Implementation Of Digital Filters Research Articles

A sampled-data feedforward programmable time-mode comb filter

In this work, a sampled-data feedforward comb filter suitable for time-mode signal processing (TMSP) is proposed. In the proposed system the input voltage is sampled and converted to a pulse-width (PWM); the pulse width is proportional to the input amplitude. The pulse width is the quantity which is being processed by the time-mode feedforward comb filter. Both input and output are synchronous with the sampling clock. The benefits of the proposed filter topology are (a) the modular and (b) hierarchical design which can be dealt with as a pure digital comb filter implementation counterpart. The frequency response of the proposed filter is digitally programmable. As a result, this filter can be used as building block for the implementation of higher-order time-mode systems. The comb filter was designed and verified through post-layout simulations in TSMC 65 nm technology process. The sampling rate was 5 MHz, the voltage supply was 1.2 V, consuming 540 μW for the use of 9-unit delay tap network and occupying a total area of 254 μm × 160 μm

Optimized FIR Filter Using Genetic Algorithms: A Case Study of ECG Signals Filter Optimization

The advancement in technology and the availability of specialized digital signal processing chips have made digital filter design and implementation more feasible in a variety of fields, including biomedical engineering. This paper makes two key contributions. First, it uses a genetic algorithm to optimize the coefficients of finite impulse response (FIR) filters. Second, it conducts a case study on using genetic algorithms to optimize FIR filters for electrocardiogram (ECG) biomedical signal noise removal. The goal of the proposed filter design approach is to achieve the desired signal bandwidth while minimizing the side lobe level and eliminating unwanted signals using a genetic algorithm. The results of a comprehensive analysis show that the genetic algorithm-based filter is more effective than conventional filter designs in terms of noise removal efficiency.

Series Motor Four Quadrants Drive DC Chopper Controllers Noise Cancellation

DC Series motor armature reaction causes EMI. EMI generates noise signals that affect feedback sensor readings, feedback controller function, and controller halting. This paper discusses software discrete filter and RLC hardware filter noise cancellation and filtering. The filter eliminates EMI-induced IGBT gate firing noise. Matlab/Simulink tests the filter circuit and digital filter control method, confirming software and hardware filter implementation.

Capricious Digital Filter Design and Implementation Using Baugh–Wooley Multiplier and Error Reduced Carry Prediction Approximate Adder for ECG Noise Removal Application

Capricious Digital Filter Design and Implementation Using Baugh–Wooley Multiplier and Error Reduced Carry Prediction Approximate Adder for ECG Noise Removal Application

IMPLEMENTATION OF QUANTUM ARITHMETIC OPERATIONS WITH INTEGER CHARACTERS USING THE QUANTUM FOURIER TRANSFORM

It has been proven that a quantum computer is superior to an electronic computer in solving some NP problems. Based on quantum operations, this article proposes a new quantum sum and quantum multiplication, and then the floating point quantum multiplier and quantum sum are created on the basis of fixed point number operations. These studies lay the foundation for the quantum implementation of digital filters. This article provides a new way to calculate the summator on a quantum computer. This method uses the quantum Fourier transform (QFT) and reduces the number of qubits needed to be added by eliminating the need to temporarily transfer bits. This approach also allows you to add a classical number to a quantum superposition without encoding a classical number in a quantum register. This method also allows for mass parallelization during its execution. Adding and multiplying capabilities based on QFT are improved with some changes. The proposed operations are compared with the operations of close quantum arithmetic. TRANSLATE with x English Arabic Hebrew Polish Bulgarian Hindi Portuguese Catalan Hmong Daw Romanian Chinese Simplified Hungarian Russian Chinese Traditional Indonesian Slovak Czech Italian Slovenian Danish Japanese Spanish Dutch Klingon Swedish English Korean Thai Estonian Latvian Turkish Finnish Lithuanian Ukrainian French Malay Urdu German Maltese Vietnamese Greek Norwegian Welsh Haitian Creole Persian   //   TRANSLATE with COPY THE URL BELOW Back EMBED THE SNIPPET BELOW IN YOUR SITE Enable collaborative features and customize widget: Bing Webmaster Portal Back //     Язык этой страницы: Английский   Перевести на Русский         Азербайджанский Албанский Амхарский Английский Арабский Армянский Африкаанс Бенгальский Бирманский Болгарский Валлийский Венгерский Вьетнамский Греческий Гуджарати Датский Иврит Индонезийский Исландский Испанский Итальянский Казахский Каннада Каталанский Китайский (традиционный) Китайский (упрощенный) Корейский Креольский (гаити) Курманджи Кхмерский Лаосский Латышский Литовский Малагасийский Малайский Малаялам Мальтийский Маори Маратхи Немецкий Непальский Нидерландский Норвежский Панджаби Персидский Польский Португальский Пушту Румынский Русский Самоанский Словацкий Словенский Тайский Тамильский Телугу Турецкий Украинский Урду Финский Французский Хинди Хорватский Чешский Шведский Эстонский Японский   Всегда переводить Английский на РусскийPRO Никогда не переводить Английский Никогда не переводить jpcsip.kaznu.kz

Kalman filter/smoother-based design and implementation of digital IIR filters

Recently, a unified framework was proposed for forward-backward filtering and penalized least-squares optimization. It was shown that forward-backward filtering can be presented as instances of penalized least-squares optimization. In other words, the output of a zero-phase digital infinite-impulse response (IIR) filter can be computed by solving a constrained optimization problem, in which the weight controlling the constraint is directly related to cutoff frequencies with closed-form equations. It was also shown that a zero-phase digital IIR filter can be formed as an optimal smoothing Wiener filter for a random process obtained from an autoregressive (AR) or AR-moving average (ARMA) model driven by input (innovation) noise in presence of an observation noise.In this paper, the problem of zero-phase digital IIR filtering is re-examined using Kalman filter/smoother. The paper shows that every zero-phase digital IIR filter can be viewed as a special case of an optimal smoothing Wiener filter. Based on the fact that the formulations of the optimum filter by Wiener and Kalman are equivalent in steady state, we present a Kalman filter/smoother framework to the design and implementation of digital IIR filters. As an example, the zero-phase digital Butterworth filter is designed using Kalman smoother and compared with the traditional design (forward filtering and backward smoothing) method.

DESIGN AND IMPLEMENTATION OF EFFICIENT DIGITAL FILTERS USING LABVIEW

In digital signal processing, the purpose of a filter is to remove unnecessary components in the signal, such as random noise. An infinite impulse response (IIR) filter is an iteration-type filter. The current pattern resulting from the IIR filter is based on the current input and output patterns. Several methods are available to detect digital IIR filters. Filters are widely used in signal processing and communication device approaches such as in channel equalization; noise reduction; radar systems; audio, speech signal, video, and biomedical signal processing, and economical, electrocardiogram, electroencephalogram, electromyography, and electrical circuit analyses. In this study, we used the Laboratory Virtual Instrument Engineering Workbench (LabVIEW) software program to analyze the design and implementation of an efficient digital filter, the results of which are presented in this paper. The important objective of this task was to design and examine different IIR filters-namely, Chebyshev, Butterworth, and Elliptic filters-using LabVIEW for the simulation processes. LabVIEW is used by graphical programming environmental engineers to implement automated research, certification, and product testing systems. In this work, the LabVIEW graphical programming tool was used because it gives accurate results in real time compared to other available simulation tools.

A Systematic Review on Distributed Arithmetic-Based Hardware Implementation of Adaptive Digital Filters

A Systematic Review on Distributed Arithmetic-Based Hardware Implementation of Adaptive Digital Filters

Design and Implementation of Communication Digital FIR Filter for Audio Signals on the FPGA Platform

Digital Signal Processing (DSP) in telecommunications is widely used for its outstanding performance. Hence, optimal results would be obtained with the sophisticated design and implementation of the digital Finite Impulse Response (FIR) filter. In this paper, the design and implementation of the FIR filter for processing the audio signal are obtained through the Field-Programmable Gate Array (FPGA) platform -Altera DE1 board. The design is divided into three primary blocks, which are the S2P Adapter block, Codec initialization block, as well as FIR filter block, which were then interfaced together; then, every block was compiled and simulated in order to obtain accurate results. The whole system was built to be functional, and finally, the frequency response was obtained. 100% significant accuracy and high quality are examined for implementing the filter in FPGA platform chip board programmed in (VHDL).

Implementation of Digital Filters for Real-Time PPG Signal Processing in VLC

Health care units depend on Wi-Fi or local area networks for biomedical data communication within their premises. The major hazard associated with biomedical data communication in healthcare units is radio frequency interference with medical devices. Visible light communication (VLC) is a developing technology that solves the issue of RF interference in healthcare units. Long-range communication is a great challenge in VLC as the transmitted data get corrupted by noise from ambient light and degrade with distance. Hence it is necessary to upgrade the signal condition and eradicate the noise due to ambient light interference. The proposed work addresses these issues and enables robust communication in healthcare units. The experimental validation of the proposed work has been executed by analyzing the VLC communication performance up to a distance of 5[Formula: see text]m. The proposed work focuses on implementation of digital filters like finite impulse response, infinite impulse response and least mean square adaptive filter for PPG signal processing. The least mean square filter showed a superior performance with a mean square error of 7.23e[Formula: see text] than the other filters, leading to an accurate diagnosis for patients without any RF interference inside healthcare units.

Design and implementation of two-dimensional digital finite impulse response filter using very high speed integrated circuit hardware description language

<span>The main purpose of this paper is to design a two-dimensional digital finite impulse response (FIR) filter using data broadcast and non-broadcast structure. The implementation of two-dimensional digital FIR filter is done using very high speed integrated circuit hardware description language (VHDL). Rectangular window method is used for calculating 2D digital FIR filter coefficients for data broadcast and non-broadcast structure. The coefficients of the one-dimensional digital FIR filter are obtained using the MATLAB filter design and analysis (FDA) tool for two different cut-off frequencies and are multiplied to get the necessary coefficient for the two-dimensional FIR filter to be designed; the simulation is done on Artix-7 series field programmable gate array (FPGA), target device (xc7a35t-cpg236) using Vivadov.2015.2. The proposed design reduces the area utilization and the power consumption when compared with the existing literature. The experimental result shows that the power consumption is improved by 97% and there is an improvement of 24% in area utilization for the two-dimensional with and without data broadcast one dimensional FIR filter structures.</span>

A Review of Techniques for Optimization and Implementation of Digital Filters on FPGA

Abstract: The use of digital filters extends to various fields. Obtaining a response that is close to the desired response is a major goal of designing digital filters. Designing of filters involves providing the specification, determining coefficients and, then the realization of filter on hardware. This paper provides an insight into the algorithms used for determining optimized filter coefficients, the realization of the filter on hardware and, some methods to reduce the hardware usage by the filter Index Terms: FPGA, FIR, IIR, MSE.

Filter design for ECG signal processing

Digital filters, as one of the most essential tools for signal processing, are often used in ECG signal processing. The hardware implementation of conventional digital filters relies on a hybrid device composed of a digital multiplier, an adder, and a delay unit. However, it has characteristics of a large amount of calculation, complex structure, and high overhead. With the development of computer technology, software implementation of signal processing and filter designing has been increasingly used. As a typical physiological signal, an ECG signal is characterized by a weak signal, complex noise, broad frequency spectrum, and little data sampling. The ECG filter designed by software can effectively weaken the useless noise in ECG signals, helping medical personnel obtain relevant information on patients’ symptoms and realize the diagnosis of diseases.

Real-time Optimization and Implementation of Multi-rate Digital FIR Kernel Filter

One ideal performance of this design is in the areas of decimation where a decimation factor of 10, 45-order and pass band ripple of 1dB and interpolation of sampled rates where a sinusoidal signal input produced a ripple free output with interpolation factor of 10, 52-order and stopband attenuation of 60dB. Owing to the multiple samples of filter length of 200, the filter performed down sampling preceded with filtering as well as up sampling preceded with filtering, hence multi-rate filter by allowing a low threshold of frequency of to be passed, blocking a high threshold of and vice versa. There was resampled output increased to 150% preceded by filtering. The filter coefficients for low pass and high pass Digital FIR filter, through the least square regression method, parks McClellan Algorithm and window methods were employed for easy optimization. More so, there was creation of 2-4-5 filter channel banks through the 2nd-level convolution of their down sampling and up sampling filtering techniques during the multi-irate filtering to ensure the design of error-free Digital FIR Filter using MATLAB File editor(M-File) and tool boxes for writing the C-programming of the design. In the analysis, the mean and standard deviation of the low pass Digital FIR Filter output during decimation and interpolation are (0.26, 6.13) and (0.004,1.22) respectively.

Perceptual equivalence of the Liljencrants–Fant and linear-filter glottal flow models

Speech glottal flow has been predominantly described in the time-domain in past decades, the Liljencrants-Fant (LF) model being the most widely used in speech analysis and synthesis, despite its computational complexity. The causal/anti-causal linear model (LFCALM) was later introduced as a digital filter implementation of LF, a mixed-phase spectral model including both anti-causal and causal filters to model the vocal-fold open and closed phases, respectively. To further simplify computation, a causal linear model (LFLM) describes the glottal flow with a fully causal set of filters. After expressing these three models under a single analytic formulation, we assessed here their perceptual consistency, when driven by a single parameter Rd related to voice quality. All possible paired combinations of signals generated using six Rd levels for each model were presented to subjects who were asked whether the two signals in each pair differed. Model pairs LFLM-LFCALM were judged similar when sharing the same Rd value, and LF was considered the same as LFLM and LFCALM given a consistent shift in Rd. Overall, the similarity between these models encourages the use of the simpler and more computationally efficient models LFCALM and LFLM in speech synthesis applications.

LOW POWER DIGITAL FILTERING USING ADAPTIVE APPROXIMATE PROCESSING: FIR FILTER STRUCTURES

Techniques for reducing power consumption in digital circuits have become increasingly important because of the growing demand for portable multimedia devices. Digital filters, being ubiquitous in such devices, are a prime candidate for low power design. Algorithmic approaches to low power frequency-selective digital filtering which are based on the concepts of adaptive approximate processing have been developed and formalized by introducing the class of approximate filtering algorithms in which the order of a digital filter is dynamically varied to provide time-varying stopband attenuation in proportion to the time-varying signal-to-noise ratio (SNR) of the input signal, while maintaining a fixed SNR at the filter output. Since power consumption in digital filter implementations is proportional to the order of the filter, dynamically varying the filter order is a strategy which may be used to conserve power. In this paper we introduce a class of approximate filter structures using FIR digital filter constituent elements. These filter structures are explored and shown to be an important element in the characterization of approximate filtering algorithms.

Biologically inspired direction‐finding for short baseline

The optimal implementation for a biologically inspired coupling structure to overcome the limitation of short baseline direction-finding is determined. This approach is inspired by the Ormia ochracea, a parasitoid insect living in North America. It can locate the crickets’ call accurately with the very small distance between its ears far beyond the accuracy of an interferometer with the same baseline. This outstanding performance depends on the mechanical coupling in its auditory system. The first research focus is on the mechanism of the coupling structure, considering not only the amplification on phase difference but also the effect on output power, which leads to the performance improvement in comparison with the traditional method. Then, the biologically inspired coupling structure is optimised to achieve the best direction-finding performance in crickets’ sound frequency, reducing the estimation error by 75% when the signal incident at boresight. To implement the actual coupling structure with optimal direction-finding performance, both the analogue circuit and the digital filter implementation are discussed, and the latter attains the theoretical optimal performance. Finally, a direction-finding system prototype is carried out to verify the advantage of digitally implemented coupling structure, and the measurement result approximates the corresponding Cram e ́ r–Rao lower bound.

Design and Implementation of Third Order Low Pass Digital FIR Filter using Pipelining Retiming Technique

This paper presents design and implementation of 3rd order low pass digital FIR filter using pipelining retiming technique. Aim of this paper is to apply pipelining retiming technique on low pass digital FIR filter and compare with the existing second order retimed FIR filter and third order broadcast, non-broadcast low pass FIR filter. MATLAB FDA tool is used to calculate digital FIR filter coefficients. Hamming and Kaiser window methods are used to find out the coefficients of the FIR filter. Broadcast and non broadcast third order low pass FIR filter architectures are used for pipelining retiming .Cutset and feed forward pipelining retiming techniques are used to retime the third order low pass digital FIR filter. We design our algorithm in VHDL and implemented on Xilinx Vivado xc7a35tcpg261-1. Area (LUT), speed and power are calculated using Xilinx Vivado 2015.2 tool. Synthesis and simulation results are discussed in this paper. Speed and area (LUT) of proposed third order low pass FIR filter is improved in comparison with existing designs.

High Performance Digital Filtering on Modified Multiply and Accumulate Unit in Residue Number System with Special Type of Moduli

The paper proposes the implementation of digital filtering using residue number system and the modified truncated multiply and accumulate unit. The work was carried out a theoretical analysis of digital filters using residue number system arithmetic and implemented hardware simulation on FPGA. FPGA hardware simulation results show that the use of residue number system allows to increase the frequency of digital filters up to about 4 times and hardware costs reduce up to 3 times compared to using a common positional number system. The obtained results open up the possibility for efficient hardware implementation of digital filters on modern devices (FPGA, ASIC and etc.) to solve practical problems, such as noise reduction, amplification and suppression of the frequency spectrum, interpolation, decimation, equalization and many others.

Prediction of loudness metrics at very high sampling frequencies using digital filters

Numerical solution of the augmented Burgers equation is generally the widely used process to predict sonic boom signatures on the ground starting from disturbances generated by an aircraft flying at supersonic speeds. Once the ground waveforms are predicted, loudness metrics are calculated as a means to quantify the impact associated with the underlying sonic booms. This work uses sBOOM, an augmented Burgers equation solver. Within sBOOM, loudness metrics are determined using time-domain digital filters rather than relying on Fast-Fourier Transforms (FFT) and their inverse counterparts to transform back and forth from the frequency-domain. For propagating signals from aircraft concepts that produce extremely low loudness levels, large sampling frequencies may be needed to achieve loudness metric convergence. This work discusses digital filters used within sBOOM, how their accuracy deteriorates at large sampling frequencies, and the enhancements to the digital filter implementation. Additionally, this work will also discuss loudness metric gradients (helpful in aircraft design exercises) obtained using the digital filtering enhancements. Finally, results with and without these modifications, both in terms of loudness metrics and their gradients, will be discussed and presented for cases which may include shaped low-boom as well as strong shock signatures.