nils-herrmann/clean_scirepeval huggingface dataset:
Externally controlled time-varying constrained notch filter

An application of a constrained notch filter (CNF) for filtering a sine signal with arbitrary varying frequency is considered. It is shown that the concept of CNF can be easily extended to time-varying CNF (TVCNF) whose coefficients vary with time according to the rule of frequency variation of the considered FM signal. Such a TVCNF is able to reject the FM signal and at the same time remain transparent to any other signal. The main problem in TVCNF implementation is to provide a proper control (synchronization) of its coefficients. An application of the TVCNF for the enhancement of an arbitrary signal corrupted by a strong FM interference is investigated. It is shown that if the synchronization is based on the input mixture, then for any negative signal-to-interference ratio (SIR<<0 dB) at the input of the TVCNF, the SIR at its output reaches 0 dB.<<ETX>>
A constrained anti-Hebbian learning algorithm for total least-squares estimation with applications to adaptive FIR and IIR filtering

In this paper, a new Hebbian-type learning algorithm for the total least-squares parameter estimation is presented. The algorithm is derived from the classical Hebbian rule. An asymptotic analysis is carried out to show that the algorithm allows the weight vector of a linear neuron unit to converge to the eigenvector associated with the smallest eigenvalue of the correlation matrix of the input signal. When the algorithm is applied to solve parameter estimation problems, the converged weights directly yield the total least-squares solution. Since the process of obtaining the estimate is optimal in the total least-squares sense, its noise rejection capability is superior to those of the least-squares-based algorithms. It is shown that the implementations of the proposed algorithm have the simplicity of those of the LMS algorithm. The applicability and performance of the algorithm are demonstrated through computer simulations of adaptive FIR and IIR parameter estimation problems.
A new radix-2/8 FFT algorithm for length-q×2m DFTs

In this paper, a new radix-2/8 fast Fourier transform (FFT) algorithm is proposed for computing the discrete Fourier transform of an arbitrary length N=q/spl times/2/sup m/, where q is an odd integer. It reduces substantially the operations such as data transfer, address generation, and twiddle factor evaluation or access to the lookup table, which contribute significantly to the execution time of FFT algorithms. It is shown that the arithmetic complexity (multiplications+additions) of the proposed algorithm is, in most cases, the same as that of the existing split-radix FFT algorithm. The basic idea behind the proposed algorithm is the use of a mixture of radix-2 and radix-8 index maps. The algorithm is expressed in a simple matrix form, thereby facilitating an easy implementation of the algorithm, and allowing for an extension to the multidimensional case. For the structural complexity, the important properties of the Cooley-Tukey approach such as the use of the butterfly scheme and in-place computation are preserved by the proposed algorithm.
Video Denoising Using Motion Compensated 3-D Wavelet Transform With Integrated Recursive Temporal Filtering

A novel framework of the motion-compensated 3-D wavelet transform (MC3DWT) for video denoising is presented in this paper. The motion-compensated temporal wavelet transform is first performed on a sliding window of video frames consisting of previously denoised frames and the current noisy frame. The 2-D spatial wavelet transform is then performed on the temporal subband frames, thus realizing a 3-D wavelet transform. Any of established wavelet-based still image denoising algorithms can then be applied to the high-pass 3-D subbands. The operation of the inverse 2-D spatial wavelet transform followed by the inverse temporal wavelet transform reconstructs the video frames in the buffer. The denoised current frame may be used as an output for real-time processing; meanwhile, the past frames can be updated, one of which may be used as a delayed output for post-processing or for real-time processing that allows some amount of delay. The proposed MC3DWT framework integrates both the spatial filtering and recursive temporal filtering into the 3-D wavelet domain and effectively exploits both the spatial and temporal redundancies. Experimental results have demonstrated a superior visual and quantitative performance of the proposed scheme for various levels of noise and motion.
Simple design procedure for active lowpass filters using exponential RC lines

Design equations are given for realising a pair of complex poles by an active distributed structure, utilising exponential RC lines. It is shown that the response of the structure closely approximates that of a 2nd-order low-pass filter. The loci of the complex poles may be considered, to a high degree of accuracy, to lie on parabolas, resulting in a simple design procedure. It is shown that the sensitivity of the structure with respect to the active element decreases with increasing taper, and that it may be made considerably lower than that of Kerwin's structure, which uses uniform RC lines.
Identification of Differentially Expressed Genes for Time-Course Microarray Data Based on Modified RM ANOVA

The regulation of gene expression is a dynamic process, hence it is of vital interest to identify and characterize changes in gene expression over time. We present here a general statistical method for detecting changes in microarray expression over time within a single biological group and is based on repeated measures (RM) ANOVA. In this method, unlike the classical F-statistic, statistical significance is determined taking into account the time dependency of the microarray data. A correction factor for this RM F-statistic is introduced leading to a higher sensitivity as well as high specificity. We investigate the two approaches that exist in the literature for calculating the p-values using resampling techniques of gene-wise p-values and pooled p-values. It is shown that the pooled p-values method compared to the method of the gene-wise p-values is more powerful, and computationally less expensive, and hence is applied along with the introduced correction factor to various synthetic data sets and a real data set. These results show that the proposed technique outperforms the current methods. The real data set results are consistent with the existing knowledge concerning the presence of the genes. The algorithms presented are implemented in R and are freely available upon request.
An alternate method for the recovery of band-limited sequences from nonuniformly decimated versions

Originally, the problem was addressed in the context of multirate signal processing, and methods based on polyphase filter banks have been proposed in the past. The procedure proposed here is based on the recovery of missing samples from an oversampled version of a bandlimited signal. Simulation results show that the power of the error in the recovery is on the order of 30 dB below the signal power. It has been shown that the proposed method, for certain decimation ratios, involves only matrix multiplications, as opposed to the design and implementation of broadband Hilbert transformers as required by the polyphase filter bank approaches. An overlap procedure is also suggested to handle the online recovery of long data sequences.<<ETX>>
Step-response-based characterization of nonuniform perfect-reconstruction filter banks

An algebraic characterization of nonuniform perfect-reconstruction (PR) filter banks with integer decimation factors is derived. The PR property is formulated in the z domain based on the response of the systems to delayed unit step signals. This leads to derivation of a new class of characterizing formulas in terms of the rate conversion factors and the transfer functions of channel filters. We use the formulas to give a set of analytic necessary and sufficient conditions for a nonuniform delay chain filter bank to be a PR system. These conditions are identical to those that characterize exact covering systems of congruence relations using roots of unity. We introduce important results from the mathematics literature on exact covering systems. These results elucidate the admissible factors of decimation in nonuniform PR delay chain systems with maximally distinct decimation factors.
A least-square design approach for 2D FIR filters with arbitrary frequency response

In this paper, a least-square approach for the design of two-dimensional (2D) complex finite-impulse response (FIR) filters with arbitrary frequency response is presented. By minimizing the frequency-domain error function and revealing some of the properties of the matrices associated with the design problem, a closed-form solution is obtained. The solution is presented as an expression for the impulse response corresponding to the desired frequency-response specification. Thus, the method avoids the usual time-consuming procedures of optimization or matrix inversion, and makes a very fast calculation of the filter's coefficients possible, it is also shown that when this solution is used to design linear-phase filters and a class of 2D phase equalizers, some further computational savings can be achieved. The novel feature of the proposed approximation approach is that it can be used to design any kind of 2D FIR filters without employing any symmetry constraint on their frequency responses. Several design examples illustrating the efficiency of the approach are considered.
Time-domain estimation of unambiguous Doppler frequency in low and medium PRF radars

Rader  described an algorithm to estimate the period of undersampled periodic signal. We extend this method to estimate ambiguous Doppler frequency in low and medium pulse-repetition frequency (PRF) radars. Multiple PRF's are employed to resolve the ambiguity. The proposed method results in a constant false alarm rate detector. The performance of the algorithm in the presence of noise is presented. It is shown that in association with another detector, the proposed method resolves the Doppler ambiguity with good accuracy even at low signal-to-noise ratios. Finally, the method is a time-domain approach and does not require any spectral analyser to estimate the frequency.
Ultrasound image despeckling in the contourlet domain using the Cauchy prior

Speckle noise reduction is a prerequisite task in images captured by ultrasonography systems due to their inherent noisy nature. In this work, we propose a new despeckling method in the contourlet domain using the Cauchy prior. The multiplicative speckle noise is first transferred to an additive one using a logarithmic transform. The logarithmically-transformed contourlet coefficients of the image and noise are assumed to be the Cauchy and Maxwell distributions, respectively. In order to estimate the noise-free contourlet coefficients, an efficient closed-form Bayesian maximum a posteriori estimator is developed. Simulations are carried out to evaluate the performance of the proposed despeckling method by using the synthetically-speckled and real ultrasound images. It is shown that the proposed method outperforms several existing techniques in terms of the signal-to-noise ratio and is able to preserve the diagnostically signific ant details of the ultrasound images.
Correction to "Symmetry constraints on two-dimensional half-plane digital transfer functions"

point arithmetic with the accuracy provided by the machine. A built-in program was used to generate white complex noise (with its real and imaginary parts being distributed uniformly between +l/fi) for feeding the two programs. A stable estimate of average NSR of each module was obtained for two different word lengths using the results of the two programs. v. RESULTS AND DISCUSSIONS For the purpose of calculation of the NSR, tl and t2 are assumed to be same. It is evident from Table I11 that the correlated model predicts the NSR much closer to the experimental results (for both 8 and 12 bits) than those predicted by the uncorrelated model. It is observed that the predicted output NSR falls short at most by 10.888 percent while those predicted by the uncorrelated model fall short by as much as 4 1.98 percent from the corresponding simulation results. While analyzing for the variances due to the coefficient quantization and rounding due to multiplication, the earlier authors  are not consistent in assuming the multiplying coefficients like +1 and ?j as noiseless. However, the present analysis considers these coefficients as noiseless for all the modules studied. The values of b and c thus obtained here are slightly different from those of Patterson and McClellan [ 61. Two truncation errors connected through more than one path, though correlated, have very small correlation coefficients. Since the analytical computation of such a correlation coefficient is more involved, it is neglected. The slight difference between simulation results and those of some correlated models could be due to such an assumption. Overscaling a module has been adopted to facilitate scaling procedure which, in turn, has introduced extra noise variance. Long-length GW DFT algorithms are derived using these shortlength DFT algorithms. Patterson and McClellan  have shown that the order in which the component algorithms are used affects the error performance. For minimum error performance, they should be used in such a way that the numbers (o;Ni)/(Ni - 1) are in increasing order. With respect to the output noise variance per unit, the modules are arranged in descending order as 7, 5, 9, 16, 3, 8, 4, and 2 which, according to the uncorrelated model, was 5, 7, 9, 3, 16, 8,4, and 2. Close agreement between theoretically predicted results and those of simulations justifies the validity of assumptions made in the analysis. The error performance of large-N GW algorithms under the assumption of correlation between truncation errors has been studied elsewhere [ 81.
Approximation of two-variable filter specifications in analog domain

This paper describes a technique for approximating 2-variable filter specifications in the continuous or analog domain. It is shown that the design of 2-variable filter functions using this approach reduces to the problem of identifying a suitable 2-variable reactance function g(s_1,s_2) and the realization of a stable single-variable transfer function T(s) . Then T(g(s_1,s_2)) is the desired 2-variable stable transfer function which is guaranteed to have at least one realization whenever T(s) and g(s_1,s_2) are realizable. Applications of the theory developed in this paper are presented in the design of lumped-distributed filters and 2-dimensional digital filters.
Statistically optimal filtering of averaged ABRS

The transient Auditory evoked Brainstem Response (ABR) requires the averaging of 2000 or more trials. A set of ABRs obtained at various stimulus intensity levels, for hearing threshold assessments can require 30-45 minutes. In an effort to reduce this acquisition time, the number of responses in the averaging is reduced resulting in noisy ABRs. Here, the authors use a damped sinusoid model of the ABRs and present a method of reducing the noise in the averaged ABRs. The new method is a two step procedure that first estimates some of the ABR model parameters which are used to form a statistically optimal line-enhancer. An ABR example is presented to show the effectiveness of this approach.
A study of two-terminal current-controlled negative resistance devices with the aid of an analogue

Some fundamental aspects of current- and voltage-controlled negative resistance devices are considered, in particular the relation of the physical mechanism involved to the equivalent circuit. Further, a unified stability criterion is presented which is applicable to both types of device. A versatile analogue of a current-controlled device is described which is compatible with a voltage-controlled analogue previously described. The high-frequency properties of this analogue are examined both experimentally and theoretically to determine the variation of the equivalent circuit parameters with frequency. With the aid of the analogue two circuit configurations are examined: a bistable trigger circuit, and a four-stable-state circuit obtained by the addition of a voltage-controlled analogue. It is believed that the techniques described enable circuits to be investigated at relatively low speeds, thus facilitating the optimization and understanding of a given configuration.
DOA estimation and tracking for signals with known waveform via symmetric sparse subarrays

In this paper, we present a novel approach to the problem of estimating and tracking the direction-of-arrival (DOA) of signals with known waveforms and unknown gains impinging on symmetric sparse subarrays. Unlike the conventional methods, which estimate the DOA based on the spatial signature of the signal with known waveform, the proposed method partitions the whole least square (LS) problem into multiple linear regression models of which each obtains a pair of DOA and gain. Here, we exploit a simple and efficient QR-decomposition-based recursive least square (QRD-RLS) technique to solve each linear regression model. Thanks to the unitary transformation for symmetric array configuration, we can decouple the pair of DOA and gain easily. Finally, several numerical examples showing the performance of the method are provided.
FPGA design and implementation of a low-power systolic array-based adaptive Viterbi decoder

In this paper, by modifying the well-known Viterbi algorithm, an adaptive Viterbi algorithm that is based on strongly connected trellis decoding is proposed. Using this algorithm, the design and a field-programmable gate array implementation of a low-power adaptive Viterbi decoder with a constraint length of 9 and a code rate of 1/2 is presented. In this design, a novel systolic array-based architecture with time multiplexing and arithmetic pipelining for implementing the proposed algorithm is used. It is shown that the proposed algorithm can reduce by up to 70% the average number of ACS computations over that by using the nonadaptive Viterbi algorithm, without degradation in the error performance. This results in lowering the switching activities of the logic cells, with a consequent reduction in the dynamic power. Further, it is shown that the total power consumption in the implementation of the proposed algorithm can be reduced by up to 43% compared to that in the implementation of the nonadaptive Viterbi algorithm, with a negligible increase in the hardware.
An efficient algorithm for the computation of the reverse jacket transform

This paper proposes an efficient algorithm to compute the reverse jacket transform algorithm by introducing a two-step decomposition strategy coupled with an appropriate use of the Kronecker product. Comparisons are carried out with the existing algorithms and the results show that a significant reduction in the number of data transfers and address generations as well as the structural complexity can be easily achieved using the proposed algorithm without increasing the arithmetic complexity. It is also shown that the three weights used in the existing reverse jacket transform are not required and just two are sufficient. Further, it is shown that a significant reduction in the number of multiplications can be achieved by using two rather than three weights, without losing the generality of the reverse jacket transform
L/sub p/ norm design of weighted order statistic filters

This paper addresses the problem of designing weighted order statistic (WOS) filters by employing an objective function given as the L/sub p/ norm of the error between the desired signal and the estimated one. The conventional design of WOS filters uses a mean absolute error (MAE) objective function, and as such, it is a special case of the general, L/sub p/ norm based design, developed here. It is shown that in stack filtering, the L/sub p/ norm can be expressed as a linear combination of the decision errors incurred by the Boolean operators at each level of the stack filter architecture. Based on this formulation of the L/sub p/ norm, both nonadaptive and adaptive algorithms for the design of L/sub p/ WOS filters are developed. A design example is considered, to illustrate the performance of the designed L/sub p/ WOS filters with different values of p. The simulation results show that the L/sub p/ WOS filters with p/spl ges/2 are capable of removing more impulsive noise compared with the conventional MAE WOS filters.
A CMOS current-controlled oscillator and its applications

In this paper, we present the design of a new current-controlled oscillator (ICO). The control current of this ICO is in a nano or sub-nano Ampere range and the sensitivity of the frequency of its output voltage to the control current is about 700 MHz per /spl mu/A. The circuit can operate with a low supply voltage, e.g. V/sub DD/ < 1 V when it is implemented with a 0.18-/spl mu/m technology. It requires no bias current, thus have very low power dissipation. The circuit can be easily implemented using a standard digital CMOS technology. Some applications of the proposed ICO, particularly those for A/D conversion, are also described in the paper.
Multiresolution DCT decomposition for multifocus image fusion

Image fusion is gaining momentum in the research community with the aim of combining all the important information from multiple images such that the fused image contains more accurate and comprehensive information than that contained in the individual images. In this paper, it is proposed to fuse multifocus images in the multiresolution DCT domain instead of the wavelet domain to reduce the computational complexity. The performance of the fused image in the proposed domain is compared with that of the wavelet domain with four recently-proposed fusion rules. The proposed method is applied on several pairs of multifocus images and the performance compared visually and quantitatively with that of wavelets. It is found that the performance of the proposed method is superior/similar to that of wavelets in terms of visual quality and quantitative parameters with extra benefits of computational efficiency and simplicity of implementation.
A Split Vector-Radix Algorithm for the 3-D Discrete Hartley Transform

In this paper, we propose a three-dimensional (3-D) split vector-radix fast Hartley transform (FHT) algorithm. The main idea behind the proposed algorithm is that the radix-2/4 approach is introduced in the decomposition of the 3-D discrete Hartley transform by using an appropriate index mapping and the Kronecker product. This provides an algorithm based on a mixture of radix-(2times2times2) and radix-(4times4times4) index maps and has a butterfly that is characterized by simple closed-form expressions. This algorithm offers substantial reductions in the numbers of multiplications, additions, data transfers, and twiddle factor evaluations or accesses to the look-up table, without a significant increase in the structural complexity compared to that of the existing 3-D vector radix FHT algorithm
Estimating the steady-state frequency of a sine-wave burst with extremely short record length

We present a technique based on modified linear prediction for accurate estimation of the steady-state frequency of a sine-wave burst with extremely short record length. The estimation procedure presented here makes use of automatic windowing of the burst location within the observation interval. Such a windowing technique eliminates the transition effects which accompany the estimators derived using the traditional LPC technique. To implement this windowing, a set of artificial state variables, given by the product of output samples and their delayed versions, is used. The advantages of this method are demonstrated using a computer simulation.