A method for digital signal processing based laser-Doppler flowmetry

Methods of digital signal processing in control and monitoring systems are relevant today and largely replace the classic analog methods. This article discusses the main methods of digital signal processing and synthesis of appropriate devices, primarily digital filters.

The field of advanced digital signal processing methods is one of the fastest developing scientific and technical disciplines, and is important in the field of Shunt Active Power Filter control methods. Shunt active power filters are highly desirable to minimize losses due to the increase in the number of nonlinear loads (deformed power). Currently, there is rapid development in new adaptive, non-adaptive, and especially hybrid methods of digital signal processing. Nowadays, modern methods of digital signal processing maintain a key role in research and industrial applications. Many of the best practices that have been used to control shunt active power in industrial practice for decades are now being surpassed in favor of new progressive approaches. This systematic research review classifies the importance of using advanced signal processing methods in the field of shunt active power filter control methods and summarizes the extant harmonic extraction methods, from the conventional approach to new progressive methods using genetic algorithms, artificial intelligence, and machine learning. Synchronization techniques are described and compared as well.

Introduction. Currently the most widely used digital signal processing methods are those using specialized digital computers, which utilize analog-to-digital conversion of not only the input signals but also the weighting coefficients of the digital filters. This conversion results in quantization noise in the input signals, which reduces the efficiency of the specialized computer. The operating algorithms of these specialized computers are determined by the discrete transfer functions of the digital filters. The transfer function of a specific digital filter is determined by a set of weighting coefficients, which, like the input signals, are quantized by level and loaded into the computer memory. When quantizing the weighting coefficients, errors arise, the magnitude of which is determined by the quantization step. Weighting coefficient quantization errors inevitably affect the frequency properties of digital filters. Objective. To evaluate the influence of the weighting coefficient quantization step on the frequency characteristics of digital low-pass and high-pass filters, band-pass and notch filters of various orders. Method. The study utilized the bilinear z-transformation of the frequency transfer functions of continuous analog filters. Results. Transfer functions in the z-plane of digital low-pass and high-pass filters, band-pass and notch filters of various orders were obtained. Distortions in the frequency transfer functions of digital filters were calculated as a function of the weighting coefficient quantization step. Conclusions. Quantization of weighting coefficients leads to inevitable distortions in the frequency transfer functions of digital filters, the extent of which is determined by the quantization step size. The influence of the quantization step on the resulting distortions in the frequency transfer functions of digital filters must be taken into account when designing digital computers, specifically when selecting the number of binary digits and the quantization step size of a specialized computer. The results of this work will be of great use to designers of digital signal processing devices.

A new, simple Digital Signal Processing (DSP) method for investigating dynamic behavior of Power Systems is considered, which combines Integrating Zero Crossing (IZC) and Quasisynchronous Sample Interpolation (QSI). It offers accurate (about 0.001%) and fast (30-40 ms) tracking, estimating and predicting the actual power-line frequency and its deviation, as well as spectral harmonic components of the actual power line voltages and currents with high accuracies (0.02%-0.05%), due to almost complete elimination of aliasing and leakage effects. The results of a computer simulation of this method and of many real-life experiments are given and explained. The proposed method (IZC, QSI) has been compared to other DSP methods used for Power System investigation and applied in Underfrequency Relay Protection (URP); Load Shedding and Restoration (LSR); Acquisition, Monitoring, and Control (AMC); Diagnostics and Quality Analysis (DQA); and in other industrial applications.

The issues related to the study of the possibilities of mathematical and software modeling of digital algorithms for discrete Fourier transform without algorithmic multiplication operations are considered. The relevance of the study is due to the need to reduce the computational complexity of digital signal processing algorithms. The purpose of the study is to formalize a numerical method for the multi–stage discrete Fourier transform of a digital complex signal. The methods of mathematical and software modeling of numerical methods of digital signal processing were used in the study. The results of the study showed and confirmed the possibility of modeling a numerical method of multi-stage discrete Fourier transform of a digital complex signal based on numerical methods of difference digital filtering with integer difference coefficients of various orders of difference and a numerical method of reducing direct calculations of complex functions to perform simple operations of addition and shift. The concept of a multi-stage discrete Fourier transform and the concept of this transformation are defined and described. Its mathematical model is proposed and the numerical method of such transformation is formalized. It is noted that it allows to reduce the computational complexity of the hardware and software implementation of digital signal processing through the use of numerical methods of differential digital filtering, directed enumeration and comparative analysis of acceptable sets of numerical values of integer coefficients, which provides the possibility of reducing the error of the numerical method of multi-stage discrete Fourier transform. To reduce the computational complexity of his algorithms for isolating the harmonic components of a digital complex signal, the principles of dividing its frequency spectrum into narrow-band spectral components, frequency shifting of these components and their low-frequency difference digital filtering without performing arithmetic multiplication operations are applied.

In this work, a board based on a Spartan-3 field-programmable gate array was designed as a hardware prototyping platform for development of a multichannel digital gamma spectrometer. The device is compatible with various detectors like high-purity germanium, NaI, and CsI detectors. Before implementing the hardware, the method for digital signal processing for gamma spectroscopy was developed. The aim of this paper is to introduce a robust tool suitable for optimizing the design of complicated systems. The characteristics of this method, such as its ability to implement adaptive shaping, are investigated. The optimum conditions for digital filtering were determined using MATLAB/Simulink. This scheme can be useful for commercial production. Simulation was used to examine each processing unit in the whole signal processing procedure including cusplike shaping. The proposed method can be used as a computer design tool for optimizing digital multichannel analyzers or digital nuclear spectrometers.

The task of programming concurrent systems is substantially more difficult than the task of programming sequential systems with respect to both correctness and efficiency. The tendency in development of embedded, DSP systems and processors are shifting to multi core and multiprocessor setups as well. The problem of easy concurrency and algorithm development is an important for embedded and DSP systems as well. The goal of this paper is to define and present a high level language that allows description and development of signal processing algorithms. With the usage of a domain specific language, we can create compact and easy to understand definition of algorithms. In the paper the authors present the advantages granted by DSL for DSP applications. The created definitions are hardware independent can be executed and functionally verified. Efficient code can be generated for various targets without porting. The design of the presented DSL allows code generation for multi-core targets in case of computing-intensive algorithms, code generation for multiple streams, threads. Code reuse is supported by merging, re-grouping, and splitting of algorithms and groups of algorithms.

In this work, we describe different configurations of optical fiber interferometers (types Michelson and Mach-Zehnder) used to measure velocities during dynamical material's behaviour studies. We detail the algorithms of processing developed and optimized to improve the performance of these interferometers especially in terms of time and frequency resolutions. Three methods of analysis of interferometric signals were studied. For Michelson interferometers, the time-frequency analysis of signals by Short-Time Fourier Transform (STFT) is compared to a time-frequency analysis by Continuous Wavelet Transform (CWT). The results have shown that the CWT was more suitable than the STFT for signals with low signal-to-noise, and low velocity and high acceleration areas. For Mach-Zehnder interferometers, the measurement is carried out by analyzing the phase shift between three interferometric signals (Triature processing). These three methods of digital signal processing were evaluated, their measurement uncertainties estimated, and their restrictions or operational limitations specified from experimental results performed on a pulsed power machine.

In today's information age, digital signal processing technology with a new form of technology has been applied to various fields of society, and with its own significant advantages, in each field has played a powerful role. In order to ensure the air quality of living and working environment, the primary task is to analyze the main pollutants in the air. In addition to traditional chemical monitoring methods, digital signal processing methods are more common to detect pollutants, but filtering technology is needed to overcome the noise interference in the detection process. In this paper, the digital filter technology in digital signal processing is described in detail, and the model of digital signal in gas detection principle is simply abstracted, and the main application of digital signal processing technology in gas detection is described at last. At the same time, the paper analyzes the application of digital signal processing technology in gas detection, hoping to provide some help to the application of this technology and the good detection and treatment of gas pollution. Finally, a signal processing device for differential gas detection and its gas detection system are introduced.

In 24-hour Non-invasive Blood Pressure (NIBP) measurement, most of the automatic devices use oscillometric method. Accurate measurement of BP at finger using an Oscillometric Waveform (OMW) remains a challenge, particularly when the waveform is collected in inflation, because of the bad signal-to-noise ratio. In the present work, we aimed at measuring Ambulatory Blood Pressure by extracting OMW signal at finger with a new method of linear inflation control and digital signal processing. With the PID control of the pump and real time signal processing, we implemented the blood pressure measurement system and new algorithm. For the validation of our suggested method, we compare the effect of our method and traditional method. Finally, the method is validated in a pilot study against Welch Allyn ProBP 3400 BP monitors.

The article considers an effective algorithm for group delay correction in the channels of multichannel information and measurement systems. The description of the method is carried out for the case of harmonics measuring for current and voltage in electrical networks. The algorithm is based on the linear interpolation method using a non-recursive digital filter. The filter coefficients are a discretized windowed sinc. After the description of the method, the results of effective band modeling of the interpolating filter are presented. In conclusion, recommendations are given to reduce the amount of processor memory in the implementation of the algorithm.

Randomization is nonlinear procedure for deliberate introduction of randomness or noise into an information sequence. Depending on the rule according to which the randomization is fulfilled, it can be casual or pseudo-random. In this paper we present the theoretical basis of the randomization procedure, and also methods of digital signal processing based on the randomization.

A correct definition of mode parameters is of great importance for examination of power quality indices under non-sinusoidal modes, these parameters being effective values of current, voltage, active and reactive power. A wavelet transform is a later method of digital signal processing. This method enables information about signal to be saved both in amplitude-frequency and time space. The technique of defining effective current values of non-sinusoidal transient mode is presented. This technique assumes the preliminary frequency decomposition of the examining signal using the packet wavelet transform. Modernization of existing mathematical models of instantaneous, active and reactive power calculation as well as distortion power under non-sinusoidal modes based on the packet wavelet transform is carried out. The comparison of simulation modeling results shows qualitative coincidence with analytical mode calculation.

The development of electric power systems has raised the requirements of relay protection — sensitivity, speed, and reliability of functioning—for all types of short circuits (SC). The raised requirements can be satisfied by using more complete damage information extracted during the traveling-wave stage of the transient process accompanying SC. The experience gained in working out directional traveling-wave relay protection is generalized and questions concerning the use of traveling-wave principles for developing smart relay protection networks with complex topology and devices making it possible to control power flows in ac lines are examined. An algorithm is proposed for tuning out external SC and determining the damaged phase by using modern methods of digital signal processing.

A software-based digital signal processing (DSP) method using the data system processor has been developed to demodulate the photoacoustic responses of a sample to the fundamental phase modulation frequency and its harmonic frequencies (up to the ninth harmonic) in step-scan FT-IR photoacoustic measurements, without the use of any additional hardware. The DSP algorithm and its sampling depth multiplexing advantages are compared to conventional hardware demodulation. Comparison of results from the DSP method to those from hardware demodulators are shown at both the phase modulation frequency and the harmonics, and application of the DSP method to step-scan photoacoustic measurements with phase modulation is discussed as it applies to obtaining depth profile information in heterogeneous materials.