Digital Signal Processing Theory Research Articles

An Explicit Form of Ramp Function

In this paper, an analytical exact form of the ramp function is presented. This seminal function constitutes a fundamental concept of the digital signal processing theory and is also involved in many other areas of applied sciences and engineering. In particular, the ramp function is performed in a simple manner as the pointwise limit of a sequence of real and continuous functions with pointwise convergence. This limit is zero for strictly negative values of the real variable x, whereas it coincides with the independent variable x for strictly positive values of the variable x. Here, one may elucidate beforehand that the pointwise limit of a sequence of continuous functions can constitute a discontinuous function, on the condition that the convergence is not uniform. The novelty of this work, when compared to other research studies concerning analytical expressions of the ramp function, is that the proposed formula is not exhibited in terms of miscellaneous special functions, e.g., gamma function, biexponential function, or any other special functions, such as error function, hyperbolic function, orthogonal polynomials, etc. Hence, this formula may be much more practical, flexible, and useful in the computational procedures, which are inserted into digital signal processing techniques and other engineering practices.

Multi-Branch Spectral Channel Attention Network for Breast Cancer Histopathology Image Classification

Deep-learning-based breast cancer image diagnosis is currently a prominent and growingly popular area of research. Existing convolutional-neural-network-related methods mainly capture breast cancer image features based on spatial domain characteristics for classification. However, according to digital signal processing theory, texture images usually contain repeated patterns and structures, which appear as intense energy at specific frequencies in the frequency domain. Motivated by this, we make an attempt to explore a breast cancer histopathology classification application in the frequency domain and further propose a novel multi-branch spectral channel attention network, i.e., the MbsCANet. It expands the interaction of frequency domain attention mechanisms from a multi-branch perspective via combining the lowest frequency features with selected high frequency information from two-dimensional discrete cosine transform, thus preventing the loss of phase information and gaining richer context information for classification. We thoroughly evaluate and analyze the MbsCANet on the publicly accessible BreakHis breast cancer histopathology dataset. It respectively achieves the optimal image-level and patient-level classification results of 99.01% and 98.87%, averagely outperforming the spatial-domain-dominated models by a large margin, and visualization results also demonstrate the effectiveness of the MbsCANet for this medical image application.

Serial digital QAM-modem based on complex band-pass IIR filters with LF-prototypes of Bessel

The article is devoted to the development and implementation of algorithms for the single-channel sequential quadrature amplitude modulation and demodulation based on digital complex band-pass IIR filters with LF-prototypes of Bessel. The use of complex IIR filters of Bessel allows to obtain orthogonal carrier signals which are close in shape to symmetric signals with practical non-overlapping spectra. The procedure for calculating the 4th-order digital low-pass Bessel filter using the generalized bilinear transformation method, which is used in the theory of digital signal processing to convert a continuous-time system into a discrete-time system, is considered. The variants of implementing the digital complex bandpass filters with a sequential structure using the complex delay method are considered, which allows to obtain the block diagrams of complex bandpass filters without additional calculations and to realize the restructuring of the central frequency without changing the frequency response shape by changing only two coefficients in complex delays. The procedure for obtaining of the expression of the impulse response of digital complex band-pass IIR filters of Bessel is considered. With the help of circuit modeling, the amplitude-frequency response and impulse response of such filter are determined. If the IIR filters are used in the modulator, the pulse characteristics of which are infinite and asymmetric, then it is necessary to use digital complex FIR filters consistent with the truncated pulse characteristic in the demodulator. But as an exception, the Bessel digital IIR filter has a phase-frequency response similar in shape closely to the linear one and, accordingly, a pulse response close to the symmetric one. In this case, the same recursive complex Bessel filters, similar to the filter in the modulator, can be used as matched filters in the demodulator. Therefore, the serial block diagram of the modem contains the same complex band-pass IIR filters with LF-prototypes of Bessel. The choice of the central frequency of the channel digital complex bandpass IIR filter is carried out by setting two coefficients. The results of circuit modeling of the data transmission system with the quadrature amplitude modulation (QAM-16) in the Microcap environment are presented, confirming the operability of the proposed option.

Electronic Auscultation System: Processing of Auscultatory Data

The purpose of research is to increase the diversity of electronic auscultation system designs with measured characteristics and testing. The series of articles includes the development of a model of the electronic auscultation system, the design of an electronic stethoscope, the manufacture of an experimental sample, the development of a methodology for measuring the amplitude-frequency characteristics of electronic and classical stethoscopes, testing of the proposed models and methods, analysis of auscultative data. The article considers mathematical methods of primary analysis of auscultative data. Recordings of normal and hard breathing were used for the experiment. Frequency, time-frequency and autocorrelation analysis of respiratory sounds was performed. Methods. The research was based on the theory of digital signal processing. The study uses auscultative data obtained from an experimental sample of an electronic auscultation system. Electronic records of hard (pathological) and normal human breathing over the left middle lung were used. Respiratory noises were converted into digital form with the following parameters: sampling frequency fd = 48 kHz; bit depth n = 24 bits; number of channels 1. To analyze the frequency information content of the recordings, their spectra were constructed. A fast Fourier transform was used to calculate the values in the spectra. Results. In the presented work, the frequency characteristics of recordings of hard and normal breathing are analyzed. The values of autocorrelation functions are obtained. An author-regression model of the process of normal breathing is obtained. The problem of analytical determination of the model order remains open and requires a separate solution. The resulting model allows you to generate an equivalent breathing noise of inhalation or exhalation when a signal with a uniform is applied to its input. Conclusion. A sample of the electronic auscultation system has been tested, methods of simplified primary analysis of auscultative data have been proposed. The difference in the results of the treatment of hard and normal breathing has no statistical significance due to the small sample.

Synthesis of a Generalized Algorithm for Processing and Generating Data on Reflected Signals from Complex Targets

Introduction. The quality of input information for trajectory processing (TP) systems can be improved by increasing the measurement accuracy of radar sensors (RS). However, in such a case, radar targets acquire the characteristics of complex targets having several marks at the output of the detector. This makes it difficult to accurately assess the kinetic parameters of targets in a TP system. In this respect, the development of a generalized algorithm for processing and generating data from the reflected signals of complex targets seems a relevant research task.Aim. To investigate reasons for the formation of complex targets and, using the theory of radar image processing, to synthesize an algorithm for processing and generating data on reflected signals from a complex target.Materials and methods. The following methodological approaches were used: the theory of digital signal processing; applied theory of radar image processing; MATLAB Simulink Toolboxes for simulating radar image processing; some prerequisites for fuzzy clustering methods.Results. Following an analysis of some characteristics of complex targets and the theory of radar image processing, an generalized algorithm was synthesized for processing and generating data of reflected signals from this class of targets. The results can be used to improve the measurement accuracy of their representative point when solving the TP problem.Conclusion. Reasons for the formation of complex targets in radar technology were analyzed. Their specific features consist in the need to accurately assess a true mark. A generalized algorithm for processing and generating these signals reflected from complex targets was proposed. The results can serve as a basis for solving the TP problem.

Frequency learning attention networks based on deep learning for automatic modulation classification in wireless communication

Deep neural networks have been recently applied in automatic modulation classification task and achieved remarkable success. However, Existing neural networks mainly focus on the purely data-driven architecture design, and fail to explore the hand-crafted feature mechanisms which are particularly significant for radio signal presentation in wireless communication. Inspired by digital signal processing theories, we propose frequency learning attention networks (FLANs) to analyze the radio spectral bias from frequency perspective, based on a multi-spectral attention mechanism for learning-based frequency components selection. FLANs are the general case of classical global average pooling and leverage identical structures of the popular neural networks. Extensive experiments have been conducted to validate the superiority of FLANs for automatic modulation classification over a wide variety of state-of-the-art methods on RADIOML 2018.01A dataset.

Pronunciation Tutor for Deaf Children based on ASR

ASR, whose full name is Automated Speech Recognition, is a technology that converts human speech into text. Speech recognition, a multidisciplinary field, is closely related to acoustics, phonetics, linguistics, digital signal processing theory, information theory, computer science and other disciplines. ASR has been applied in educational technology such as deaf children's education in this day and age. This paper makes a preview of a project in which a computer-aided tutor for deaf children instruction based on the speech recognition technology. This tutor utilizes three effective models and is combined with data mining technology. Two evaluation approaches and overview of embedded experiment are also detailed in this paper.

A high-precision harmonic distortion measurement algorithm and its system design based on time-domain matrix operations

In this paper, a harmonic distortion measurement algorithm and its system design based on time-domain matrix operations are proposed, which can quickly calculate the harmonic amplitudes, phases and total harmonic distortion of a signal. To realize the calculation of harmonic signal parameters in the time domain, we proposed a method to convert the nonlinear function representing the signal into a linear function, constructed a matrix equation, and solved the equation by the row transformation method, which significantly reduced the complexity of the algorithm. Finally, based on the proposed algorithm, a system design was implemented by the digital signal processing theory, and the measurements showed that the system achieves an amplitude error of nearly 1% and a THD error of less than 0.3% when measuring the generated 10 kHz fundamental with 5-order harmonics. The system can be used to measure and calculate the amplitudes and THD of harmonic distortion signals with high precision.

Virtual Experimental Project Design in the Digital Signal Processing Course

This paper proposes the design and development of virtual experimental projects in the Digital Signal Processing course, using MATLAB, Proteus, and CCS platforms to develop a library of typical experimental cases for biomedical engineering majors and discusses the design process. Based on these typical cases, this paper explores the secondary design for innovative engineering practice case teaching, which can promote students’ understanding and mastery of digital signal processing theories, algorithms, and technologies in an intuitive, flexible, and efficient way; quickly build new innovative engineering case models and further cultivate students’ engineering application ability as well as innovative thinking.

Improving the Structure of a Signal Used for Real-Time Calibrating of the Receiving Channels of Digital Transceiver Modules in Digital Phased Antenna Arrays

Introduction. Modern digital phased array antenna (DPAA) systems incorporate a large number of identical transceiver modules (TMs). These modules require real-time calibration with a high level of accuracy. In a previous work, we proposed a real-time calibration method for all receiver channels, which is based on the use of a calibration signal (CalSig) of the same frequency spectrum as the reflected signal and modulated in phase and amplitude by BPSK and OOK codes, respectively. This method was found to have a number of advantages over conventional approaches. However, the use of the same CalSig sample for all receiving channels increases the noise power gain at the output of a digital beam-forming unit (DBU). To overcome this limitation, we set out to improve the structure of CalSigs by making them pseudo-orthogonal. As a result, the noise power gain at the DBU output can be significantly reduced compared to that obtained in our previous work.Aim. To propose an improved design of a controlled amplitude modulation code OOK generator, which allows creation of pseudo-orthogonal CalSigs. As a result, the noise power gain at the output will increase insignificantly, thus having no negative effect on the quality of digital beam forming, signal processing and calibration.Materials and methods. Theory of system engineering and technology; theory of digital signal processing; system analysis; mathematical modeling.Results. An improved CalSig for calibrating the receiving channels of TMs was obtained. A structural diagram allowing the formation of pseudo-orthogonal CalSigs was synthesized.Conclusions. We proposed a new approach to improving the structure of signals used for real-time calibrating the DPAA receiving channels. A structural diagram of an amplitude-modulated OOK code generator for pseudo-orthogonal CalSigs was developed.

Shared Operation of a Barrage Jammer and Reference Signal Sources for Determining Ground-Based Radio Transmitters Operating via Geostationary Relay Satellites

Introduction. There are incidences of jamming the users of satellite communication systems, who apply e.g. geostationary relay satellites, and the illegal use of such satellite resources. These actions can be both intentional and unintentional, and, among other things, be caused by non-compliance with electromagnetic compatibility standards on part of other users of satellite communication systems. For a prompt and high -quality response of radio monitoring services and satellite operators to these illegal actions, it seems urgent to develop methods for accurate determination of the geolocation of radio emission sources.Aim. To develop a method for improving the accuracy of determining the coordinates of ground-based radio emission sources operating via geostationary relay satellites based on shared operation of a barrage jammer and reference signal sources.Materials and methods. The research was conducted using the statistical theory of radio engineering systems, the theory of digital signal processing and the method of simulation.Results. A method was developed for improving the geolocation accuracy of ground-based radio emission sources operating via geostationary relay satellites based on shared operation of a barrage jammer and reference signal sources. A method for resolving ambiguity regarding the true correlation peak of a reference source signal using a signal from a barrage jammer was described. An expression was obtained for the probability of a correct solution when resolving such ambiguity. As a result, the estimates of geolocation accuracy obtained using the developed method were compared with those obtained by a conventional method relying on the usage of 3 different reference stations.Conclusions. The method proposed in this paper makes it possible to achieve a relatively high accuracy when determining the geolocation of ground-based radio emission sources in the Earth’s regions of interest, at the same time as involving no organizational and financial costs for the installation of a large number of reference stations.

Учет фазовых искажений, вызванных движением геостационарных ретрансляторов, при местоопределении наземных источников радиоизлучения

Currently, there are often cases of illegal use of the resource of relay satellites located in geostationary orbit, and the creation of unintentional and deliberate interference with legal users of satellite communication systems, for example due to non-compliance with the power standards of radio transmitting devices and antenna radiation patterns, as well as the rules for the frequency spectrum regulating. One of the possible stages of the response by the radio frequency service and satellite systems operators to such situations may be an operational assessment of locating of the interference ground radio emission sources that violate the established requirements. The existing methods for estimating the coordinates of radio emission sources operating through geostationary satellites-repeaters involve calculating the Cross-Ambiguity Function (CAF) of signals received from several satellites that relay the signals of the main and side lobes of the antenna pattern of the geolocated source. In the case of a low received signals SNR, it is required to record signals for a long time, and in such cases, to achieve a sufficient SNR at the correlator output, it is necessary to take into account not only the Doppler frequency shift between the signals, but also the change in the frequency shift caused by the change in the velocity vectors of the repeater satellites. The aim of this work is to study the recording duration, at which it is required to take into account phase distortions caused by a change in the speed of the repeater satellite and their effect on the SNR at the correlator output, as well as to develop a method for accounting for such distortions. The theory of digital signal processing and the method of simulation were used as research methods. As a result of the study, an assessment was made of the duration of the signal recording, at which the Doppler frequency shift can be considered constant; introduced the concept of a modified CAF, which takes into account the change in the Doppler frequency shift due to its approximation by a linear function; the maximum duration of signal recording was estimated, at which the proposed linear approximation is valid. It is concluded that in the case of using the modified CAF, the minimum duration of signal recording, at which the absence of correlator output SNR degradation will be guaranteed, is 8.1 times greater than when using the traditional CAF.

Sliding Space-Frequency Processing of Finite Two-Dimensional Real Discrete Signals Based on the Fourier Transform with Variable Parameters

In the field of Fourier processing of finite signals, three main directions of scientific research have been identified: Fourier processing of one-dimensional finite signals - processing of scalar functions of a scalar argument, Fourier processing of two-dimensional finite signals - processing of scalar functions of a vector argument, multichannel Fourier processing - processing of vector functions of a scalar argument. As part of the creation of a generalized theory of Fourier processing of finite signals, the authors proposed: the theory of spectral analysis of discrete signals at finite intervals in the bases of parametric exponential functions and the theory of two-dimensional digital signal processing in Fourier bases with variable parameters. The developed theories, generalizing the theory of Fourier processing of one-dimensional and two-dimensional signals, are based: on the introduction of new concepts of the shift of finite discrete signals in one-dimensional and two-dimensional cases and the introduction of new basic Fourier processing systems of discrete signals, which have the properties of multiplicativity, functions in the system. The mathematical apparatus of two-dimensional discrete Fourier transform with variable parameters in matrix and algebraic form is considered. A new method for processing finite two-dimensional real discrete signals in the spatial-frequency domain based on the discrete Fourier transform with variable parameters, the method of sliding spatial-frequency processing, has been introduced. An efficient method and algorithm for fast diagonal sliding spatial-frequency processing of finite two-dimensional real discrete signals based on the discrete Fourier transform with variable parameters has been developed. The estimation of the efficiency and effectiveness of the algorithm of the diagonal sliding two-dimensional discrete Fourier transform with variable parameters from the point of view of computational costs is carried out. As a result of experimental studies on model two-dimensional discrete finite signals, the validity, efficiency and reliability of the proposed method of sliding spatial-frequency processing of finite two-dimensional real discrete signals based on the discrete Fourier transform with variable parameters have been proved. A comparison (from the point of view of computational costs) of the developed method of sliding spatial-frequency processing of finite two-dimensional real discrete signals based on the discrete Fourier transform with variable parameters with the standard method of sliding processing of this type of signals is carried out.

Accuracy of Positioning of Ground Sources Using Geostationary Repeaters

Introduction. Currently, there is a significant increase in the number of relay satellites in geostationary orbit. However, frequent incidents of illegal use of the satellites frequency resource, as well as unintentional and deliberate interference with other users are fixed. In this regard, it becomes necessary to evaluate accuracy and applicability of various methods for determining the location of sources of illegal and interfering radio emission with different signal parameters and with different levels of uncertainty for relay satellite coordinates and velocities.Aim. To study and to evaluate the accuracy of methods of geolocation of radio emission sources operating through geostationary relay satellites, with different signal parameters and with different levels of uncertainty for relay satellite coordinates and velocities.Materials and methods. Imitation modeling and the theory of digital signal processing were used.Results. Factors influencing the accuracy of the estimation of TDOA and FDOA parameters when determining the position of radio emission sources, which operate via relay satellites located in geostationary orbit, were considered. As a result of simulation, the estimate of the accuracy of the considered geolocation methods was obtained. It depends on the bandwidth of radio emission source signal, on the recording duration and on the level of a priori uncertainty relatively the relay satellites coordinates and velocities. Recommendations for the application of the considered methods in various conditions were formulated.Conclusions. Conclusions and recommendations formulated as a result of the study, will allow one to choose the most appropriate geolocation method to improve the accuracy of radio emission sources locating depending on conditions and signal parameters.

A Method of Increasing Digital Filter Performance Based on Truncated Multiply-Accumulate Units

This paper proposes new digital filter architecture based on a modified multiply-accumulate (MAC) unit architecture called truncated MAC (TMAC), with the aim of increasing the performance of digital filtering. This paper provides a theoretical analysis of the proposed TMAC units and their hardware simulation. Theoretical analysis demonstrated that replacing conventional MAC units with modified TMAC units, as the basis for the implementation of digital filters, can theoretically reduce the filtering time by 29.86%. Hardware simulation showed that TMAC units increased the performance of digital filters by up to 10.89% compared to digital filters using conventional MAC units, but were associated with increased hardware costs. The results of this research can be used in the theory of digital signal processing to solve practical problems such as noise reduction, amplification and suppression of the frequency spectrum, interpolation, decimation, equalization and many others.

A New Operation Method of Relay Protection Hardwarewith VFC Pulse Train as Variable

Introduction In the basic theory of digital signal processing， the calculus operation of continuous time variable function can be transformed into the difference and accumulation operation of corresponding discrete signal.At present， the application of this principle in microcomputer type relay protection is based on the software operation of fixed interval sampling value.In this paper， A new application type of the above theory is proposed， which integrates the sampling and operation process into the discrete， quantization and coding links of A/D conversion， so as to simplify the computational overhead and improve the protection performance. Methods The real-time sequence and delay sequence of VFC serial pulse signal were taken as variables， the counter was taken as the core computing hardware， and the single and multiple floating operation Windows were constructed with pulse delay devices to realize the hardware synchronization of calculus and phasor addition and subtraction.Based on the mature theory of software protection algorithm， the new algorithm formula is obtained by interval transformation and the loop construction mode is proposed. Result After discussion and calculation， the following results are obtained. The new method can be used to construct the instantaneous integral value of the electrical quantity， the instantaneous sampling value and the effective value of the definite integral. Conclusion Although the operation is based on the same basic principle as software operation， the physical process of operation is very simple， which can complete high-precision synchronous operation， and form a variety of operators， which can construct a protection scheme of multiple principles.

Advanced Digital Signal Processing Techniques on the Classification of the Heart Sound Signals

Speech processing subject primarily depends on the digital signal processing (DSP) methods, such as convolution, discrete Fourier transform (DFT), fast Fourier transforms (FFT), finite impulse response (FIR) and infinite impulse response (IIR) filters, FFT recursive and non-recursive digital filters, FFT processing, random signal theory, adaptive filters, upsampling and downsampling, etc. Recursive and non-recursive digital filters are primarily deployed to absorb the signal of interest signals and to block the unwanted signals (noise). Broadly, low-pass, high-pass, band-pass, and band-stop filters are implemented for filtering functions. In frequent, the DSP theories can be used for further biomedical engineering domains like biomedical imaging (MRI, ultrasound, CT, X-ray, PET) and genetic signal analysis-cum-processing too. In this article, the experiments such as voiced/unvoiced detection, formants estimation using FFT and spectrograms, pitch estimation and tracking and yes/no sound classification are used. Also, the analysis of normal/abnormal heart sound signals using simple energy computation and the zero-crossing rate and their results are obtained. For the entire study, the Matlab R2018a tool is used to obtain the simulation results. At last, the criticism, feedbacks, comments, reactions from the student are detailed for the exceptional development of the course.

Single channel pipelined variable‐length FFT processor design

In this study, a single channel pipelined variable-length fast Fourier transformation (FFT) processor design is presented. With the development of science and technology, digital signal processing has been widely used in many fields. The most basic algorithm of digital signal processing theory, FFT, is widely used in radar signal processing, remote sensing, image processing, communications and other fields. A high real-time and high-throughput FFT processor design method based on the field-programmable gate array (FPGA) platforms is proposed to meet the needs of the synthetic aperture radar imaging digital signal processing. The authors conducted the ISE synthesis based on the Xilinx FPGA platform using xc6vlx760ff1760-1 as the target chip and the ModelSim simulation, combined with Matlab, to verify the correctness of this design.

ПІДХОДИ ДО СТВОРЕННЯ ПРОГРАМНОГО АГЕНТА НА ОСНОВІ МЕТАЕВРИСТИК І ШТУЧНИХ НЕЙРОННИХ МЕРЕЖ

In this article, has been developed a software agent based on meta-heuristics and artificial neural networks. The analysis of existing classes of agents and the selected reactive agent with internal state, which is well suited for partially observable, dynamic and non-episodic media, was carried out, and this agent has an internal state that preserves the state of the environment, obtained on the basis of the history of acts of perception, in the form of structured data. Were proposed approaches to create an agent based on meta-heuristics and an agent based on an artificial neural network. The development of reactive agents with internal state, based on the PSO (particle swarm optimization) metaheuristics, which are related to individual particles and to a whole swarm and interact by messages was proposed. Also, has been proposed an approach to the creation of a reactive agent with an internal state based on the Elman recurrent neural network. The agent-based approach allows combining different areas of artificial intelligence, digital signal processing, mathematical modeling, and game theory. The proposed agents were implemented using the JADE (Java Agent Development Framework) toolkit, which is one of the most popular tools for the creation of agent systems. A numerical study was made to determine the parameters of the swarm PSO metaheuristics and the Elman recurrent neural network. As a purpose function, the Rastrigin test function has been used. The number of visits to the website of DonNTU was used as an input sample for the Elman network. The minimum average square error forecast was the criterion for choosing the structure of a network model. 10 hiding neurons were used to predict the number of visits to the website page, since, with increasing of hidden neurons number, the change in the error value is small. To determine the number of particles in the swarm, a series of experiments was conducted, the results of which are presented by graphs. The proposed approaches can be used in intelligent computer systems.

МЕТОД ПІДВИЩЕННЯ ЗАВАДОЗАХИСТУ ВИСОКОШВИДКІСНИХ КАНАЛІВ ПЕРЕДАЧІ ДАНИХ БЕЗЕКІПАЖНИХ МАЛОРОЗМІРНИХ СУДЕН

Protection of the underwater space within the territorial sea of Ukraine is one of the functions of the Armed Forces of Ukraine. Responsibility for the exercise of these functions lies directly with the Naval Forces of the Armed Forces of Ukraine. The illumination of the situation in the underwater space is one of the tasks for the Navy's ships, the successful solution of which contributes to the functions of ensuring the protection of the underwater space, increasing the protection of ships from underwater threats in the parking lots and during tasks in less dangerous areas. In order to perform these functions, the article analyzes the two-step method of additional gating of analogue-digital converter counters during I / Q-demodulation of OFDM signals, and more significantly different way of additional gating of digital signal counts in a "sliding window" mode is offered. The above proposals can be used as a basis for further scientific research on the development of the theory of digital signal processing, as well as in the production of digital segments of control systems and communication complexes of small scale surface vessels based on modern technologies.