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Signal Processing, IEEE Transactions on

Issue 10 • Date 1995

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Displaying Results 1 - 25 of 25
  • Appendixes to "An adaptive estimation of periodic signals using a Fourier linear combiner" [Erratum]

    Publication Year: 1995 , Page(s): 2435 - 2437
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (176 KB)  

    Due to a post-production error in the above-named work [ibid., vol. 42, no. 1, pp. 1-10, Jan. 1994], the Appendices, which were referred to repeatedly, did not appear at the end of the paper. The entire Appendix for this paper is published here in its entirety. View full abstract»

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  • Design of time-frequency representations using a multiform, tiltable exponential kernel

    Publication Year: 1995 , Page(s): 2283 - 2301
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1364 KB)  

    A novel Cohen's (1981) class time-frequency representation with a tiltable, generalized exponential kernel capable of attaining a wide diversity of shapes in the ambiguity function plane is proposed for improving the time-frequency analysis of multicomponent signals. The first advantage of the proposed kernel is its ability to generate a wider variety of passband shapes, e.g., rotated ellipses, generalized hyperbolas, diamonds, rectangles, parallel strips at arbitrary angles, crosses, snowflakes, etc., and narrower transition regions than conventional Cohen's class kernels; this versatility enables the new kernel to suppress undesirable cross terms in a broader variety of time-frequency scenarios. The second advantage of the new kernel is that closed form design equations can now be easily derived to select kernel parameters that meet or exceed a given set of user specified passband and stopband design criteria in the ambiguity function plane. Thirdly, it is shown that simple constraints on the parameters of the new kernel can be used to guarantee many desirable properties of time-frequency representations. The well known Choi-Williams (1989) exponential kernel, the generalized exponential kernel, and Nuttall's (1990) tilted Gaussian kernel are special cases of the proposed kernel View full abstract»

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  • Unitary equivalence: a new twist on signal processing

    Publication Year: 1995 , Page(s): 2269 - 2282
    Cited by:  Papers (100)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1244 KB)  

    Unitary similarity transformations furnish a powerful vehicle for generating infinite generic classes of signal analysis and processing tools based on concepts different from time, frequency, and scale. Implementation of these new tools involves simply preprocessing the signal by a unitary transformation, performing standard processing on the transformed signal, and then (in some cases) transforming the resulting output. The resulting unitarily equivalent systems can focus on the critical signal characteristics in large classes of signals and, hence, prove useful for representing and processing signals that are not well matched by current techniques. As specific examples of this procedure, we generalize linear time-invariant systems, orthonormal basis and frame decompositions, and joint time-frequency and time-scale distributions. These applications illustrate the utility of the unitary equivalence concept for uniting seemingly disparate approaches proposed in the literature View full abstract»

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  • Adaptive harmonic jammer canceler

    Publication Year: 1995 , Page(s): 2323 - 2331
    Cited by:  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (672 KB)  

    A new adaptive harmonic jammer canceler is proposed. It is based on the use of two sensors that enable an adaptive generation of a reference signal that is uncorrelated with the desired signal. This reference signal is used for the reconstruction of the desired signal by an adaptive subtraction method. This canceler is well suited to radio communications. A theoretical analysis of the convergence of the coupled algorithms is presented with the help of the associated ordinary differential equation introduced by L. Ljung. Numerical simulations illustrate the different proposed algorithms View full abstract»

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  • A new approach for deriving scattered coefficients of pipelined IIR filters

    Publication Year: 1995 , Page(s): 2405 - 2406
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (160 KB)  

    The synthesis of a particular class of pipelined IIR digital filter requires the denominator of the z-transform transfer function to be expressed in scattered form. The existing technique for deriving the scattered coefficient values requires finding the zeros of the denominator polynomial. The author presents a new approach for deriving the scattered coefficient values that does not require the evaluation of the zeros of the transfer function's denominator polynomial View full abstract»

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  • Discrete Gabor structures and optimal representations

    Publication Year: 1995 , Page(s): 2258 - 2268
    Cited by:  Papers (18)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (824 KB)  

    The idea of Gabor's (1946) signal expansion is to represent a signal in terms of a discrete set of time-shifted and frequency modulated signals that are localized in the time-frequency (or phase) space. We present detailed descriptions of the block and banded structures for the Gabor matrices. Based on the explicit descriptions of the sparsity of such matrices, we can establish the sparse form of the Gabor matrix and obtain the dual Gabor atom (mother wavelet), the inverse of the Gabor frame operator, and carry out the discrete finite Gabor transform in a very efficient way. Some explicit sufficient and also necessary conditions are derived for a Gabor atom g to generate a Gabor frame with respect to a TF-lattice (a, b) View full abstract»

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  • Complex, linear-phase filters for efficient image coding

    Publication Year: 1995 , Page(s): 2425 - 2427
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (320 KB)  

    With the exception of the Haar basis, real-valued orthogonal wavelet filter banks with compact support lack symmetry and therefore do not possess linear phase. This has led to the use of biorthogonal filters for coding of images and other multidimensional data. There are, however, complex solutions permitting the construction of compactly supported, orthogonal linear phase QMF filter banks. By explicitly seeking solutions in which the imaginary part of the filter coefficients is small enough to be approximated to zero, real symmetric filters can be obtained that achieve excellent compression performance View full abstract»

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  • Combined instrumental variable and subspace fitting approach to parameter estimation of noisy input-output systems

    Publication Year: 1995 , Page(s): 2386 - 2397
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (916 KB)  

    The paper considers the problem of estimating the parameters of linear discrete-time systems from noise-corrupted input-output measurements, under fairly general conditions: the output and input noises may be auto-correlated and they may be cross-correlated as well. By using the instrumental-variable (IV) principle a covariance matrix is obtained, the singular vectors of which bear complete information on the parameters of the system under study. A weighted subspace fitting (WSF) procedure is then employed on the sample singular vectors to derive estimates of the parameters of the system. The combined IV-WSF method proposed in the present paper is noniterative and simple to use. Its large-sample statistical performance is analyzed in detail and the theoretical results so obtained are used to predict the behavior of the method in samples with practical lengths. Several numerical examples are included to show the agreement between the theoretically predicted and the empirically observed performances View full abstract»

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  • An efficient calculation of the moments of matched and mismatched hidden Markov models

    Publication Year: 1995 , Page(s): 2422 - 2425
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (336 KB)  

    Streit (1990) analysed the classification of an unknown hidden Markov model (HMM) using a set of prescribed HMMs. He proposed a suboptimal test statistic that can be approximated by certain moments for this classification. In the present paper the algorithm given by Streit to derive these moments is reformulated in a matrix algebra setting that gives a better insight into the algorithm. Also, an asymptotic analysis of the algorithm is derived using the reformulation View full abstract»

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  • Experimental results of localization of moving underwater signal by adaptive beamforming

    Publication Year: 1995 , Page(s): 2249 - 2257
    Cited by:  Papers (19)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (812 KB)  

    The problem of weak moving signal localization and tracking in the presence of single motionless strong interference is investigated using real data of an underwater experiment in the Baltic sea (Sept. 1990) with a horizontal receiving array of 64 hydrophones and with two independent powerful narrowband sources imitating the signal and interference. Three simple adaptive beamforming methods were employed for the experimental data processing. The first one is based on the well-known projection approach to adaptive beamforming, the second method uses the adaptive canceler approach (also termed the dipole pattern method), and the third method combines these approaches. The signal-to-interference power ratio (SIR) threshold of the signal localization and tracking is evaluated by a special technique, which allows examination of the considered algorithms with change of the SIR in consecutive order. The results of the data processing show the high possibilities of signal localization in the presence of strong interference. The combined method performs better than the methods considered and enables localization of the signal source up to an SIR≃-25 dB View full abstract»

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  • Transform domain IIR filtering

    Publication Year: 1995 , Page(s): 2431 - 2434
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB)  

    The paper provides a generalization on the concept of the transform domain filtering (TDF). TDF is block-based filtering, or vector-filtering, applied to transform domain data to get the effect of desired time domain filtering. The result is transform domain IIR filtering (IIR-TDF) and it takes in the existing TDF, or FIR-TDF, as its special case. The resulting structures are two-fold-direct I form and direct II form-and are both block-filtering structures View full abstract»

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  • Multiscale difference equation signal models. I. Theory

    Publication Year: 1995 , Page(s): 2332 - 2345
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1064 KB)  

    The paper studies multiscale difference equation models for l-D and M-D signals. In this modeling technique, the signal of interest is viewed as a solution to a multiscale difference equation (MSDE). The model completely characterizes the signal as well as a number of its higher derivatives. It provides a recursive signal interpolation scheme as a function of scale. It also leads naturally to multigrid signal filtering, detection and estimation algorithms. An MSDE model must be uniquely decodable, i.e., it must correspond to a unique signal. Therefore, one must guarantee that the modeling MSDE has a unique solution. The authors investigate the existence and uniqueness of L1 and L2 solutions-to multiscale difference equations. Using Fourier domain techniques, they derive conditions for the existence of L1 solutions to an MSDE. They provide conditions under which the L1 solution is unique (up to a multiplicative constant) and has compact support. They also derive sufficient, but not necessary, conditions for the existence of a unique L2 solution to a subclass of MSDEs. The results extend known facts about the solutions of two-scale difference equations. The paper concludes with several examples of MSDE signal models that highlight the modeling advantages of MSDEs over two-scale difference equation models View full abstract»

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  • Multirate modeling of AR/MA stochastic signals and its application to the combined estimation-interpolation problem

    Publication Year: 1995 , Page(s): 2302 - 2312
    Cited by:  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1000 KB)  

    The use of the Kalman filter is investigated in this work for interpolating and estimating values of an AR or MA stochastic signal when only a noisy, down-sampled version of the signal can be measured. A multirate modeling theory of the AR/MA stochastic signals is first derived from a block state-space viewpoint. The missing samples are embedded in the state vector so that missing signal reconstruction problem becomes a state estimation scheme. Next, Kalman state estimation theory is introduced to treat the combined estimation-interpolation problem. Some extensions are also discussed for variations of the original basic problem. The proposed Kalman reconstruction filter can be also applied toward recovering missing speech packets in a packet switching network with packet interleaving configuration. By analysis of state estimation theory, the proposed Kalman reconstruction filters produce minimum-variance estimates of the original signals. Simulation results indicate that the multirate Kalman reconstruction filters possess better estimation/interpolation performances than a Wiener reconstruction filter under adequate numerical complexity View full abstract»

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  • An ESPRIT-like method for quadratic phase coupling estimation

    Publication Year: 1995 , Page(s): 2346 - 2360
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1212 KB)  

    Two algorithms are proposed for estimating the quadratically coupled frequency pairs (QC pairs) in a signal consisting of complex sinusoids in white noise. Three matrices are constructed from the complex third-order cumulants of the noisy signal, the second and third being time shifted versions of the first. The list of coupled frequencies is obtained from the rank reducing numbers of the matrix pencil formed from the first matrix and either of the latter two. The first algorithm then pairs these components by relating quadratic coupling to the intersection of generalized eigenspaces corresponding to two of these frequencies. The coupling strengths are also obtained in terms of generalized eigenvectors in this intersection space. The second algorithm constructs a two-parameter matrix pencil using all the three matrices. The rank reducing pairs of this pencil on the unit circle yield the QC pairs and the associated generalized eigenvectors: the coupling strengths View full abstract»

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  • An adaptive optimal-kernel time-frequency representation

    Publication Year: 1995 , Page(s): 2361 - 2371
    Cited by:  Papers (57)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (876 KB)  

    Time-frequency representations with fixed windows or kernels figure prominently in many applications, but perform well only for limited classes of signals. Representations with signal-dependent kernels can overcome this limitation. However, while they often perform well, most existing schemes are block-oriented techniques unsuitable for on-line implementation or for tracking signal components with characteristics that change with time. The time-frequency representation developed in the present paper, based on a signal-dependent radially Gaussian kernel that adapts over time, surmounts these difficulties. The method employs a short-time ambiguity function both for kernel optimization and as an intermediate step in computing constant-time slices of the representation. Careful algorithm design provides reasonably efficient computation and allows on-line implementation. Certain enhancements, such as cone-kernel constraints and approximate retention of marginals, are easily incorporated with little additional computation. While somewhat more expensive than fixed kernel representations, this new technique often provides much better performance. Several examples illustrate its behavior on synthetic and real-world signals View full abstract»

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  • On reconstruction methods for processing finite-length signals with paraunitary filter banks

    Publication Year: 1995 , Page(s): 2407 - 2410
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (364 KB)  

    New expressions are developed for the perfect reconstruction of the boundary regions of a finite-length signal after subband processing. The time-invariant filter bank is required to be uniform and paraunitary, using FIR filters regardless of phase or symmetry. They accommodate a linear boundary extension in the analysis section, and avoid periodic extensions or storage of extended subband signals. The reconstruction methods are based on the formulation of linear systems that are built as a function of the filters View full abstract»

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  • Weighted least-squares design and characterization of complex FIR filters

    Publication Year: 1995 , Page(s): 2398 - 2401
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (324 KB)  

    The paper presents two novel weighted least-squares methods for the design of complex coefficient finite impulse response (FIR) filters to attain specified arbitrary multiband magnitude and linear or arbitrary phase responses. These methods are computationally efficient, requiring only the solution of a Toeplitz system of N linear equations for an N-length filter that can be obtained in o(N2) operations. Illustrative filter design examples are presented View full abstract»

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  • Second-order statistics of morphological dilation and erosion of a memoryless source

    Publication Year: 1995 , Page(s): 2418 - 2422
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (400 KB)  

    The paper shows that morphological dilation and erosion preserve strict sense stationarity, derives the 2-D probability distribution and density functions of dilated and eroded stationary memoryless sources, and presents numerical results for a memoryless uniform source, which shows identical closed-form autocovariance functions of dilated and eroded sources View full abstract»

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  • Blind restoration of linearly degraded discrete signals by Gibbs sampling

    Publication Year: 1995 , Page(s): 2410 - 2413
    Cited by:  Papers (25)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (352 KB)  

    This paper addresses the problem of simultaneous parameter estimation and restoration of discrete-valued signals that are blurred by an unknown FIR filter and contaminated by additive Gaussian white noise with unknown variance. Assuming that the signals are stationary Markov chains with known state space but unknown initial and transition probabilities, Bayesian inference of all unknown quantities is made from the blurred and noisy observations. A Monte Carlo procedure, called the Gibbs sampler, is employed to calculate the Bayesian estimates. Simulation results are presented to demonstrate the effectiveness of the method View full abstract»

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  • Stochastic differential equations: an approach to the generation of continuous non-Gaussian processes

    Publication Year: 1995 , Page(s): 2372 - 2385
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (904 KB)  

    The generation of continuous random processes with jointly specified probability density and covariation functions is considered. The proposed approach is based on the interpretation of the simulated process as a stationary output of a nonlinear dynamic system, excited by white Gaussian noise and described by a system of a first-order stochastic differential equations (SDE). The authors explore how the statistical characteristics of the equation's solution depends on the form of its operator and on the intensity of the input noise. Some aspects of the approximate synthesis of stochastic differential equations and examples of their application to the generation of non-Gaussian continuous processes are considered. The approach should be useful in signal processing when it is necessary to translate the available a priori information on the real random process into the language of its Markov model as well as in simulation of continuous correlated processes with the known probability density function View full abstract»

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  • Constraints on the cutoff frequencies of Mth-band linear-phase FIR filters

    Publication Year: 1995 , Page(s): 2401 - 2405
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (392 KB)  

    Constraints are derived for the cutoff frequencies of linear-phase FIR Mth-band filters such that the filters have good passband and stopband characteristics, i.e. ones that very closely approximate an ordinary (non Mth-band) filter designed using some optimal method. Constraints on lowpass filters are first considered, and the results are extended to multiband filters View full abstract»

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  • Bursting in the LMS algorithm

    Publication Year: 1995 , Page(s): 2414 - 2417
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (260 KB)  

    The least mean square (LMS) algorithm is known to converge in the mean and in the mean square. However, during short time periods, the error sequence can blow up and cause severe disturbances, especially for non-Gaussian processes. The paper discusses potential short time unstable behavior of the LMS algorithm for spherically invariant random processes (SIRP) like Gaussian, Laplacian, and K0. The result of this investigation is that the probability for bursting decreases with the step size. However, since a smaller step size also causes a slower convergence rate, one has to choose a tradeoff between convergence speed and the frequence of bursting View full abstract»

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  • Shape calibration for a nominally linear equispaced array

    Publication Year: 1995 , Page(s): 2241 - 2248
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (680 KB)  

    We consider a thin flexible line array of equispaced hydrophones that is towed through the sea and develop a statistical test that allows us to decide whether the array is straight or not. The motion of the towing ship, the ocean currents and other forces induce deformations on the array and affect the performance of spatial processing of the data developed under the assumption that the array is straight. When the ship is maneuvering, the processing is generally turned off for long periods of time, an extremely penalizing situation that can be overcome by applying our scheme. Indeed, if the test-applied to the whole array-decides that it is out of shape, the same test is applied to parts of the array to determine the maximum size of admissible subarrays on which the standard processing can be pursued. By combining the bearings estimated by the different subarrays, we reconstruct a piecewise linear estimate of the shape of the array. The approach allows the handling of quite important deformations with no need for a cooperating source View full abstract»

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  • Two-channel IIR QMF banks with approximately linear-phase analysis and synthesis filters

    Publication Year: 1995 , Page(s): 2313 - 2322
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (840 KB)  

    Perfect linear-phase two-channel QMF banks require the use of finite impulse response (FIR) analysis and synthesis filters. Although they are less expensive and yield superior stopband characteristics, perfect linear phase cannot be achieved with stable infinite impulse response (IIR) filters. Thus, IIR designs usually incorporate a postprocessing equalizer that is optimized to reduce the phase distortion of the entire filter bank. However, the analysis and synthesis filters of such an IIR filter bank are not linear phase. In this paper, a computationally simple method to obtain IIR analysis and synthesis filters that possess negligible phase distortion is presented. The method is based on first applying the balanced reduction procedure to obtain nearly allpass IIR polyphase components and then approximating these with perfect allpass IIR polyphase components. The resulting IIR designs already have only negligible phase distortion. However, if required, further improvement may be achieved through optimization of the filter parameters. For this purpose, a suitable objective function is presented. Bounds for the magnitude and phase errors of the designs are also derived. Design examples indicate that the derived IIR filter banks are more efficient in terms of computational complexity than the FIR prototypes and perfect reconstruction FIR filter banks. Although the PR FIR filter banks when implemented with the one-multiplier lattice structure and IIR filter banks are comparable in terms of computational complexity, the former is very sensitive to coefficient quantization effects View full abstract»

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  • Fast base extension and precise scaling in RNS for look-up table implementations

    Publication Year: 1995 , Page(s): 2427 - 2430
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (292 KB)  

    Both base extension and scaling are fundamental operations in residue computing and several techniques have been proposed previously for their efficient implementation. Using look-up tables, the best result (log2 n table took-up cycles, where n is the number of residue moduli in the system) has been obtained by using the Chinese remainder theorem (CRT) at the expenses of a redundant representation of the numbers and of an approximated scaling. The CRT approach is reconsidered and it is shown that the same average time performances (log2 n lookup cycles) can be achieved without any redundancy and with a precise result for scaling View full abstract»

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IEEE Transactions on Signal Processing covers novel theory, algorithms, performance analyses and applications of techniques for the processing, understanding, learning, retrieval, mining, and extraction of information from signals

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Editor-in-Chief
Sergios Theodoridis
University of Athens