Performance Of Digital Filter Research Articles

Implementation of Digital Filter Using Verilog HDL

Abstract: In the context of Very Large Scale Integration technology, where addition is essential, digital filters are essential elements. The performance of the Finite Impulse Response Filter is highly dependent on the speed of its multiplier unit, making itstand out among the others. In order to improve the effectiveness of the FIR filter, we suggest using a Wallace tree multiplier. This novel method offers improvements over conventional multipliers, with a decrease in latency being one of the main advantages. Significant improvements in latency are obtained by using Xilinx tools and implementing this filter design in Verilog HDL. The Wallace tree multiplier is perfect for FIR filter construction in low-voltage and low-power VLSI applications since it has benefits like higher operating frequency and reduced power consumption. This development might improve the effectiveness and performance of digital filters, especially in settings with limited resources, opening the way for more robust VLSI systems

Graphic tools for analysing the influence of noise and aperiodic components in the performance of digital filters: a case study

An accurate phasor estimation is an important task for a correct operation of control and protection devices. The presence of aperiodic components in electrical signals is a major problem for the correct performance of digital filters. In addition, their stability is heavily conditioned by the distortion resulting from the noise inherent to signal processing. Consequently, a complete study of the behaviour of any filtering algorithm (in the presence of noise and aperiodic components) must be carried out before its implementation. Usually, numerical performance indices are used to measure the accuracy of a proposed method. In order to ease the analysis and comparison of different digital filters, a new set of graphic tools is presented in this paper. To illustrate the application of these graphic tools, a case study is presented with three different filtering algorithms developed for protection relaying.

Исследование практических аспектов ОМП на ЛЭП сверхвысокого напряжения по векторным измерениям на базе RTDS

Application of Phasor Measurement Unit (PMU) is a modern technology in electric power systems with quite a few possible applications. Remote identification of fault location (FL) at extra-high voltage transmission lines is one of the possible applications of PMU. The number of research works dedicated to this issue is rather limited. Also, there is no in-depth analysis of the PMU digital filter performance in case of short circuit; some factors that affect the FL accuracy are not covered; and the number of methods to identify FL remotely is insufficient. The current research goal is to study all the above-mentioned issues. The research has been conducted using the methods of mathematical modeling of fault-initiated transients represented with a real-time digital simulator RTDS (including the RSCAD software). A total of 16 FL expressions both in Russian and foreign scientific papers have been analyzed. Monte-Carlo method has been used to assess the impact of various factors on the FL accuracy. The PMU functionality presented in the RTDS simulator has been used to obtain required synchrophasors. A typical timespan of faults at extra-high voltage transmission lines has been deduced (it turns out to be at least three cycles of the fundamental frequency component). Dynamic characteristics of both M and P class PMUs have been thoroughly examined. The authors have made a conclusion that PMU-based FL is as accurate as that provided by conventional FL units. The influence of some major factors on the FL results has been investigated as well. PMU-based FL is considered to be a promising application. At the same time, P filter class PMUs are primarily used. One is to avoid getting synchrophasor data of different class PMUs at the line terminals. The obtained results are meant to be introduced to improve the PMU-based FL accuracy, especially in the context of processing real PMU data.

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.

Performance analysis of multi-rate signal processing digital filters on FPGA

Multi-rate signal processing, an important part of the design of a digital frequency converter, is realized mainly based on interpolation and decimation, which match the sampling rate between the baseband and high-frequency processing side, especially in down conversion. However, the design of a digital filter is important for realizing multi-rate interpolation and decimation, which is highlighted in this paper. To analyze the digital filter performance in multi-rate signal processing, the ordinary finite impulse response (FIR) filter and more efficient digital filter are discussed respectively. The ordinary FIR filters use a Hamming window to design, while a more efficient digital filter includes a cascaded integrate comb (CIC) and half-band filter. Sampling rate transformation factor is 12 in this design, which is cascaded by three stages. Each stage corresponding to the conversion factor is 3, 2, and 2. Both of these design methods are implemented on the FPGA development board. The hardware resource occupancy and the error rate of the signal amplitude in decimation show that the efficient digital filter is superior to the digital filter designed by the Hamming window in the real-time processing.

Hankel Norm Performance of Interfered Fixed-Point State-Space Digital Filters with Quantization/Overflow Nonlinearities

This paper presents a new linear matrix inequality (LMI)-based approach for the Hankel norm performance of fixed-point state-space digital filters subjected to external interference or excitation of finite duration and operating under various concatenations of quantization and overflow nonlinearities. The approach can be used to check the attenuation of undesired memory effects of past excitations on future outputs in the digital filters and to test the asymptotic stability in the absence of external input. The convex optimization problem with LMI constraints is formulated to determine the minimum Hankel norm performance of the digital filter. Finally, numerical simulations are provided to illustrate the effectiveness of the proposed approach.

Hankel Norm Performance of Digital Filters Associated With Saturation

This brief proposes a new approach to the examination and reduction of the undesired memory effect of digital filters joined with saturation, which is known as system ringing. A new criterion is established to ascertain the Hankel norm performance of digital filters joined with saturation. Using this criterion, it is possible to check the reduction of undesired responses of the filters to past excitations. The asymptotic stability for the digital filters is ensured without external input under the proposed criterion. Three improved criteria for the Hankel norm performance of digital filters are also presented based on the diagonally dominant matrix approach. A numerical simulation shows the usefulness of the developed method.

DIGITAL FILTERS IMPLEMENTATION IN MICROPROCESSOR-BASED RELAY PROTECTION

This article presents the implementation of digital filters used in digital relay protection current measuring elements. Mathematical descriptions of the considered digital filters as well as the computer programs for their coefficients calculation are described. It has been shown that in order to reliable estimate the digital filter performance its input signals waveforms must be close to the actual secondary current waveform of the current transformer to which the digital protection with the estimated digital filter is connected. For these purposes in MatLab–Simulink dynamic simulation environment the power system and the current measuring element models were developed. Performed calculations allowed to reveal that the exponentially decaying DC component which in some cases contains in primary fault current drives the current transformer core into saturation even when its nominal parameters are not exceeded. This results in distortion of the current transformer secondary current which in this case contains higher and inter-harmonics. Moreover, such harmonic content is not completely taking into account during coefficients calculation of the considered digital filters what results in signal magnitude estimation inaccuracy. Comparison of the digital filters response to the above-mentioned input signals allowed to find out such digital filter implementations which enable signal magnitude estimation with a minimum error. Ways of filtering quality improvement concerned with the window functions are proposed. Thus, the joint usage of digital filter and Hamming window allows to achieve the zero value of the signal magnitude gain factor in high-frequency range and substantially suppress all spectral components above 100 Hz. The increasing of the signal magnitude settling time in this case can be reduced by choosing the most optimal parameters of the all components of the current measuring element.

DESIGN OF FIR FILTER USING LOOK UP TABLE AND MEMORY BASED APPROACH

The main objective of this Paper is to develop Finite Impulse Response (FIR) Filter using look-up table (LUT) and memory based processor. The configuration is to reduce memory size, logic gate count and improve the speed of operation. Digital filters are becoming ubiquitous in audio applications. As a result, good digital filter performance is important to audio system design. Digital filters differ from conventional analog filters by their use of finite precision to represent signals and coefficients and finite precision arithmetic to compute the filter response. In this Paper, FIR filter is implemented in Xilinx ise using VHDL language. VHDL coding for the FIR filter is implemented in this Paper and waveforms are observed through simulation.

Dynamic Simulation Control Algorithm Based on the Time Domain

In view of badly transplanting of analog filter and low cost performance of digital filter for the washing out signal methods used by dynamic simulator, this paper proposed a computer intelligent time domain method. We decompose signal with the computer intelligence in the time domain, and convert the signal into the corresponding movement form respectively, then get the final result by overlaying them. The experimental results show that this method not only can achieve the effect of the traditional methods, better portability and faster computation speed, but also can be achieved directly on general computers.

Quantitative Investigation of Digital Filters inElectrocardiogram with Simulated Noises

Electrocardiogram (ECG) signal plays a vital role in the primary diagnosis and monitoring of the health of heart. For the features extraction of the ECG signals such as R-peak, QRS complexes, T-waves etc., the significant noises have to be cancelled. The most significant noises corrupted the ECG signal are power line interference (50/60Hz), Electromyographic (EMG) noise due to motion artifacts, muscle contraction, baseline wanders due to respiration and perspiration, and instrumentation noise. Designing digital filters to suppress these noises sits in a quite important position for ECG signal processing and analysis. This paper presents the application of software digital filters in order to effectively eliminate these noises from the ECG signals. Several types of digital filters were designed and implemented along with their strengths and weaknesses. The quantitative properties of implemented digital filters were investigated with the ECG signals from MIT-BIH Arrhythmia Database as the test data. All the work was done with MATLAB ® . The noises were simulated and added to the test data. The performance of digital filters was described by the comparison of power spectra of the filtered noisy signal and the original database ECG recordings and the mean square error.

이진수의 최소 디지트 표현과 공통 부분식 소거법을 이용한 디지털 필터의 성능 개선에 관한 연구

Digital filters are indispensible element in digital signal processing area. The performance of digital filter based on adding and multiplying operation, such as computational speed and power consuming is determined by the orders and coefficients of filter which has on effect area of semiconductor chip when it is implemented by VLSI technology. In this research, in order to performance improvement of digital filter, we proposed the algorithm to speed-up the operation of digital filter associated with the minimum signed digit representation of binary number system and method to simplify the digital filter design associated with common subexpression elimination. The performance of proposed method is evaluated by the computational speed and design-simplicity by experimental implemented digital filter on FPGA.

Transmission Line Distance Protection Using Wavelet Transform Algorithm

The performance of digital filters for transmission line distance protection significantly influences the effectiveness of the relaying operation. This paper proposes a wavelet-based algorithm, which can be implemented in microprocessor-based protective relays. Simulation results are presented to illustrate the effectiveness of this new algorithm for transmission line distance protection.

MODIFIED METHOD FOR SIGNAL TO NOISE RATIO IMPROVEMENT USING DIGITAL FILTERS

Traditionally, digital filters were used for signal-to-noise ratio (SNR) improvement. This paper proposes a modified method to enhance and improve the digital filter performance. This method is based on the combination of coherent time averaging technique and digital filtering. This paper discusses the processing of single and multiple input sinusoidal signals with additive noise. By using this method in processing multiple signals, we can isolate the weak in-band signals from the strong out-of-band signals, which consequently improve the weak signal delectability. Cascaded digital filtering is applied using similar or different types of digital filters. A comparison study is provided for single stage and cascaded stages digital filters. It is shown that this modified method leads to better results in SNR improvement of weak signals than in the case of the direct application of digital filtering.

A nonrecursive digital filter for data transmission

High-speed digital data transmission systems often require special filter functions for spectral shaping with prescribed phase characteristics. One such filter is the cosine roll-off filter, which has applications in both serial and parallel data transmission systems. This filter is of special interest, because in such applications it is common to utilize two identical cosine roll-off filters in a cascade connection, i.e., one in the transmitter and one in the receiver. The composite response of the cascade connection has the raised-cosine roll-off characteristic, which yields low intersymbol interference. Because a linear phase characteristic is further required, the digital filter type chosen for this application is nonrecursive. The difference equation coefficients are determined from the coefficients of a Fourier series expansion of the magnitude-frequency characteristic in the form of a cosine series. The linear phase characteristic is the direct result of setting the sine terms of the Fourier series to zero. Two criteria have been selected as a measure of the digital filter performance: minimum stop-band attenuation and an RMS measure of intersymbol interference. Performance was calculated as a function of the number of terms in the cosine series used to approximate the desired function. Performance graphs, which summarize the results of this investigation, are included. The application of digital filtering techniques in the field of data transmission systems is shown to have practical value.