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ANALYSIS OF THE CONTROLLABILITY OF SOME DIGITAL FILTERS WITH A FINITE IMPULSE RESPONSE
D. A. Guzhva , К.О. Sever, А. А. Morozov2021-08-11Abstract ▼This overview article covers finite impulse response filters and filter banks. The use of these filters
for hearing aids is considered. Ways to compensate for hearing loss and ways to increase loudness
using broadband amplification are considered. A schematic diagram of a method for digital
signal processing using a bank of filters, as well as a technique for synthesizing interpolation filters
with low computational complexity, is presented. Also, the application of the MATLAB system for the
synthesis of narrow-band non-recursive FIR filters, their design procedure, methodology and examples
are considered. Finite Impulse Response (FIR) filters and filter banks have specific properties
that guarantee stability. Therefore, they are popular in many applications such as communication
systems, audio signal processing, biomedical instruments, and so on. Unfortunately, due to the longer
wavelength, the cost of implementing an FIR filter is usually not higher than an infinite impulse response
(IIR) filter that meets the same requirements. It is well known that the length of an FIR filter is
inversely proportional to its transition bandwidth. Therefore, the disadvantage becomes acute when a
given filter has a narrow transition band. The main goal is to consider computationally efficient
methods for designing FIR filters and filter banks. The masking method (FRM) results in significant
savings in the number of multipliers. Next, a 16-band, low group delay, non-equal-spacing digital
FIR filter bank is considered. Overall latency is significantly reduced as a result of a new filter structure
that reduces the interpolation factor for prototype filters. Masking filter may be an interpolated
finite impulse response (IFIR) filter that helps reduce complexity. -
METHOD AND ALGORITHM FOR SYNTHESIS OF CONTROLLED CHEBYSHEV DIGITAL FILTERS OF THE FIRST KIND OF LOW FREQUENCIES BASED ON THE BILINEAR TRANSFORMATION METHOD
I.I. Turulin, S.M. Hussein2022-11-01Abstract ▼The article presents a technique for the synthesis of controlled digital recursive Chebyshev lowpass
filters of the first kind with an infinite impulse response. The frequency response of such filters has
ripples in the passband and is as flat as possible in the stopband. Controllability is understood as an
explicit dependence of the filter coefficients on the cutoff frequency. The technique is based on the bilinear
transformation of the transfer function of the analog low-pass filter prototype and the frequency
transformation of the amplitude-frequency characteristics of the obtained digital filter. The main idea ofthe technique is that for an analog prototype filter with a cutoff frequency of 1 rad/s, the parameters of
the transfer function of biquadratic or bilinear links, which have the dimension of frequency, will be
numerically equal to the correction factors for similar parameters of a controlled filter with an arbitrary
cutoff frequency. As an example, the synthesis of a digital Chebyshev filter of the first kind of the fifth
order is considered. In this article, the transfer function of an arbitrary order filter is represented as a
cascade connection of II order links if the filter is of an even order. In the case of an odd order greater
than one, one cascaded link of the first order is added. Despite the relative simplicity of the frequency
conversion, in its practical use for digital filters synthesized using computer-aided design of digital filters
(or using reference books containing calculated prototype low-pass filters for various approximations
of the frequency response of an ideal low-pass filter), a series arises non-trivial specific moments
that complicate the engineering use of this method of synthesizing controlled digital filters. Therefore, in
addition to the technique, a step-by-step algorithm has been developed that allows one to synthesize a
filter without knowing these moments. The algorithm is implemented in the Mathcad environment; as
an example, a digital recursive Chebyshev filter of the 1st kind of the 5th order is calculated.
The example shows the calculated coefficients of a digital controlled low-pass filter, which explicitly
depend on the cutoff frequency, the amplitude-frequency characteristics of this filter and its lowfrequency
prototype converted into a filter with the same cutoff frequency, the amplitude-frequency
characteristics are given in the same coordinates. Due to the good formalization of the algorithm, the
latter is suitable for the implementation of computer-aided design systems for controlled digital
Chebyshev low-pass filters of the first kind. -
ANALYSIS OF STABILITY AND FEATURES OF PRACTICAL IMPLEMENTATION OF HOGENAUER FILTERS AS RECURSIVE DIGITAL FILTERS WITH FINITE IMPULSE RESPONSE
I.Е. Moiseenko , S. P. Tarasov , I.I. Turulin37-462025-10-01Abstract ▼The article considers the issues of stability of cascade integrator-comb (CIC) filters used in digital signal processing, including decimation and interpolation. A brief review of modern publications on the architectural optimization of CIC filters is given. The main attention is paid to increasing the stability of filters to the overflow of the bit grid, analyzing their stability and the method of synthesis of recursive FIR filters (filters with a finite impulse response). For a better understanding of the nature of the stability of CIC filters, the paper presents mathematical calculations illustrating the features of the accumulation of the constant component for various block configurations. A change in the structure of the CIC filter is proposed, consisting in the permutation of the integrator and comb filter blocks. It is proved that such a change prevents the accumulation of the constant component of the signal in the integrators and, therefore, eliminates the overflow of the bit grid due to the accumulation of the constant component in the integrator. This approach is based on the property of linear filters, according to which changing the order of inclusion does not affect the transfer function. amplitude-frequency characteristic, but in the case of digital implementations it allows to significantly reduce the probability of overflow. The possibilities of hardware and software implementation of such structures are considered from the point of view of minimizing the loss of accuracy and increasing the reliability of digital signal processing systems. It is proposed to use integers or numbers with a fixed point to eliminate the accumulation of quantization errors. In addition, a program in Python was developed that implements a CIC filter taking into account the stability of the constant component in the input signal and the accurate execution of operations. The obtained results are compared with modern approaches presented in scientific research in recent years. The proposed solutions can be useful in developing digital filters for systems with limited computing resources and increased stability requirements.
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COMPARATIVE ANALYSIS OF TWO FILTERING METHODS TO ELIMINATE NOISE IN AN IMAGE OF DIFFERENT DEGREES OF NOISE
K.O. Sever, I.I. Turulin, D.A. Guzhva2021-08-11Abstract ▼In modern photography and video technology, any image in the process of its creation is
distorted by various types of noise. There are various types of noise, but in practice, impulsive and
Gaussian noise models are the most common. Attenuation of the effect of noise is achieved by filtering.
At the moment, there is no universal filter that suppresses noise data at various intens ities
of distortion. Therefore, an important aspect is to determine the field of application of each
type of filter when suppressing noise in the image and creating a filter, consisting of a combination
of different filtering methods for optimal image cleaning. The article presents a comparative
analysis of median filtering and Wiener filtering to eliminate impulse and Gaussian noise in
the image with different degrees of noise. For modeling, we used one image, separately distorted
by impulse and separately by Gaussian noise with pixel distortion probabilities from 1% to
99% inclusive. Filtration was performed with windows equal to 3x3 and 5x5. As a result, we
obtained numerical estimates of the image filtering quality based on the peak signal-to-noise
ratio (PSNR). On the basis of the data obtained, the application of the investigated filters, their
modifications, advantages and disadvantages were analyzed, as well as recommendations for
their use were given. As a result of a comparative analysis of the studied types of filtering for
noisy images, it was found that the median filter with a 3x3 window copes better with image
cleaning from low-intensity impulse noise and with a 5x5 window - with image cleaning with an
average noise intensity. Also, the median filter does a better job of filtering out Waussian noise
at its medium and high rms deviations. The Wiener filter with 3x3 and 5x5 windows better fi lters
Gaussian noise at small values of its root-mean-square deviation. Also, the Wiener filter
copes better with impulse noise with relatively high noise power. -
ANALYTICAL REDUNDANCY IN THE AUTOMATIC CONTROL SYSTEM OF AN AIRCRAFT TURBOJET BYPASS ENGINE BASED ON OPTIMAL OBSERVERS
А. А. Inozemtsev, А.S. Pleshivykh, I.N. Gribkov, А.N. Sazhenkov, N.G. Lamanova2022-05-26Abstract ▼An analytical redundancy in the automatic control system of a bypass turbojet engine
(ACS turbofan engine) based on optimal observers is proposed. This article is based on previously
obtained results in previous author's works and is a generalization and analysis of these results in
order to develop a methodology for improving the fault tolerance of ACS turbofan engines. This
method is based on the use of optimal observers: the Kalman filter and the Yazvinsky filter, consistent
with the mathematical model of the ACS turbofan engine. The analysis of the mathematical model of
the ACS was carried out using the least squares method in a moving window. The accuracy of identification
of the mathematical model and the required delay time are ensured by optimizing the width
of the moving window. Estimated with the help of optimal observers, the output vector of the ACS
turbofan engine includes the following parameters: the rotor speed of the low-pressure compressor
rotor nв, the rotor speed of the high-pressure compressor nk, the air pressure behind the highpressure
compressor PK, the gas temperature behind the low-pressure turbine TT. When modeling the
Kalman filter, a correlation analysis of the input signals was preliminarily carried out. The rationale
for the advantage of the adaptive Yazvinsky filter compared to the Kalman filter is given. The results
of mathematical modeling of the algorithmic method of reserving the measurement channel of the
ACS turbofan engine based on the data of flight tests of a bypass engine of the PS-90A type as part of
the main narrow-body aircraft TU-214 both in stationary and transient modes are presented. Statistical
analysis of errors in estimation of the output vector of ACS turbofan engines based on the Kalman
and Yazvinsky filter has been carried out. It is shown that the proposed analytical redundancy algorithm
ensures the fulfillment of the requirements for the accuracy and stability of estimates of the
output vector of ACS turbofan engines when using the Yazvinsky filter and can be recommended for
use in advanced ACS turbofan engines. Based on the results proposed redundancy method, a direction
for further research has been formed. -
EMULATION OF A TECHNICAL VISION SYSTEM BASED ON VIRTUAL IR AND ULTRASONIC SENSORS FOR MOBILE ROBOT NAVIGATION
F.М. Tseeva , N.Е. Arabov , А.М. Bozieva , Z. V. Shomakhov2026-04-29Abstract ▼The relevance of this research is driven by the growing need for safe validation of navigation algorithms for autonomous mobile robots operating in cluttered and dynamically changing environments, where the use of physical equipment entails risks of damage and high costs. The aim of the work is to develop a rigorous methodology for software emulation of a technical vision system based on complementary virtual infrared and ultrasonic sensors. To achieve this aim, the following tasks were solved: formalization of the kinematic model of a differential drive with a state vector [x, y, θ]ᵀ; mathematical description of nonlinear triangulation for IR sensors and the physics of ultrasound propagation using the time-of-flight method d = c·t/2; integration of additive Gaussian noises with parameters σus = 0.005 m, σir = 0.002 m; implementation of heterogeneous data fusion using an Unscented Kalman Filter. The navigation controllers employed were the artificial potential field method with attractive and repulsive components, and fuzzy logic controllers. Experimental validation in a Python simulation environment of a maze with static obstacles demonstrated an average positioning error of 0.2 m in spherical configurations and an obstacle detection accuracy of 89.61%. The novelty of the proposed approach lies in providing a deterministic link between theoretical trajectory planning and physical implementation through the synergistic use of optical and acoustic sensory modalities. The practical significance of the work consists in a substantial reduction in the development and testing time for intelligent robotic systems, owing to the possibility of preliminary debugging of perception, data fusion, and navigation algorithms in controlled emulation conditions without the need for expensive hardware
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FORMATION OF THE IMPULSE RESPONSE OF A RECURSIVE LOW-PASS FILTER WITH FINITE IMPULSE RESPONSE AS A SUM OF QUASI-HARMONICS OF A TRUNCATED FOURIER SERIES
D.I. Bakshun , I.I. Turulin221-2282025-12-30Abstract ▼The problem of reducing the number of arithmetic operations in digital filtering algorithms is highly relevant, as it directly impacts power consumption, processing speed, and hardware costs. Under strict power efficiency requirements for mobile and embedded systems, minimizing multiplication and addition operations becomes a critical design factor. This paper presents a method for implementing a recursive filter with a finite impulse response (FIR) based on a truncated sinc function smoothed by a window (weighting function), represented as a sum of quasi-harmonic functions. These quasi-harmonic functions with different frequencies are polynomials of degree r. The study adopts a second-degree polynomial as a baseline and proposes a numerical method for increasing the polynomial order to improve the accuracy of the approximation. Accuracy analysis demonstrates that using 4th- and 6th-order polynomials achieves an approximation error of less than 1%. The coefficients of the non-recursive part of the filter are computed via inverse finite differences of the original FIR impulse response. These coefficients are integers whose values depend on the number of samples (length) of the half-period of the quasi-sinusoidal function, simplifying the implementation of such a recursive FIR (RFIR) filter on a field-programmable gate array (FPGA). Numerical analysis of finite differences for each quasi-sinusoid revealed that quadratic approximation requires only 16 samples but results in relatively high side-lobe levels (–30 dB). Switching to 4th-order approximation increases the number of non-zero coefficients to 20 and significantly reduces
(by 13 dB) the stopband magnitude of the frequency response, reaching –43 dB. -
MODELING OF TWO MICROSTRIP FILTERS OF THE CENTIMETER RANGE
A.N. Zikiy , P. N. Zlaman2021-01-19Abstract ▼Band-pass filters are an integral part of any radio equipment. They determine the selectivity of
the receiver for all channels of reception. The aim of this work is to modelling two microstrip filters
of the centimeter wave range. The object of study in this work are two microstrip filters at a frequency
of 5.75 GHz and 4.6 GHz. Such filters can be used in the converter of the centimeter wave range
as a signal and heterodyne filters. Two filters were simulated in the Microwave Office environment.
The results are presented in the form of models of two filters and four amplitude-frequency characteristics.
Given are the geometric dimensions of the filters, sufficient for their manufacture on material
RT5870 of Rogers company. Filters have a bandwidth of 200 MHz and a loss in bandwidth of not
more than 3 dB. Losses in the stop band for the signal filter were at least 45 dB, and at least 35 dB
for the heterodyne filter, which is a very good result for a two-pole filter. Acceptable electrical parameters,
small dimensions and moderate cost of manufacturing filters allows them to be widely used
in professional and amateur radio equipment. To improve the manufacturability of manufacturing, a
material with a low dielectric constant was selected. At the same time, gaps and tolerances on theaccuracy of their manufacture are acceptable. The design of the filters allows them to be easily
integrated with other components of the converter: a low-noise amplifier, a mixer, an intermediate
frequency amplifier, an amplifier in the local oscillator path. -
APPLICATION OF THE FOUR-POLE POINCARE-STEKLOV IN INTERFACE CONSTRUCTION FOR HARDWARE IN THE LOOP SIMULATION
M.N. Maksimov, S.M. Maksimova2022-01-31Abstract ▼The article considers the possibility of using the Poincare-Steklov filter to build an interface
for hardware in the loop (HIL) simulation of system. The Z and Y forms of the filter representation
are given. HIL simulation involves splitting the initial system into parts, with one part being modeled
numerically on a computer, and the second part is represented by a real physical object. The
parts of the system exchange data with each other through a hardware-software interface, which
can be implemented in different ways and should ensure stability, as well as convergence of the
results of HIL simulation to the results of modeling the original system. The variants of constructing
software and hardware interfaces ITM, TLM, TFA, PCD, DIM, GCS and the Poincare-Steklov
filter are described in the relevant literature sources.The article shows how the original nonlinear
system was divided into parts using the Poincare-Steklov filter, which, accordingly, led to the splitting
into parts of the system of equations describing the behavior of the original system. Next, it
was shown how the values of the stabilizing parameters of the Poincare-Steklov filter were calculated
and the systems of equations of the system divided into parts were corrected in accordance
with the obtained values. At the next stage, the article presents the results of numerical modeling
of the initial and partitioned system in MATLAB. When modeling in parts, the parts of the system
exchanged data with each other at each step of the simulation only once with a delay of h. This method of numerical modeling of a system divided into parts is as close as possible to the processes occurring
during semi-natural modeling of systems. A comparison of the obtained simulation results of
the initial and the system divided into parts allowed us to conclude that the Poincare-Steklov filter,
with the correct choice of the values of the stabilizing parameters, allows for the stability and convergence
of the results of semi-natural modeling of both linear and nonlinear systems. -
USING THE FOUR-POLE REPRESENTATION OF THE POINCARE-STEKLOV FILTER FOR HARDWARE IN THE LOOP SIMULATION OF NONLINEAR SYSTEMS
M.N. Maksimov, S.M. Maksimova2022-01-31Abstract ▼The article shows the possibility of using the Poincare-Steklov filter to ensure the stability of
harware in the loop (HIL) simulation of nonlinear systems.HIL simulation involves splitting the
initial system into parts, with one part being modeled numerically on a computer, and the second
part is represented by a real physical object. The parts of the system exchange data with each
other through a hardware-software interface, which can be implemented in different ways and
should ensure stability, as well as convergence of the results of HIL simulation to the results of
modeling the original system. The variants of constructing software and hardware interfaces ITM,
TLM, TFA, PCD, DIM, GCS and the Poincare-Steklov filter are described in the relevant literature
sources. The article shows how the original nonlinear system was divided into parts using the
Poincare-Steklov filter, which, accordingly, led to the splitting into parts of the system of equations
describing the behavior of the original system. Next, it was shown how the values of the stabilizing
parameters of the Poincare-Steklov filter were calculated and the systems of equations of the system
divided into parts were corrected in accordance with the obtained values. At the next stage,
the article presents the results of numerical modeling of the initial and partitioned system in
MATLAB. When modeling in parts, the parts of the system exchanged data with each other at each
step of the simulation only once with a delay of h. This method of numerical modeling of a system
divided into parts is as close as possible to the processes occurring during semi-natural modeling
of systems. A comparison of the obtained simulation results of the initial and the system divided
into parts allowed us to conclude that the Poincare-Steklov filter, with the correct choice of the
values of the stabilizing parameters, allows for the stability and convergence of the results of seminatural
modeling of both linear and nonlinear systems -
ANALYSIS OF MATLAB SYSTEM APPLICABILITY FOR SYNTHESIS OF CONTROLLED BUTTERWORTH DIGITAL RECURSIVE IIR FILTERS
Hussein Sh. Mogheer, I. I. Turulin2021-08-11Abstract ▼A number of digital signal processing applications use controlled IIR digital recursive filters.
The word " controlled " refers to filters whose structure ratios clearly depend on the cut rate or
boundary frequencies. Controlled IIR filters can be synthesized using a variety of tools to compute
traditional, uncontrolled IIR filters. The article dealt with the synthesis uncontrolled IIR filters and
analyzed the suitability of the presentation of the results of synthesis for the construction of controlled
IIR filters. The design techniques based on MATLAB (2021a) and the fundamental concepts of IIRButterworth
digital filters were described and explained. The composited signal was processed by the
analyzed filter to find whether it met the filtering progress criteria. To check the calculated filters, the
Simulink prototype was used, as well as FDA tool of signal processing toolbox. Based on the results
obtained, a conclusion was made about the applicability of the MAT LAB (2021a) system for the
synthesis of controlled digital recursive IIR-Butterworth filters. The analyzed technique was more
efficient, faster, decreased the tasks and found the results are satisfying. -
ANALYSIS OF CERTAIN WEIGHT FUNCTIONS (WINDOWS) AND THEIR APPROXIMATIONS FOR IMPLEMENTATION OF CONTROLLED RECURSIVE LOW-PASS FILTERS WITH A FINITE IMPULSE RESPONSE ON THEIR BASIS
T.V. Shushkevich , А.А. Morozov, I. I. Turulin2021-11-14Abstract ▼There are various types of weighting functions, the so-called windows in digital signal processing,
such as rectangular (Dirichlet window), triangular (Bartlett window), Vallee-Poussin window,
Kaiser-Bessel window, Barsilon-Temesh window, Hann, Bohman, Blackman, Gauss (Weierstrass),
Dolph - Chebyshev, Hamming windows and many others and ideal characteristics of standard filters
such as low-pass, high-pass, bandpass filters. The purpose of this review article is to determine the most
suitable weighting function for implementation on its basis of a controlled recursive low-pass filter with
a finite impulse response. This article presents an analysis of only some of the above windows and their
approximations, namely the Dolph - Chebyshev window, the Gauss (Weierstrass) window and the
Hamming window. In addition to the analysis, the synthesis of recursive filters with a finite impulse
response for weighting data based on the selected windows and their approximations was considered.
The method of synthesis of Dolph-Chebyshev windows is considered. The implementation of the Gauss
(Weierstrass) window is considered. Methods for approximating the Hamming window and methods
and several algorithms for developing filters with FIR in the form of this window are considered. The
estimation of parameters dependencies some quick window of the maximum level of the side lobes.
Based on the data obtained, conclusions were drawn about the selection of the most suitable and
demonstrating maximum performance windows, suitable for implementation on its basis of a controlled
recursive low-pass filter with a finite impulse response. -
EVALUATION OF THE STATE OF DYNAMIC WEIGHING BY THE KALMAN FILTER METHOD
Е.V. Zargaryan, Y.A. Zargaryan, А. Y. Nomerchuk2022-05-26Abstract ▼Currently, due to widespread computerization, the development of automated control systems
is relevant. Due to the development of small businesses, the purchase of commercially available
systems is a very expensive solution. It is possible to create similar control systems based on
inexpensive microprocessor kits (in this particular case, the K1816VE35 microprocessor kit is
used). In the future, such a system will not be difficult to improve, and it is also easy to implementinterfacing with various electronic computers (control from a personal computer). A system for
measuring and regulating bulk raw materials (an automated weighing system) is to be developed,
which provides control of the pneumatic transport automation with a 2-speed rotary dispenser,
through which bulk raw materials are fed to a weighing hopper suspended on a load-bearing device.
Measuring the weight of the bulk mass in the hopper of the scales, followed by automatic
control of unloading of bulk raw materials from the hopper. The profitability of any industrial
operation involving the weighing of raw materials, work in progress and finished products directly
depends on the accuracy of the weight data. However, even when using high-precision weighing
equipment, the method of collecting, recording and processing weight data for the micro ingredients
system may be subject to errors and inaccuracies. This can cause a potential revenue drain
that is difficult to detect and verify. In many cases, it is assumed that the cause of the problem is
related to the weighing equipment, whereas in fact it is related to the traditional data collection
and management system. In many factories where bulk products are mixed in batches, dosing
scales is a manual, time-consuming operation in which the ingredients are weighed separately
before loading into a blender or other technological container. A significant number of such plants
can benefit from the installation of an automated weighing and dosing system. -
PARTICLE FILTER BASED DETECTION OF DYNAMIC OBJECTS ON AN ACCUMULATED OCCUPANCY MAP
I.О. Shepel2022-08-09Abstract ▼The paper considers the problem of detecting dynamic obstacles on the accumulated occupancy
map generated by the computer vision system of a mobile robot. The purpose of this research is to
improve the quality of the obstacle detection algorithm by adding a particle filter to find moving objects
from the map data. In the paper, the problem of correct accumulation of data in the occupancy
map and reducing the delay in updating the map cells in which the object moves is solved. The modification
of the particle filter presented in the paper is able to work correctly with dynamic obstacles
in a wide range of speeds; it is resistant to outliers caused by random generation of the initial particles
velocities, and is workable under real conditions in real time in an environment with a lot of
moving objects. A heuristic has been created that reduces the number of misclassifications in occluded
areas. It is shown that the algorithm for detecting dynamic objects in the map is invariant to the
type of sensors used in the vision system, and an implementation combined with an accumulated
occupancy map is described. The algorithm is implemented and tested on board an autonomous mobile
robot, as well as on an open dataset. The article also provides a comparison with other approaches
of dynamic obstacles detection, as well as calculated performance metrics for all analyzed
methods for computers based on the GPU Nvidia RTX 3070 and Jetson AGX Xavier. Promising directions
for further research to improve the presented algorithm are formulated. -
REDUCTION APPROACH OF TRANSIENT PROCESS OVERSHOOT IN CONTROLLED TUNABLE LOW PASS DIGITAL FILTER
Al-Karawi Hussein Shakor Mogheer, I.I. Turulin2022-08-09Abstract ▼Digital signal processing is widely used in modern technology, including robotics, medical
technology, etc. Thus, the controlled digital filters are used to eliminate the constant component of
the output signal at the output of an analog-to-digital converter. This also reduces the low-frequency
interference level spread out on the frequency axis to the left of the lower boundary of the signal
spectrum. In actual situations, signals are subjected to a variety of disruptions and noises; however,
applying a filter may suppress these noises and produce a clean signal. Controllability means the
explicit dependence of the filter coefficients on the cutoff frequency. A transient occurrence can arise
in a digital filter, which is indicated in the overshoot of the signal. A change in the cutoff frequency
during filtering operation could cause this transient event. In this report, a Butterworth LPF filter is
used to offer a compensation strategy for reducing this overshoot. A transient is an overshoot (drive)
in the result timing chart. This drive is an after effect of the adjustment of the coefficients (boundaries)
of the filter during filtering (this is classified "on-the-fly tuning"). By using the MATLAB program,
the transient process resulting from the restructuring of the filter was investigated and the
formulas were checked compensation of this transient process. It was found that the application of
such compensation reduced the negative effects of the transition process. This decrease depends on
the order of the filter, the adjustment coefficient (the ratio of cut frequencies before and after adjustment)
and the moment of adjustment (for the periodic signal). -
PRINTED LATTICES APPLICATION TO ELECTROMAGNETIC WAVES POLARIZATION CONTROL AT MICROWAVES
А.О. Kasyanov38-602025-07-31Abstract ▼The paper is devoted to consideration of variants of solution to some problems of diffraction
for printed arrays with anisotropic inhomogeneities. The rotation of the plane of polarization is
realized by means of a twist-reflector. On the basis of computational and full-scale experiments it
was shown that twist reflectors based on microstrip reflective type antenna arrays (RAA) can be
quite broadband. The obvious advantage of such twist-reflectors is simplicity of the technology of
their manufacture, small thickness and weight. In addition, in twist-reflectors based on printed
RAA, it is possible to constructively realize the necessary phase distributions of the field on their
surfaces. Thus, it becomes possible to use such twist-reflectors to create phase-correcting twistreflectors.
It is shown that introduction of impedance loads in the construction of the re-radiating
elements of the printed RAA opens additional possibilities for controlling the field scattered by it.
Thus, twist reflectors with improved characteristics can be realized on the basis of microstrip-pin
RAA. The results of numerical studies based on the author algorithms and computational programs
are presented. Besides, spatial polarizing filters and a controllable twist-reflectors based
on microstrip RAA are considered. If microstrip elements of a twist-reflector contain controllable
loads, then qualitatively new type of the microwave device is formed. This device enables to control
polarization of the scattered field. Such devices can be named by polarizing modulators or
manipulators. By means of printed RAA with elements of complex topology the spatial polarizers,
which can transform electromagnetic waves with linear polarization in waves with rotating polarization
and, in particular, in waves with circular polarization, are realized








