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Journal articles on the topic 'Signal processing Electric filters'

Author: Grafiati

Published: 4 June 2021

Last updated: 2 February 2022

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1

Smolarik, Lukas, Dusan Mudroncik, and Lubos Ondriga. "ECG Signal Processing." Advanced Materials Research 749 (August 2013): 394–400. http://dx.doi.org/10.4028/www.scientific.net/amr.749.394.

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Electrocardiography (ECG) is a diagnostic method that allows sensing and record the electric activity of heart [. The measurement of electrical activity is used as a standard twelve-point system. At each of these leads to measure the useful signal and interference was measured. The intensity of interference depends on the artefacts (electrical lines, brum, motion artefacts, muscle, interference from the environment, etc.). For correct evaluation of measured signal there is a need to processing the measured signal to suitable form. At present, the use of electrocardiograms with sensors with contact scanning are difficult to set a time so we decided to use the principle of non-contact sensing. Such a device to measure the ECG was constructed under the project. The disadvantage of such devices is a problem with a high level of noise, which degrades a useful signal. The aim of this article is to pre-process the signals obtained from non-contact sensing. The contactless devices are powered from the network and battery. The electrodes were connected by way of Eithoven bipolar leads. Signals were pre-treated with suitable filters so that they are also appropriate for their subsequent analysis. In the filtration ECG signals was used as a method of linear (low pass filter, high pass, IIR (Infinite Impulse Response) peak, notch filter. The results of many signals clearly demonstrate removing noise in the ECG signals to the point that is also suitable for their analysis.

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Gadsden, S. A., M. Al-Shabi, and S. R. Habibi. "Estimation Strategies for the Condition Monitoring of a Battery System in a Hybrid Electric Vehicle." ISRN Signal Processing 2011 (April 13, 2011): 1–17. http://dx.doi.org/10.5402/2011/120351.

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This paper discusses the application of condition monitoring to a battery system used in a hybrid electric vehicle (HEV). Battery condition management systems (BCMSs) are employed to ensure the safe, efficient, and reliable operation of a battery, ultimately to guarantee the availability of electric power. This is critical for the case of the HEV to ensure greater overall energy efficiency and the availability of reliable electrical supply. This paper considers the use of state and parameter estimation techniques for the condition monitoring of batteries. A comparative study is presented in which the Kalman and the extended Kalman filters (KF/EKF), the particle filter (PF), the quadrature Kalman filter (QKF), and the smooth variable structure filter (SVSF) are used for battery condition monitoring. These comparisons are made based on estimation error, robustness, sensitivity to noise, and computational time.

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3

HAAVISTO, P., M. GABBOUJ, and Y. NEUVO. "MEDIAN BASED IDEMPOTENT FILTERS." Journal of Circuits, Systems and Computers 01, no. 02 (June 1991): 125–48. http://dx.doi.org/10.1142/s0218126691000021.

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Idempotent filters produce a root signal in a single filter pass, i.e. the filter output is invariant to further filterings with the same filter. In this paper median based idempotent filter structures are introduced. Two approaches to generate these filters are studied: weighted median filters and median filter cascades. Two subclasses of n-dimensional idempotent weighted median filters, called Class 1 and Class 2 filters in the paper, are introduced. It is shown that both Class 1 and Class 2 filters suppress impulsive noise from n-dimensional input signals and yet have almost no effect on the non-corrupted parts of the signal. These filters are therefore well-suited for example for preprocessing purposes. An application to speech processing is described. Other likely applications of these filters are in image processing and, also, in image sequence processing, where the filter mask is typically 3-dimensional. Sufficient conditions for a filter cascade to be idempotent are given. Two idempotent median filter cascades and their advantages are discussed.

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Vydmysh, Andriy, Oleksandr Voznyak, Igor Kupchuk, and Dmitry Boyko. "RESEARCH OF MEDIAN FILTERING OF ONE-DIMENSIONAL SIGNALS." Vibrations in engineering and technology, no. 1(96) (August 27, 2020): 88–102. http://dx.doi.org/10.37128/2306-8744-2020-1-10.

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This paper considers the principles of digital signal processing, the general provisions of digital filtering, existing methods of noise filtering in electrical signals, the median filters of one-dimensional signals are studied in detail. To solve this problem, the classification of digital signal processing tools is presented. Since the most effective for filtering noise in electrical signals are digital filters, they are given the most attention. The main purpose of signal filtering is the need to extract the information contained in them. This information, which is usually present in the amplitude of the signal (absolute or relative), in frequency or spectral composition, in phase or in the relative time dependences of several signals. The classification of existing digital filters is carried out. For further development, a median filter was selected, which belongs to the class of heuristics and is one of the most effective in filtering signals from impulse noise and white noise. Highlighting the advantages and disadvantages, a review of existing software that implements the median filter. It is established that the urgent task is to increase the processing speed and reduce resource costs in the implementation of such filters, developed an algorithm for fast median filtering, conducted an experimental test of software-implemented median filters with different apertures at different levels of fluctuation noise. This program meets all the requirements of modern norms and standards, allows its practical use to solve real problems of signal processing. In order to increase the speed of information processing, a median filtering algorithm based on difference matrices using the threshold saturation function has been developed. Developed software that implements the proposed algorithm. Schemes of the main program, reading of values of a signal from a file, filtering, sorting of data on amplitude, a choice of a window of elements, a choice of the registered values are presented. The conditions of data registration and ADC parameters to ensure efficient operation of filters are also defined

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5

Chhavi Saxena, Dr, Dr Avinash Sharma, Dr Rahul Srivastav, and Dr Hemant Kumar Gupta. "Denoising of Ecg Signals Using Fir & Iir Filter: a Performance Analysis." International Journal of Engineering & Technology 7, no. 4.12 (October 4, 2018): 1. http://dx.doi.org/10.14419/ijet.v7i4.12.20982.

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Electrocardiogram (ECG) signal is the electrical recording of coronary heart activity. It is a common routine and vital cardiac diagnostic tool in which in electric signals are measured and recorded to recognize the practical status of heart, but ECG signal can be distorted with noise as, numerous artifacts corrupt the unique ECG signal and decreases it quality. Consequently, there may be a need to eliminate such artifacts from the authentic signal and enhance its quality for better interpretation. ECG signals are very low frequency signals of approximately 0.5Hz-100Hz and digital filters are used as efficient approach for noise removal of such low frequency signals. Noise may be any interference because of movement artifacts or due to power device that are present wherein ECG has been taken. Consequently, ECG signal processing has emerged as a common and effective tool for research and clinical practices. This paper gives the comparative evaluation of FIR and IIR filters and their performances from the ECG signal for proper understanding and display of the ECG signal.

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6

Stewart, R. W., J. J. Soraghan, and T. S. Durrani. "Noncanonical FIR filters and adaptive signal processing." Electronics Letters 25, no. 6 (1989): 414. http://dx.doi.org/10.1049/el:19890285.

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7

Ruiz-Alzola, J., C. Alberola-López, and C. F. Westin. "Kriging filters for multidimensional signal processing." Signal Processing 85, no. 2 (February 2005): 413–39. http://dx.doi.org/10.1016/j.sigpro.2004.09.009.

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8

Terrell, T. J. "Book Review: Digital Filters and Signal Processing." International Journal of Electrical Engineering & Education 24, no. 1 (January 1987): 86. http://dx.doi.org/10.1177/002072098702400123.

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9

Havlicek, J. P., G. R. Katz, and J. C. McKeeman. "Even length median filters in optimal signal processing." Electronics Letters 28, no. 13 (1992): 1258. http://dx.doi.org/10.1049/el:19920795.

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10

Morency, Matthew W., and Geert Leus. "Graphon Filters: Graph Signal Processing in the Limit." IEEE Transactions on Signal Processing 69 (2021): 1740–54. http://dx.doi.org/10.1109/tsp.2021.3061575.

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11

Gaydecki, Patrick. "The Foundations of Digital Signal Processing Using Signal Wizard Systems®." International Journal of Electrical Engineering & Education 49, no. 3 (July 2012): 310–20. http://dx.doi.org/10.7227/ijeee.49.3.10.

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Signal Wizard Systems® is a digital signal processing (DSP) research venture within the School of EEE at the University of Manchester, UK. It specialises in the development and supply of real-time DSP products for audio signal analysis and processing. The unique and underpinning philosophy of these products is their ease of use. The systems require minimal knowledge of DSP theory on the part of the user and none of the mathematics associated with digital filter design. Filters and other algorithms can be designed in seconds, downloaded and executed in real time with just a few mouse clicks. Since 2004 Signal Wizard products have been sold all over the world for applications ranging from noise suppression, adaptive filtering and system modelling to musical instrument research. In particular, their ease of use ensures that they are ideally suited for teaching simple and more advanced concepts in DSP both at undergraduate and postgraduate level. For this purpose, a DSP laboratory teaching package has been developed using the Signal Wizard range of devices, and has proven an invaluable tool for training our student cohort in the practical aspects of DSP engineering design and programming.

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12

Siswanto, Antonius, Cheng-Yuan Chang, and Sen M. Kuo. "Multirate Audio-Integrated Feedback Active Noise Control Systems Using Decimated-Band Adaptive Filters for Reducing Narrowband Noises." Sensors 20, no. 22 (November 23, 2020): 6693. http://dx.doi.org/10.3390/s20226693.

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Audio-integrated feedback active noise control (AFANC) systems deliver wideband audio signals and cancel low frequency narrowband noises simultaneously. The conventional AFANC system uses single-rate processing with fullband adaptive active noise control (ANC) filter for generating anti-noise signal and fullband audio cancelation filter for audio-interference cancelation. The conventional system requires a high sampling rate for audio processing. Thus, the fullband adaptive filters require long filter lengths, resulting in high computational complexity and impracticality in real-time system. This paper proposes a multirate AFANC system using decimated-band adaptive filters (DAFs) to decrease the required filter lengths. The decimated-band adaptive ANC filter is updated by the proposed decimated filtered-X least mean square (FXLMS) algorithm, and the decimated-band audio cancelation filter can be obtained by the proposed on-line and off-line decimated secondary-path modeling algorithms. The computational complexity can be decreased significantly in the proposed AFANC system with good enough noise reduction and fast convergence speed, which were verified in the analysis and computer simulations. The proposed AFANC system was implemented for an active headrest system, and the real-time performances were tested in real-time experiments.

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13

de Queiroz, R. L., and P. A. Stein. "LUT Filters for Quantized Processing of Signals." IEEE Transactions on Signal Processing 52, no. 3 (March 2004): 687–93. http://dx.doi.org/10.1109/tsp.2003.822357.

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Belyaev, Alexander G., and Pierre-Alain Fayolle. "Two iterative methods for reverse image filtering." Signal, Image and Video Processing 15, no. 7 (April 8, 2021): 1565–73. http://dx.doi.org/10.1007/s11760-021-01889-3.

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AbstractWe consider the problem of recovering an original image $${\varvec{x}}$$ x from its filtered version $${\varvec{y}}={\varvec{f}}({\varvec{x}})$$ y = f ( x ) , assuming that the internal structure of the filter $${\varvec{f}}(\cdot )$$ f ( · ) is unknown to us (i.e., we can only query the filter as a black-box and, for example, cannot invert it). We present two new iterative methods to attack the problem, analyze, and evaluate them on various smoothing and edge-preserving image filters.

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15

Vazquez, Jon, Jose Felix Miñambres, Miguel Angel Zorrozua, and Jorge Lázaro. "Phasor Estimation of Transient Electrical Signals Composed of Harmonics and Interharmonics." Energies 14, no. 16 (August 20, 2021): 5166. http://dx.doi.org/10.3390/en14165166.

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Numerical relays have become essential tools for carrying out the protection and surveillance tasks of electrical power systems. These relays implement their functions from the phasors of the electrical network signals, estimated by digital processing using digital filters. Digital filters must meet certain requirements, such as providing a fast and effective response to increasingly complex transient signals made up of components that make the estimation process difficult. In addition to the decreasing exponential (decaying dc offset), harmonics, and signal noise, it must be added that the interharmonic components that in recent years have acquired great relevance, mainly because of the increase in non-linear loads and the extensive use of power electronic systems. The presence of these interharmonic components causes a poor response in most of the filters implemented today. This article presents the design of a new digital filter, C-CharmDF (Cleaned Characteristic Harmonic Digital Filter) for phasor estimation on noisy transient signals with decreasing exponential components, harmonics, and interharmonics. A detailed study was carried out for severe transient situations and stationary signals. It was found that the method can be suitable for relays that implement both fault location functions and system protection functions.

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Hidayat, Rahmad, Ninik Sri Lestari, Herawati Herawati, Givy Devira Ramady, Sudarmanto Sudarmanto, and Farhan Adani. "An approach of adaptive notch filtering design for electrocardiogram noise cancellation." Indonesian Journal of Electrical Engineering and Computer Science 22, no. 3 (June 1, 2021): 1303. http://dx.doi.org/10.11591/ijeecs.v22.i3.pp1303-1311.

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An electrocardiogram (ECG) is a means of measuring and monitoring important signals from heart activity. One of the major biomedical signal issues such as ECG is the issue of separating the desired signal from noise or interference. Different kinds of digital filters are used to distinguish the signal components from the unwanted frequency range to the ECG signal. To address the question of noise to the ECG signal, in this paper the digital notch filter IIR 47 Hz is designed and simulated to demonstrate the elimination of 47 Hz noise to obtain an accurate ECG signal. The full architecture of the structure and coefficient of the IIR notch filter was carried out using the FDA Tool. Then the model is finished with the help of Simulink and the MATLAB script was to filter out the 47 Hz noise from the signal of ECG. For this purpose, the normalized least mean square (NLMS) algorithm was used. The results indicate that before being filtered and after being filtered it clearly shows the elimination of 47 Hz noise in the signal of the ECG. These results also show the accuracy of the design technique and provide an easy model to filter out noise in the ECG signal.

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17

POPOV, Dmitrii, and Sergey SMOLSKIY. "Signal processing in recursive rejection filters in the transient mode." TURKISH JOURNAL OF ELECTRICAL ENGINEERING & COMPUTER SCIENCES 26 (2018): 194–203. http://dx.doi.org/10.3906/elk-1705-215.

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18

Liang, Yongsheng, Wei Liu, Shuangyan Yi, Huoxiang Yang, and Zhenyu He. "Filter pruning-based two-step feature map reconstruction." Signal, Image and Video Processing 15, no. 7 (March 31, 2021): 1555–63. http://dx.doi.org/10.1007/s11760-021-01888-4.

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AbstractIn deep neural network compression, channel/filter pruning is widely used for compressing the pre-trained network by judging the redundant channels/filters. In this paper, we propose a two-step filter pruning method to judge the redundant channels/filters layer by layer. The first step is to design a filter selection scheme based on $$\ell _{2,1}$$ ℓ 2 , 1 -norm by reconstructing the feature map of current layer. More specifically, the filter selection scheme aims to solve a joint $$\ell _{2,1}$$ ℓ 2 , 1 -norm minimization problem, i.e., both the regularization term and feature map reconstruction error term are constrained by $$\ell _{2,1}$$ ℓ 2 , 1 -norm. The $$\ell _{2,1}$$ ℓ 2 , 1 -norm regularization plays a role in the channel/filter selection, while the $$\ell _{2,1}$$ ℓ 2 , 1 -norm feature map reconstruction error term plays a role in the robust reconstruction. In this way, the proposed filter selection scheme can learn a column-sparse coefficient representation matrix that can indicate the redundancy of filters. Since pruning the redundant filters in current layer might dramatically influence the output feature map of the following layer, the second step needs to update the filters of the following layer to assure output of feature map approximates to that of baseline. Experimental results demonstrate the effectiveness of this proposed method. For example, our pruned VGG-16 on ImageNet achieves $$4\times $$ 4 × speedup with 0.95% top-5 accuracy drop. Our pruned ResNet-50 on ImageNet achieves $$2\times $$ 2 × speedup with 1.56% top-5 accuracy drop. Our pruned MobileNet on ImageNet achieves $$2\times $$ 2 × speedup with 1.20% top-5 accuracy drop.

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19

Grant, P. M. "Digital signal processing. Part 1: Digital filters and the DFT." Electronics & Communications Engineering Journal 5, no. 1 (1993): 13. http://dx.doi.org/10.1049/ecej:19930003.

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20

Kumar, Krishna, Rajlaxmi Pandey, Sankha Subhra Bhattacharjee, and Nithin V. George. "Exponential Hyperbolic Cosine Robust Adaptive Filters for Audio Signal Processing." IEEE Signal Processing Letters 28 (2021): 1410–14. http://dx.doi.org/10.1109/lsp.2021.3093862.

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21

Vallicelli, Elia Arturo, and Marcello De Matteis. "Analog Filters Design for Improving Precision in Proton Sound Detectors." Journal of Low Power Electronics and Applications 11, no. 1 (March 18, 2021): 12. http://dx.doi.org/10.3390/jlpea11010012.

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This paper analyzes how to improve the precision of ionoacoustic proton range verification by optimizing the analog signal processing stages with particular emphasis on analog filters. The ionoacoustic technique allows one to spatially detect the proton beam penetration depth/range in a water absorber, with interesting possible applications in real-time beam monitoring during hadron therapy treatments. The state of the art uses nonoptimized detectors that have low signal quality and thus require a higher total dose, which is not compatible with clinical applications. For these reasons, a comprehensive analysis of acoustic signal bandwidth, signal-to-noise-ratio and noise power/bandwidth will be presented. The correlation between these signal-quality parameters with maximum achievable proton range measurement precision will be discussed. In particular, the use of an optimized analog filter allows one to decrease the dose required to achieve a given precision by as much as 98.4% compared to a nonoptimized filter approach.

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Rohini, R., N. V. Satya Narayana, and Durgesh Nandan. "A Crystal View on the Design of FIR Filter." Journal of Computational and Theoretical Nanoscience 17, no. 9 (July 1, 2020): 4235–38. http://dx.doi.org/10.1166/jctn.2020.9052.

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In audio and video signal processing main element is the FIR filter. This paper presents complete information regarding the FIR filters. It also focuses on the design of FIR filters which provide low-area, energy-delay, low-power consumption, high-speed, low critical path, and low complexity. Implementation of FIR filters with different methods like memory-based VLSI architecture, filters for sampling rate conversion, linear phase FIR filters, optimal hybrid form FIR filters, Nyquist filters, hybrid multiplier less FIR filters, low complexity FIR filters, variable partition hybrid form FIR filters, area efficiency FIR filters are discussed in this paper. The objective of this paper to provide all related information regarding FIR filters at one platform.

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23

NERURKAR, SHAILESH B., and KHALID H. ABED. "LOW POWER DIGITAL DECIMATION FILTER FOR RF WIRELESS COMMUNICATIONS." Journal of Circuits, Systems and Computers 17, no. 02 (April 2008): 239–51. http://dx.doi.org/10.1142/s0218126608004241.

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In this paper, we present a unique low power decimation filter architecture for RF wireless applications. To implement the low power decimation filter, we considered low power design techniques such as multi-rate, multi-stage signal processing, proper selection of decimation factor, one multiplier realization of 1/3-band filters, and poly-phase 1/2-band filters. We have designed three conventional decimation filter architectures using a single-stage FIR filter, a three-stage FIR filter, and a three-stage half-band FIR filter. Compared to the 55-tap comb-FIR filter architecture, the proposed decimation filter has only 13 taps, and requires 76% less hardware and consumes 64% less power.

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24

Binek, Malgorzata, Andrzej Kanicki, and Pawel Rozga. "Application of an Artificial Neural Network for Measurements of Synchrophasor Indicators in the Power System." Energies 14, no. 9 (April 30, 2021): 2570. http://dx.doi.org/10.3390/en14092570.

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Dynamic phenomena in electric power systems require fast and accurate algorithms for processing signals. The processing results include synchrophasor parameters, e.g., varying amplitude, phase or frequency of sinusoidal voltage or current signals. This paper presents a novel estimation method of synchrophasor parameters that comply with the requirements of IEEE/IEC standards. The authors analyzed an algorithm for measuring the phasor magnitude by means of a selected artificial neural network (ANN), an algorithm for estimating the phasor phase and frequency that makes use of the zero-crossing method. The original components of the presented approach are: the method of the synchrophasor magnitude estimation by means of a suitably trained and applied radial basic function (RBF); the idea of using two algorithms operating simultaneously to estimate the synchrophasor magnitude, phase and frequency that apply identical calculation methods are different in that the first one filters the input signal using the FIR filter and the second one operates without any filter; and the algorithm calculating corrections of the phase shift between the input and output signal and the algorithm calculating corrections of the magnitude estimation. The error results obtained from the applied algorithms were compared with those of the quadrature filter method and the ones presented in literature, as well as with the permissible values of the errors. In all cases, these results were lower than the permissible values and at least equal to the values found in the literature.

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25

Kokil, Priyanka, and Swapnil Sadashiv Shinde. "An improved criterion for peak-to-peak realization of direct-form interfered digital filters employing saturation nonlinearities." COMPEL: The International Journal for Computation and Mathematics in Electrical and Electronic Engineering 34, no. 3 (May 5, 2015): 996–1010. http://dx.doi.org/10.1108/compel-07-2014-0166.

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Purpose – The purpose of this paper is to present a criterion for global asymptotic stability of state-space direct-form digital filters employing saturation arithmetic. Design/methodology/approach – An elegant use of induced l ∞ approach (also known as a peak-to-peak approach) is made to develop a criterion for the overflow stability of state-space direct-form digital filters. Findings – The criterion not only guarantees asymptotic stability but also reduces the effect of external interference. The presented method yields better interference attenuation level as compared to a recently reported method. Numerical examples are given to illustrate the effectiveness of the proposed method. Practical implications – Digital filters are important dynamical systems in signal processing which are used for the processing of discrete signals. During the implementation of higher-order digital filter in hardware or software, introduction of external interference is unavoidable. Therefore, stability analysis of digital filters in the presence of external interference is of much practical importance. Originality/value – The main result of the paper is reported for the first time and it is useful to establish the asymptotic stability of digital filters in the presence of external disturbances.

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26

Ward, C. "An Accelerator for Real-Time Digital Signal Processing with Microprocessors." International Journal of Electrical Engineering & Education 24, no. 1 (January 1987): 65–72. http://dx.doi.org/10.1177/002072098702400114.

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An accelerator consisting of a fast digital multiplier and A/D and D/A converters is designed for the BBC microcomputer. The circuit enables ‘hands-on’ experience of digital signal processing to be provided at minimal cost. Examples of implementations of FIR filters and an autocorrelation algorithm are provided.

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27

TAKAHASHI, NOBUAKI, TSUYOSHI OTAKE, and MAMORU TANAKA. "NONLINEAR INTERPOLATIVE EFFECT OF FEEDBACK TEMPLATE FOR IMAGE PROCESSING BY DISCRETE-TIME CELLULAR NEURAL NETWORK." Journal of Circuits, Systems and Computers 12, no. 04 (August 2003): 505–18. http://dx.doi.org/10.1142/s0218126603001008.

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Recently a discrete-time cellular neural network (DT-CNN) is applied to many image processing applications such as compression and reconstruction, recognition and so on. Conventional image processing techniques such as the discrete cosine transformation (DCT) and wavelet transforms work as a simple filter and do not make good use of interpolative dynamics by the feedback A template, which is one of the significant characteristics of a cellular neural network (CNN). If CNN is applied to a filter by an only feedforward B template, one should make a model which consists of digital filters using high speed signal processing modules such as a high speed digital signal processor. This paper describes the nonlinear interpolative effect of the feedback A template, by showing the evaluation of image compression and reconstruction.

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28

Vo, B. "Continuous-time envelope constrained filter design via orthonormal filters." IEE Proceedings - Vision, Image, and Signal Processing 142, no. 6 (1995): 389. http://dx.doi.org/10.1049/ip-vis:19952292.

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Zhang, Ying, Yubin Zhu, Kaining Han, Junchao Wang, and Jianhao Hu. "A High-Accuracy Stochastic FIR Filter with Adaptive Scaling Algorithm and Antithetic Variables Method." Electronics 10, no. 16 (August 11, 2021): 1937. http://dx.doi.org/10.3390/electronics10161937.

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Digital filter is an important fundamental component in digital signal processing (DSP) systems. Among the digital filters, the finite impulse response (FIR) filter is one of the most commonly used schemes. As a low-complexity hardware implementation technique, stochastic computing has been applied to overcome the huge hardware cost problem of high-order FIR filters. However, the stochastic FIR filter (SFIR) scheme suffers from long processing latency and accuracy degradation. In this paper, the bit stream representation noise is theoretically analyzed, and an adaptive scaling algorithm (ASA) is proposed to improve the accuracy of SFIR with the same bit stream length. Furthermore, a novel antithetic variables method is proposed to further improve the accuracy. According to the simulation results on a 64-tap FIR filter, the ASA and AV methods gain 17 dB and 6 dB on the signal-to-noise ratio (SNR), respectively. The hardware implementation results are also presented in this paper, which illustrates that the proposed ASA-AV-SFIR filter increases 4.6 times hardware efficiency with respect to the existing SFIR schemes.

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30

Sun, Tong, Moncef Gabbouj, and Yrjö Neuvo. "Adaptive L-filters with applications in signal and image processing." Signal Processing 38, no. 3 (August 1994): 331–44. http://dx.doi.org/10.1016/0165-1684(94)90153-8.

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Felja, Meryem, Asmae Bencheqroune, Mohammed Karim, and Ghita Bennis. "Removing Artifacts From Eeg Signal Using Wavelet Transform and Conventional Filters." International Journal of Circuits, Systems and Signal Processing 15 (March 12, 2021): 172–77. http://dx.doi.org/10.46300/9106.2021.15.19.

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Electroencephalogram (EEG) is a signal of an electrical nature reflecting the neuronal activities of the brain. It is used for the diagnosis of certain cerebral pathologies. However, it becomes more difficult to identify and analyze it when it is corrupted by artifacts of non-cerebral origin such as eye movements, cardiac activities ..., therefore, it is essential to remove these parasitic signals. In literature, there are different techniques for removing artifacts. This paper proposes and discusses a new EEG de-noising technique, based on a combination of wavelet transforms and conventional filters. The results of the proposed method are evaluated using three common criteria: signal-to-noise-ratio (SNR), mean square error (MSE) and cross correletion function (CCF). These experimental results demonstrate that the proposed approach can be an effective tool for removing artifact without suppression of any signal components.

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Balster, Eric J., Francis D. Fradette, Frank A. Scarpino, and Kerry L. Hill. "Time-Domain Matrix Analysis of Polyphase FIR Filters." International Journal of Electrical Engineering & Education 49, no. 3 (July 2012): 275–90. http://dx.doi.org/10.7227/ijeee.49.3.7.

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Polyphase filter design is a common subject studied in discrete systems analysis and digital signal processing (DSP) courses. However, the classic z-domain analysis, utilizing the noble identities, gives a conclusion to the true physical structures of polyphase filters which may not be obvious to many students. The proposed time-domain analysis provides a more straightforward development of polyphase implementation of interpolation and decimation functions, and hopes to provide students with a more visual representation of the polyphase interpolation and decimation processes. Results from a student survey show that over 73% of students believe that the proposed polyphase analysis strengthened their understanding of polyphase filters, and over 71% would prefer to use the proposed method over the traditional z-domain analysis when explaining polyphase filters to others.

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TOUMAZOU, CHRISTOFER. "ANALOGUE SIGNAL PROCESSING: THE “CURRENT” WAY OF THINKING." International Journal of High Speed Electronics and Systems 03, no. 03n04 (September 1992): 297–336. http://dx.doi.org/10.1142/s0129156492000126.

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This paper presents a tutorial review of the current-mode approach to analogue electronic circuit design, in particular some of the author’s own research over the past decade covering the development of a new generation of technology specific current-mode analogue signal processing. In this paper various technologies are represented with key current-mode building blocks ranging from current-conveyors, current-feedback operational amplifiers, linear transconductance amplifiers to applications ranging from current-mode active filters to low-noise current-mode optical preamplifiers. Advantages, future trends and perspectives of the current-mode approach are highlighted throughout the paper. The theoretical basis for many of the examples dates back many years but it is only recently due to advances in process technology that many of these techniques have now become a practical reality.

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Al-Safi, Amean. "ECG signal denoising using a novel approach of adaptive filters for real-time processing." International Journal of Electrical and Computer Engineering (IJECE) 11, no. 2 (April 1, 2021): 1243. http://dx.doi.org/10.11591/ijece.v11i2.pp1243-1249.

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Electrocardiogram (ECG) is considered as the main signal that can be used to diagnose different kinds of diseases related to human heart. During the recording process, it is usually contaminated with different kinds of noise which includes power-line interference, baseline wandering and muscle contraction. In order to clean the ECG signal, several noise removal techniques have been used such as adaptive filters, empirical mode decomposition, Hilbert-Huang transform, wavelet-based algorithm, discrete wavelet transforms, modulus maxima of wavelet transform, patch based method, and many more. Unfortunately, all the presented methods cannot be used for online processing since it takes long time to clean the ECG signal. The current research presents a unique method for ECG denoising using a novel approach of adaptive filters. The suggested method was tested by using a simulated signal using MATLAB software under different scenarios. Instead of using a reference signal for ECG signal denoising, the presented model uses a unite delay and the primary ECG signal itself. Least mean square (LMS), normalized least mean square (NLMS), and Leaky LMS were used as adaptation algorithms in this paper.

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Prisukhina, Ilona, Dmitry Borisenko, and Sergey Lunev. "Simulation Model of Electric Code-Modulated Signal in Russian Systems of Interval Control of Train Movement Based on Track Circuit." SPIIRAS Proceedings 18, no. 5 (September 19, 2019): 1212–38. http://dx.doi.org/10.15622/sp.2019.18.5.1212-1238.

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Systems of interval control of train movement Signaling systems, which are currently in service in Russian railways, use the electric track circuit as the main data channel between signals and locomotives. Code-modulated electric signals transferred through that channel are frequently get corrupted which leads to railway traffic delays. Decoding of the electric signal received from a track circuit can be represented as an image classification problem, and thus the stability of the data channel could be significantly improved. However, to build such a classifier based on some machine learning algorithm, one needs a large dataset. In this article, a simulation model to synthesize this dataset is proposed. The structure of the computer model matches the main stages of the electric code-modulated signal generation in a track circuit: code signal generator, rails, locomotive receiver. Based on code signal generator schematic and waveform diagrams, a generator algorithm is developed. At this stage, we modeled timings of electric code signals according to the specification as well as their random deviations caused by various factors. The analysis of substitution circuits of the rail line revealed that it has the properties of a low-pass filter. So, the rail line using the Butterworth digital filter with corresponding parameters is modeled. Additionally, at this stage, random noise during transmission was taken into account. A similar technique is applied for modeling of a locomotive receiver which has a band-pass filter as the first signal processing block. Thus, the proposed simulation model consists of a set of algorithms which run in series. By varying the parameters of the model, one can synthesize waveform diagrams of the electric code-modulated signal received by the locomotive equipment from a track circuit working in various modes and conditions.

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Wildhaber, Reto A., Nour Zalmai, Marcel Jacomet, and Hans-Andrea Loeliger. "Windowed State-Space Filters for Signal Detection and Separation." IEEE Transactions on Signal Processing 66, no. 14 (July 15, 2018): 3768–83. http://dx.doi.org/10.1109/tsp.2018.2833804.

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Orguner, Umut, and Fredrik Gustafsson. "Target Tracking With Particle Filters Under Signal Propagation Delays." IEEE Transactions on Signal Processing 59, no. 6 (June 2011): 2485–95. http://dx.doi.org/10.1109/tsp.2011.2122260.

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Kumar, B. V. K. Vijaya. "Partial Information Filters." Digital Signal Processing 4, no. 3 (July 1994): 147–53. http://dx.doi.org/10.1006/dspr.1994.1014.

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Kari, Dariush, Ali H. Mirza, Farhan Khan, Huseyin Ozkan, and Suleyman S. Kozat. "Boosted adaptive filters." Digital Signal Processing 81 (October 2018): 61–78. http://dx.doi.org/10.1016/j.dsp.2018.07.012.

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Jung, Jun Mo, and Jong-Wha Chong. "A Low Power FIR Filter Design for Image Processing." VLSI Design 12, no. 3 (January 1, 2001): 391–97. http://dx.doi.org/10.1155/2001/54974.

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In this paper, a new low power design method of the FIR filter for image processing is proposed. Because the correlation between adjacent pixels is very high in image data, the clock gating technique can be a good candidate for low power strategy. However, the conventional clock gating strategy that is applied independently to every flip-flop of the filter give rise to too much additional area overhead and couldn't get a good result in the power reduction. In our method, each tap register, which is used to delay the input data in the filter, is partitioned into two sub-registers according to the correlation characteristic of its input space. For the sub-register which highly correlated data is inputted into, the dynamic power consumption is reduced by diminishing switching activity of the clock signal. We can also reduce the additional hardware overhead by propagating the clock gating control signal of the first tap register to other tap registers. To identify the efficiency of the proposed design method, we perform the experiments on some filters that are designed in VHDL. The power estimation tool says that the proposed method can reduce the power dissipation of the filter by more than 18% compared to the conventional filter design methods.

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Kontogiannopoulos, N., and Costas Psychalinos. "Switched-Current Filters Revisited: Square-Root Domain Sampled-Data Filters." IEEE Transactions on Circuits and Systems II: Express Briefs 53, no. 12 (December 2006): 1373–77. http://dx.doi.org/10.1109/tcsii.2006.885969.

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Erdinc, O., P. Willett, and Y. Bar-Shalom. "The Bin-Occupancy Filter and Its Connection to the PHD Filters." IEEE Transactions on Signal Processing 57, no. 11 (November 2009): 4232–46. http://dx.doi.org/10.1109/tsp.2009.2025816.

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Wodecki, Jacek. "Time-Varying Spectral Kurtosis: Generalization of Spectral Kurtosis for Local Damage Detection in Rotating Machines under Time-Varying Operating Conditions." Sensors 21, no. 11 (May 21, 2021): 3590. http://dx.doi.org/10.3390/s21113590.

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Vibration-based local damage detection in rotating machines (i.e., rolling element bearings) is typically a problem of detecting low-energy cyclic impulsive modulations in the measured signal. This can be challenging as both the amplitude of a single damage-related impulse and the distance between impulses might be changing in time. From the signal processing point of view, this means time varying regarding the the signal-to-noise ratio, location of information in the frequency domain, and loss of periodicity (this remains cyclic but not periodic). One of the many attempted approaches to this problem is filtration using custom filters derived in a data-driven fashion. One of the methods to obtain such filters is a selector approach, where the value of a certain statistic is calculated for individual frequency bands of a signal that results in the magnitude response of a filter. In this approach, each chosen statistic will yield different results, and the obtained filter will be focused on different frequency bands focusing on different behaviors. One of the most popular and powerful selectors is spectral kurtosis as popularized by Antoni, which uses kurtosis as an operational statistic. Unfortunately, after closer inspection, it is easy to notice that, although selectors can significantly enhance the signal, they accumulate a great deal of noise and other background content of signals, which occupies the space (or rather time) in between the impulses. Another disadvantage is that such filters are time-invariant, because, in the principle of their construction, they are not adaptive, and even slight changes in the signal yield suboptimal results either by missing relevant data when the conditions in the signal change (i.e., informative impulses widen in bandwidth), or by accumulating unnecessary noise when the relevant information is not present (in between impulses or in frequency bands that impulses no longer occupy). To address this issue, I propose generalization of the selector approach using the example of spectral kurtosis. This assumes creating a time-varying selector that can be seen as a spatial filter in the time-frequency domain. The time-varying SK (TVSK) is estimated for segments of the signal, and, instead of a vector of SK-based filter coefficients, one obtains a TVSK-based matrix of coefficients that takes into account the time-varying properties of the signal. The obtained structure is then binarized and used as a filter. The presented method is tested using a simulated signal as well as two real-life signals measured on heavy-duty bearings in two different types of machine.

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Kumar, Ahlad. "Complementary metal‐oxide semiconductor implementation of digital filters for signal processing applications." IET Circuits, Devices & Systems 9, no. 4 (July 2015): 290–98. http://dx.doi.org/10.1049/iet-cds.2014.0236.

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Li, Y., G. R. Arce, and J. Bacca. "Weighted Median Filters for Multichannel Signals." IEEE Transactions on Signal Processing 54, no. 11 (November 2006): 4271–81. http://dx.doi.org/10.1109/tsp.2006.881208.

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Hou, Zhuo, Sanmin Shen, Yong Ye, Jiahao Deng, Yuting Liu, Qing Meng, and Zuodong Duan. "Research on the Linear Acceleration Sensor Signal Acquisition Technology Based on the High-Order Anti-Aliasing Cauer Filter." Journal of Circuits, Systems and Computers 29, no. 01 (March 21, 2019): 2050008. http://dx.doi.org/10.1142/s0218126620500085.

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A linear acceleration sensor integrated into an inertial measurement unit and its signal processing technology are presented in this paper. Based on the characteristics of the acceleration sensor, before analog-to-digital conversion, a design method for optimizing and conditioning the output signal in levels of frequency with the high-order anti-aliasing Cauer filter is proposed. Compared with the previously published papers, here we not only focus on the anti-aliasing filtering effect under a single channel, but also pay more attention to the anti-aliasing filtering effect with more data to the same type of channels with the same cut-off frequency and different types of channels with different cut-off frequencies. Similar to other kinds of filters, this paper points out that the high-order anti-aliasing Cauer filter also has its inherent delay characteristic. And this paper also reveals the qualitative relationship between frequency and time delay in different testing environments by using various delay test data. Compared with the previously published papers, through the simple solution processing with the true attitude data, this paper further estimates the error of simple attitude signal processing.

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RAMÍREZ, JAVIER, UWE MEYER-BÄSE, and ANTONIO GARCÍA. "EFFICIENT RNS-BASED DESIGN OF PROGRAMMABLE FIR FILTERS TARGETING FPL TECHNOLOGY." Journal of Circuits, Systems and Computers 14, no. 01 (February 2005): 165–77. http://dx.doi.org/10.1142/s0218126605002131.

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FIR filters are routinely used in the implementation of modern digital signal processing systems. Their efficient implementation using commercially available VLSI technology is a subject of continuous study and development. This paper presents the residue number system (RNS) implementation of reduced-complexity and high-performance FIR filters, using modern Altera APEX20K field-programmable logic (FPL) devices. Index arithmetic over Galois fields and the Quadratic Residue Number System (QRNS), along with a selection of a small wordwidth modulus set, are the keys for attaining low complexity and high throughput in real and complex FIR filters. RNS–FPL merged FIR filters demonstrated its superiority when compared to 2C (two's complement) filters, being about 65% faster and requiring fewer logic elements for most study cases. Special attention was paid to an efficient implementation of the multi-operand modulo adders. The replacement of a classical modulo adder tree by a binary adder with extended precision followed by a single modulo reduction stage reduced area requirements by 10% for a 32-tap FIR filter. On the other hand, an index arithmetic QRNS-based complex FIR filter yielded up to 60% performance improvement over a three-multiplier-per-tap 2C filter, while requiring fewer LEs for filters having more than eight taps. Particularly, a 32-tap filter needed 24% LEs less than the classical design.

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Zhao, Hui, Shengnan Li, Hongyu Yang, and Quan Zhou. "A stability controlling-based approach for designing 1-D variable fractional delay all-pass filters." COMPEL - The international journal for computation and mathematics in electrical and electronic engineering 37, no. 6 (November 5, 2018): 2224–31. http://dx.doi.org/10.1108/compel-01-2018-0053.

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Purpose Variable fractional delay filtering is an important technology in signal processing; the research shows that all-pass variable fractional delay (VFD) filters achieve higher design accuracy than FIR VFD filters; therefore, the design, analysis and implementation of all-pass VFD filters are of great importance. Design/methodology/approach In this paper, a two-stage approach for the design of general 1-D stable VFD all-pass filters is proposed. The method takes the desired group delay range [N−1, N], where N is the filter order. Findings The design algorithm is decomposed into two design stages: first, a set of fixed delay all-pass filters are designed by minimizing a set of objective functions defined in terms of approximating error criterion and filter stability constraint. Then, the design result is determined by fitting each of the fixed delay all-pass filter coefficients as 1-D polynomials. A design example together with its comparisons with those of the recent literature studies is given to justify the effectiveness of the proposed design method. Originality/value An illustrating design example shows that the method proposed can achieve better filter performances than the existing ones.

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Sonkur, Bülent. "Adaptive digital filters and signal analyses." Signal Processing 18, no. 2 (October 1989): 226–27. http://dx.doi.org/10.1016/0165-1684(89)90053-4.

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Lopes, Wilder Bezerra, and Cassio Guimaraes Lopes. "Geometric-Algebra Adaptive Filters." IEEE Transactions on Signal Processing 67, no. 14 (July 15, 2019): 3649–62. http://dx.doi.org/10.1109/tsp.2019.2916028.

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