nils-herrmann/clean_scirepeval huggingface dataset:
A Signal-Perturbation-Free Transmit Scheme for MIMO-OFDM Channel Estimation

In this paper, a novel signal-perturbation-free (SPF) approach is presented for frequency-selective multiple-input-multiple-output orthogonal frequency division multiplexing channel estimation. First, an efficient transmit scheme, which bears partial information of the correlation matrix of the transmitted signal called SPF data, is proposed for the cancellation of signal-perturbation error at the receiver. A detailed transmit structure is designed to implement the SPF scheme, which is then employed along with linear prediction (LP) to derive a new semiblind channel-estimation algorithm. It is shown that the new transmit scheme can completely cancel the signal-perturbation error in the noise-free case while being able to sufficiently suppress the perturbation error in noisy conditions. It is also shown that the SPF data needs only to be transmitted over a small number of subcarriers, and its overhead to the overall transmission is negligible as compared with regular pilot signals. Computer simulations show that the proposed SPF solution significantly outperforms the LP semiblind method without using the proposed transmit scheme as well as the least square method in terms of the mean-square error of the channel estimate.
A novel normalization technique for multimodal biometric systems

A multimodal biometric system consolidates multiple biometric sources and mitigates the limitations of the unimodal biometric system. The consolidation of information can be done at various levels of fusion. In this paper, a normalization technique for score-level fusion based on a new anchor, which is computed from the raw score set, has been proposed. This new anchor is independent of the statistatical properies of the biometric system (e.g., equal error rate). This work focuses on the improvement of the multimodal biometric system that consists of at least one weak classifier and does not have a prior knowledge of the genuine/impostor score distribution. The experimental results show that the performance of simple non-weighted fusion preceded by our EER independent anchor based normalization technique is better than that of the weak classifier of the system and is superior to that of two other normalization methods.
Some new results in the design of Mth-band FIR filters

In this paper, the design of Mth-band FIR filters via semidefinite programming (SDP) is investigated. Some of the frequency response properties of a general nonlinear-phase Mth-band filter are first examined, yielding a few constraints for the amplitude and phase responses of the Mth-band filter. An SDP-based optimization problem is formulated for the least-square and minimax design of Mth-band filters. Although the formulation is established for a nonlinear phase filter, the design result shows that an exactly linear phase can be obtained when the desired phase/delay characteristic is properly specified.
A parallel/pipelined algorithm for the computation of MDCT and IMDCT

A novel parallel/pipelined algorithm that leads to a scalable parallel/pipelined hardware architecture for the computation of the MDCT and the IMDCT required for the Advanced Audio Compression (AAC) standard has been presented. The development of algorithms for the MDCT and the IMDCT for the Audio Compression-3 (AC-3) standard as well as the short and long transforms in these two wide band audio coding standards follow the approach described.
Characterization of B-spline digital filters

Digital filters arising in the B-spline signal processing are characterized in a unified manner in this paper. The transfer functions of these filters are the z-transforms of the uniformly sampled central B-splines shifted by an arbitrary value. The transfer functions are cascades of an FIR kernel filter and a simple moving average FIR filter. For certain values of the shift parameter, the filters are identical to those referred to as the B-spline digital filters in the literature. The filters thus form a general family of B-spline digital filters. The kernel part of the B-spline filters may be used for transforming a discrete-time signal to a representation based on the B-spline coefficients. The B-spline filters may also be used to convert a sequence of B-spline coefficients to a discrete-time spline signal. The contributions of the paper are as follows. A unifying recurrence relation enabling the computation of the impulse response coefficients of the B-spline kernel filters is derived. An accompanying recurrence relation is also obtained for the entire transfer function of the kernel filters. The recurrences are valid for arbitrary values of the shift parameter. It is proved that the roots of the transfer functions of the kernel filters are distinct, negative and real. We also prove that the roots of the kernel filters of successive orders interlace. The results regarding the location of the zeros are also valid for arbitrary values of the shift parameter. The relation of the kernel filters to the Eulerian polynomials is discussed. It is shown that for certain choices of the parameters the kernel filters are equivalent to the classical Eulerian polynomials that frequently arise in combinatorics. An alternative closed-form expression for the kernel filters in the Bernstein form is also derived. Besides their importance in unifying the existing results on B-spline filters, the generalized family of B-spline filters studied in this paper find applications in fractional delay of B-spline signals.
Mixed phase ARMA system identification by bispectrum: a group delay approach

A method of identification of a mixed-phase system (driven by a zero mean nonGaussian white noise) from its output, by an autoregressive moving-average (ARMA) model is proposed. This is achieved by relating the bispectrum phase and magnitude to the system group delay functions. The method uses only the bispectrum information and it is applicable for MA, AR, or ARMA systems with either mixed-phase poles, mixed-phase zeros, or both. The identification does not involve any solution of a system of equations and is free from the limitations found in many of the existing methods. Its evaluation for different ARMA terms indicates that its performance is good in terms of root-mean-square (which accounts both for bias and variance), since the normalized sum of the sample mean-square error is about 1%.<<ETX>>
Generation of two-dimensional digital functions without nonessential singularities of the second kind

A class of two-variable Hurwitz polynomials called very strict Hurwitz polynomials (VSHP's) is defined and its properties are studied. Their application in the generation of two-variable functions without nonessential singularities of the second kind is indicated. Necessary and sufficient conditions on general and reactance one-variable to two-variable transformations so that they yield two-variable transfer functions without singularities of the second kind are obtained.
2-D IIR digital filters for deinterlacing of video signals

Investigates the application of 2D IIR digital filters for deinterlacing of video signals. A special class of diamond filters is presented, which employs only two coefficients and has approximately zero-phase response. The implementation of this filter is very efficient and the stability is guaranteed even under finite arithmetic conditions. This filter is tested with a group of sequences. Very promising results are obtained.<<ETX>>
Comparison of the effects of quantization on digital filters

A variety of digital filter network structures is available. Different network structures posses different quantization properties. The fixed-point roundoff noise and coefficient sensitivity properties of the Swamy-Thyagarajan Wave Digital Filter, the cascade canonic, the Sedlmeyer-Fettweis Wave Digital Filter and the Generalized-Immittance Converter structures are examined in the present paper. The relative power spectral density and the statistical wordlength of each structure are computed.
Image denoising in wavelet domain using the vector-based hidden Markov model

Denoising problems can be regarded as that of a prior probability modeling in an estimation task. The performance of the estimator is intimately related on the correctness of the model. This paper proposes a new wavelet-domain image denoising method using the minimum mean square error (MMSE) estimator. The vector-based hidden Markov model (HMM) is used as the prior for modeling the wavelet coefficients of an image. This model is an effective statistical model for the wavelet coefficients, since it is capable of capturing both the subband marginal distribution and the inter-scale, intra-scale and cross-orientation dependencies of the wavelet coefficients. Using this prior, a Weiner filter, which is derived using a MMSE estimator, is developed for estimating the denoised coefficients. Experiments are conducted on standard images to evaluate the performance of the proposed method. Simulation results are provided to show that the proposed denoising method can effectively reduce the noise in yielding higher values for the peak signal-to-noise ratio along with better visual quality than that provided by some of the other existing methods.
A new blind-block reciprocal parametric transform

In this paper, we define a blind-block reciprocal- transpose matrix operator (BBRT). One of the interesting proprieties of the BBRT matrix is that it is very easy to obtain its inverse. We then propose a new blind-block reciprocal parametric transform and show that its matrix operator is a BBRT matrix. The transform has a large number of independent parameters that are useful for many applications and can specifically be used as an additional secret key for encryption and watermarking. It is shown that the proposed transform reduces to some of the existing transforms and also to new classes of parametric transforms having some desirable proprieties.
Low power current comparator cell for weak current operations

In this paper, we present a novel current comparator cell that converts a current signal to a two-level voltage through an adaptive thresholding. By applying a special compensation scheme, the non-uniformity of the threshold value in an array of such cells is lowered to less than 1%. The comparator cell can work at extremely low input currents, with the power dissipation in the range of nono-Watts or lower. The proposed circuit is simple and operates at high speed, since the operation of the cell is not dependent on a charge accumulation process. It can be fabricated in a single poly process, thus reducing the cost and also making it compatible with digital technology.
Resolution of range and Doppler ambiguities in medium PRF radars in multiple-target environment

In medium pulse-repetition frequency (PRF) radars, ambiguities may arise in both range and Doppler measurements. Efficient techniques have been established  to resolve the range ambiguity of a single isolated target using multiple PRF's. In this paper, we describe a simple algorithm to resolve the Doppler ambiguity using the discrete Fourier transform (DFT) output of two PRF's. A condition on the relative values of the two PRF's is derived to account for the errors due to the finite bandwidth of DFT filters. A third PRF is used to identify the declarations of a particular target in different PRF's. Range ambiguities are then resolved in a straightforward manner. A fourth PRF is made use of to extract blind-speed targets. The proposed method is computationally efficient and can be used even when ambiguous returns from several targets are received.
Design of a transistor-mismatch-insensitive switched-current memory cell

In this paper, we present a new and efficient switched-current memory cell consisting of six MOS transistors. A charge re-adjusting procedure is implemented in the memory cell to ensure an acceptable processing accuracy. Functionally, the current memory is insensitive to transistor parameter mismatch. The cell is currently under fabrication using 1.5-/spl mu/m p-well single-poly technology with an area of about 50/spl times/25 /spl mu/m/sup 2/. Applications of the proposed memory cell for the design of a multi-purpose analog signal processing operator are also discussed.
A feedforward timing recovery scheme using two samples per symbol: algorithm, performance and implementation issues

Feedforward symbol timing recovery techniques are widely used in burst-mode transmission systems. In this paper, a timing estimation method that uses two samples/symbol sampling rate is investigated. We provide a theoretical justification for the algorithm and present the variance of the timing estimate for different parameters including roll-off factor and estimation interval length. The timing estimation performance of a modified version of the algorithm is also investigated. Some of the implementation issues are discussed.
A necessary and sufficient condition for the BIBO stability of general-order bode-type variable-amplitude wave-digital equalizers

Recently, the authors developed a new synthesis technique for the design of higher-order Bode-type variable-amplitude (VA) wave-digital (WD) equalizers. The salient feature of the resulting VA WD equalizers is that they permit the continuous variation of the WD equalizer transfer function from a shaping transfer function to its inverse by changing the value of a single variable digital multiplier only. The proposed design technique was based on the WD realization of the corresponding positive-real analog prototype shaping impedance function, and on the realization of the equalizer transfer function as the reflectance of the shaping impedance function with respect to the constituent variable digital multiplier. This paper is concerned with an investigation of the bounded-input bounded-output (BIBO) stability of general-order VA WD equalizers. It is shown that the resulting conditions are both necessary and sufficient for the BIBO stability of the VA WD equalizers for the entire range of values for the variable digital multiplier. These conditions can be checked in a straightforward fashion in terms of the characteristics of the shaping transfer function alone. An application example is given to illustrate the main results.
Complete characterization of systems for simultaneous Lagrangian upsampling and fractional-sample delaying

We present a complete formulation and an exact solution to the problem of designing systems for simultaneous sampling rate increase and fractional-sample delay in the Lagrangian sense. The problem may be regarded as that of a linear transformation, i.e., scaling, and/or shifting, of the uniform sampling grid of a discrete-time signal having a Newton series representation. It is proved that the solution forms a three-parameter family of maximally flat finite impulse response digital filters with a variable group-delay at the zero frequency. Various properties of the solution, including Nyquist properties and conditions for a linear phase response are analyzed. The solution, obtained in the closed form, is exact for polynomial inputs. We show that it is also suited for processing discrete-time versions of certain continuous-time bandlimited signals and signals having a rational Laplace transform. We then derive a generalization of the solution by augmenting the family with a fourth parameter that controls the number of multiple zeros at the roots of unity. This four-parameter family contains various types of maximally flat filters including those due to Herrmann and Baher. We list specific conditions on the four parameters to obtain many of the maximally flat filters reported in the literature. A significant part of the family of systems characterized by the solutions has been hitherto unknown. Examples are provided to elucidate this part as well.
Necessary and sufficient conditions for the stability of a class of 2-D recursive digital filters

In this paper we propose an alternative approach for testing the stability of two-dimensional causal recursive digital filters which are devoid of nonessential singularities of the second kind. We make a detailed comparison of the computational efficiency of this method with two other algebraic methods available in the literature. We also derive, using the approach presented in this paper, a simple set of necessary and sufficient conditions for the stability of a class of 2-D recursive filters.
Direct Fault Location in Electrical Power Systems

Fault location in electric power transmission or distribution systems is essential to the maintenance of reliable and continuous service. As a result, increased effort is being focused on fault locating procedures in distribution systems such that faults may be automatically isolated and unfaulted feeders restored to service in a short time. Presented is an^ alternative to existing fault locating procedures based upon the concept of the adjoint circuit and the recently derived theorem on the maximum number of independent simulations in N-port circuits. The theoretical approach is described for open circuit, short circuit, and mixed open and short circuit faults, followed by a numerical example applied to fault location in radial distribution feeders.
A necessary and sufficient condition for the BIBO stability of general-order Bode-type variable-amplitude wave-digital equalizers

Recently, the authors developed a new synthesis technique for the design of higher-order Bode-type variable-amplitude (VA) wave-digital (WD) equalizers. The salient feature of the resulting VA WD equalizers is that they permit the continuous variation of the WD equalizer transfer function from a shaping transfer function to its inverse by changing the value of a single variable digital multiplier only. The proposed design technique was based on the WD realization of the corresponding positive-real analog prototype shaping impedance function, and on the realization of the equalizer transfer function as the reflectance of the shaping impedance function with respect to the constituent variable digital multiplier. This paper is concerned with an investigation of the bounded-input bounded-output (BIBO) stability of general-order VA WD equalizers. It is shown that the resulting conditions are both necessary and sufficient for the BIBO stability of the VA WD equalizers for the entire range of values for the variable digital multiplier. These conditions can be checked in a straightforward fashion in terms of the characteristics of the shaping transfer function alone. An application example is given to illustrate the main results.
Accurate estimation of the glottal flow derivative using iteratively reweighted 1-norm minimization

The problem of estimating the exact shape of the glottal flow derivative (GFD) using reweighted 1-norm minimization of the second derivative of the GFD is addressed in this paper. By using physiological models of the glottal flow derivative, such as the Liljencrants-Fant (LF) and Rosenberg models, it is intuitively found that the second derivative of those models is highly sparse. Based on this observation an iteratively reweighted 1-norm minimization algorithm is proposed to accurately estimate the vocal tract of the speech signal by exploiting the sparsity of the second derivative of the GFD (the residual of the linear prediction model). An experimental study using a data set of 40 vowels /a/ and /e/, 20 for each, is conducted, showing the efficiency, in terms of the number of iterations and the total run-time reduction, of the proposed algorithm. Furthermore, the results of estimating the GFD of two vowels /a/ & /e/ using Joint Source-Filter Model Optimization and our proposed method, demonstrate the accuracy, in terms of similarity to the physiological model and precise synthesis, of our proposed algorithm.
A continuous tracking algorithm for long-term memory motion estimation [video coding]

It is known that long-term memory motion estimation can bring a significant coding gain to a video coding system. In this paper, a fast algorithm for long-term memory motion estimation is presented. The proposed algorithm attempts to track the direction of the best long-term motion vector in a frame-by-frame manner from the most recent reference frame to the oldest reference frame. Nine direction patterns for the direction tracking are used and the search locations of the long-term memory motion estimation are adaptively selected according to the chosen direction pattern. Simulation results demonstrate that the proposed algorithms can speed up long-term motion estimation by more than 40 times.
A trace-back-free Viterbi decoder using a new survival path management algorithm

The problem of survival memory management of a Viterbi decoder is classically solved either by a power-hungry register-exchange implementation, or by a trace-back scheme with large latency. In this paper, we show that by introducing a novel matrix transformation operator to the algebraic foundation originally proposed in Fettweis (1995), we obtain a new survival-path-management algorithm that sorts the survival paths in parallel with the decoding of the information bits to eliminate long recursive trace-back operations. Furthermore, it avoids power and complexity problems associated with register exchange by only tracing and recording the changes of the survival path with a set of dynamic pointers. A prototype of the Viterbi decoder for the GSM TCH/FS system has been mapped into a Xilinx Virtex xcv400HQ240-6 operating at 23.6 MHz with a decoding throughput of 3.37 Mbit/s and 80 mW power consumption, a 43% increase over the same application based on StarCore SC140 DSP in terms of throughput per MHz.
Modified MRAC based on Lyapunov theory for improved controller efficiency

In this paper, a control algorithm is derived by cascading a feedback controller with the conventional MRAC (Model Reference Adaptive Control) based on Lyapunov theory. The stability of the derived algorithm is proved using Lyapunov stability criterion. The proposed controller is designed to control a pneumatic globe control valve and simulation is performed using MATLAB/Simulink software. The pneumatic globe control valve is further simulated with the individual PI controller and the MRAC controller, and the response of the three controllers is compared. The results show that the proposed controller has a fast and smooth convergence of the plant output with respect to the reference model, thereby showing robustness of the controller.
Adaptive beamforming with joint robustness against covariance matrix uncertainty and signal steering vector mismatch

A novel adaptive beamforming algorithm that is robust against covariance matrix uncertainty and the signal steering vector mismatch is proposed. Based on the optimisation of the diagonal loading covariance matrix in the sense of minimum mean square error, the mismatch vector between the actual steering vector and the presumed one is estimated iteratively by solving a quadratic convex optimisation problem, which is then used to update the steering vector to obtain the optimal beamformer.
An efficient approach for the design of nearly perfect-reconstruction QMF banks

In this work, an approach for the design of a class of quadrature mirror filter (QMF) banks is presented based on a combination of IIR analysis filters and FIR synthesis filters. By examining the phase property of the all-pass based analysis filter, a condition on the synthesis part for a perfect reconstruction is established, making it possible to realize a nearly perfect-reconstruction (PR) filter bank by using a nonlinear-phase FIR synthesis filter. The distinct feature of the proposed mixed IIR/FIR filter bank is that it enjoys both a small reconstruction error and a low design and implementation complexity.