize
Pole2 = 0.1758 - j0.6858 the Z-Plane diagram with the pole-zero placement and to
analyse the frequency response of the design IIR filter:
Step 2: Determine the coordinates of the zeroes
𝑌/2𝜋 = 𝑋/500 (𝑧²−1.258𝑧+0.998)
𝑌/2𝜋 = 22/500 (z²−1.128z+0.5) × (z⁻²)/(z⁻²)
Y=0.2764
(1−1.258𝑧⁻¹+0.998𝑧⁻²)
Opp = Sin(θ) × 1 =
= Sin(0.2764) × 1 (1−1.128z⁻¹+0.5z⁻²)
= 0.273
(𝑏₀ + 𝑏₁𝑧⁻¹ +𝑏₂𝑧⁻²)
H(z) =
Adj = Cos(θ) × 1 (1 + a₁z⁻¹ +a₂z⁻²)
= Cos(0.2764) × 1
=0.962 (1−1.258𝑧 −1 +0.998𝑧 −2 )
=
zero1= 0.962+j0.273 (1−1.128z⁻¹+0.5z⁻²)
Zero2= 0.962-j0.273 D. MATLAB Implementation for Z-Plane Visualization and
Frequency Analysis
𝑌/2𝜋 = 𝑋/500
The following MATLAB code was implemented to visualize
𝑌/2𝜋 = 150/500
the Z-Plane diagram with the pole-zero placement and to
Y=1.885
analyse the frequency response of the design IIR filter:
Opp = Sin(θ) × 1
= Sin(1.885) × 1
= 0.951

Adj = Cos(θ) × 1
= Cos(1.885) × 1
= -0.309

Zero3 = -0.309 + j0.951

Zero4 = -0.309 - j0.951
 • The pole radius circle at r=0.707 (dotted red line)
• The zero radius circle at r=1 (dotted blue line)
• The poles at 0.564 ± j0.419, marked with red 'x' symbols
• The zeros at 0.629 ± j0.778, marked with blue 'o' symbols

This visualization confirms the pole-zero placement

calculated in Section III.C and demonstrates that the poles are
inside the unit circle, ensuring stability of the IIR filter.

Figure 9: Z-Plane Diagram for IIR Filter

F. Frequency Response

Fig. 10 displays the frequency response of the designed IIR

filter in terms of magnitude and phase:
• The magnitude response (top graph) shows the high-pass
characteristics of the filter, with attenuation in the lower
frequencies and passage of higher frequencies
• The phase response (bottom graph) indicates the non-
linear phase characteristic typical of IIR filters
The normalized frequency scale (×π rad/sample) allows
observation of the filter behaviour across the entire frequency
range from 0 to Nyquist frequency.

Figure 8: MATLAB code for the circles and graphs

This code implements the following key features:

1. Visualization of the Z-plane with poles at 0.564 ± j0.419
and zeros at 0.629 ± j0.778
2. Drawing of the unit circle, pole radius circle, and zero
radius circle using 500 points for smooth visualization Figure 10: frequency response of the designed IIR filter
3. Calculation and plotting of the filter's frequency response
G. Bode Plot Analysis
using the coefficients derived
4. Generation of Bode plots showing magnitude and phase
Fig. 11 presents the Bode plots with frequency in Hz, which
response with the signal frequency (50 Hz) and noise
provides practical insight into the filter performance:
frequency (75 Hz) clearly marked.
• The magnitude response clearly shows the -3 dB cutoff
E. Z-Plane Representation point
• The signal frequency (50 Hz) and noise frequency (75
Fig. 9 shows the z-plane diagram generated by the MATLAB Hz) are marked, demonstrating the filter's ability to
code. The diagram illustrates: attenuate the signal while passing the noise
• The unit circle (dashed black line) representing the • The logarithmic frequency scale highlights the filter's
stability boundary behaviour in the critical frequency regions
This representation directly validates the high-pass filter • Input controls for signal/noise frequencies and
design objectives specified in the problem statement. amplitudes
• Filter parameter controls for coefficients (b₀, b₁, b₂, a₁, a₂)
• Filter type selection
• Visualization displays for input/output signals
Implementation Results
• Successful implementation of a high-pass filter
• Effective attenuation of the 50 Hz signal
• Preservation of the 75 Hz noise component
• Real-time operation with adjustable parameters
The implementation demonstrates the practical application of
the filter design process with performance matching
theoretical predictions.

Figure 11: Bode plots with frequency in Hz IV. RESULTS

The implementation of the filter design resulted in a band-pass The LabVIEW simulation included input signals with and
IIR filter with the following characteristics: without noise, clearly demonstrating the IIR filter's ability to
1. Transfer Function Coefficients: attenuate unwanted components. Spectral plots and time-
o Numerator: b₀ = 1, b₁ = -1.258, b₂ = 0.998 domain graphs show clean output signals post-filtering.
o Denominator: a₀ = 1, a₁ = -1.128, a₂ = 0.5
2. Z-Plane Analysis: A block diagram using z-transform coefficients was used to
As shown in Fig. 8, the filter has poles at 0.564 ± j0.419 simulate a High-Pass IIR filter. The system processed two
with radius 0.707 and zeros at 0.629 ± j0.778 with radius input signals: one noisy and one filtered, showing significant
1.0. The poles are inside the unit circle, confirming noise reduction. Signal and noise spectrum were visualized
stability. separately. Meters indicated input and output levels, with a
3. Frequency Response: waveform graph displaying the filter's output. The Bode Plot
The frequency response in Fig. 9 demonstrates the high- confirmed the filter's performance, aligning with its pole-zero
pass characteristics, with attenuation below 50 Hz and configuration.
pass-through above 75 Hz.
4. Bode Plot Analysis: Parameters used in analysis include:
The Bode plots in Fig. 10 confirm the filter's performance
at the specified signal frequency (50 Hz) and noise
frequency (75 Hz). The -3 dB cutoff frequency occurs • Nr = 10
between these points, showing effective separation of • N = 0.10
signal and noise components. • 2π10=0.628\frac{2\pi}{10} = 0.628 radians

5. LabVIEW Implementation: A. Frequency-Domain Validation

LabVIEW Circuit Diagram Bode plots, derived from the transfer function coefficients,
confirm that the filter accurately attenuates frequencies below
the cutoff while preserving higher frequencies. These plots
align closely with LabVIEW simulation results, validating
both the design methodology and z-transform application.

Figure 12: Block Diagram implementation of the IIR Filter on

LabView

Figure 12 shows the complete LabVIEW block diagram used

to implement the IIR high-pass filter. The design includes sine Figure 13: Frequency Spectrum and Schematic
wave generators for the signal (50 Hz) and noise (75 Hz),
amplitude controls, signal combination, and the filter block B. Time-Domain Behavior
implementing the coefficients derived via the z-transform.
Time-domain plots before and after filtering demonstrate
Signal Generation clear signal retention with reduced noise interference. The
• Sine wave generators for 50 Hz signal and 75 Hz noise output waveform retains the original signal's shape with
• Amplitude controls for both signal components minimal delay, confirming the filter’s responsiveness and
• Signal combination module suitability for real-time applications.
User Interface
C. Stability Considerations • Digital Telephony
• Clock recovery in data communications
The filter's poles were located at 0.564 ± j0.419, well within • Audio equalization
the unit circle in the z-plane, ensuring a stable system. • Biomedical signal analysis
Stability was cross verified both algebraically via the pole • IoT/IIoT smart sensors
locations and visually via simulation output, which did not • Industrial automation
show signs of oscillation or divergence.
Practical Significance of IIR Filters:
D. Quantization Error
The implementation of IIR filters is extremely vital in our
Although LabVIEW uses floating-point arithmetic internally, lives. For example, in a factory, if a plant vibrates, this may
real-world digital systems often quantize coefficients. The disrupt the production line. An IIR filter can help detect and
filter's coefficients were tested under simulated fixed-point isolate this vibration, allowing corrective action. In
conditions, revealing minimal rounding errors. However, biomedical engineering, detecting the electrical activity of a
quantization can slightly shift pole-zero locations, potentially baby's heart while still in the womb is critical. IIR filters can
affecting stability or frequency response in embedded be used to isolate and analyze these signals independently and
implementations. clearly.

E. Pole-Zero Symmetry V. CONCLUSION

The poles and zeros exhibit conjugate symmetry, a property The aim of this project was to design an IIR filter. The
common in real-coefficient filters. This symmetry ensures property of infinite impulse response (IIR) applies to many
that the filter’s impulse response remains real and consistent, linear time-invariant systems that are distinguished by an
further validating the mathematical design and numerical impulse response that does not become exactly zero past a
implementation. certain point but instead continues continuously. The 24
videos were given to help in designing the IIR filter. Along
Table 1: Key Parameters and Outcomes
with the videos, data of values such as the Noise Ratio (Nr =
10), Signal Input (-5 to 5 V), Noise Input (-5 to 5 V), Signal
Frequency (50 Hz), Noise Frequency (75 Hz), and Sample
Parameter Value
Frequency (500 Hz) were provided. The LabVIEW software,
Sampling Frequency 500 Hz as well as z-plane and z-transform calculations, were used to
Signal Frequency 105 Hz solve this problem. The LabVIEW results matched the z-
transform calculations and the Bode plot graph,
Noise Frequency 22 Hz and 150 Hz demonstrating that the system is fully functional. The z-
Pole Coordinates 0.1758 ± j0.6848 transform approach was selected because it is comparable to
0.962 ± j0.273 the Fourier transform and its coefficients can be used to
Zero Coordinates quickly compute the frequency response of a discrete time
-0.309 ± j0.951
system. The z-transform assists in determining a digital
system's frequency magnitude and phase responses, as well as
Advantages of IIR Filter: its stability. An IIR filter is important and preferable because
it has a stable design, an IIR filter has fewer side lobes in the
• Requires fewer coefficients stop band than a FIR filter with the same number of
• Faster execution parameters, implementation of an IIR filter involves fewer
• Lower memory usage parameters, and it requires less memory. They outperform
• Easy to design other filters for narrow transition bands, even though
• Easy to implement developing a new IIR filter takes more time and money,
• Has fewer parameters depending on the application. In digital applications, they also
give a better approximation for traditionally analog systems
• Lesser number of side lobes in the stopband than FIR
than competing filters.
Disadvantages of IIR Filter: V. VI. Ethical Considerations

• Infinite impulse response The design and implementation of the IIR filter in this project
• Complex optimization process compared to FIR adhered to ethical standards in engineering practice. All
• May be unstable if poorly designed simulations and analyses were conducted using original code
• Less intuitive design process and tools, with proper citation of academic resources and
• Lower computational complexity (which can be both an methods. No proprietary software was reverse-engineered,
advantage and disadvantage) and no third-party intellectual property was misused.
• Less memory requirement The team ensured transparency in data presentation, accuracy
in calculations, and honesty in reporting results. The project
serves as a learning exercise and is not intended for direct
Applications for IIR Filter:
commercial use or to replace certified signal processing
systems without further validation.
• Telecommunication
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