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IEEE Transactions on Signal Processing

Issue 6 • June 1992

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Displaying Results 1 - 25 of 41
  • Correction to 'Comments on 'A curiosum concerning discrete time convolution',' plus a remark

    Publication Year: 1992
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (83 KB)

    In the author's comments to an article by Hall and Wise (see ibid., vol.38, no.6, p.1059, 1990) included a typographic error which is corrected. He also points out that the sequences used in the example become L/sub 1/ norm sequences.<> View full abstract»

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  • Gain-adapted hidden Markov models for recognition of clean and noisy speech

    Publication Year: 1992, Page(s):1303 - 1316
    Cited by:  Papers (39)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1184 KB)

    In applying hidden Markov modeling for recognition of speech signals, the matching of the energy contour of the signal to the energy contour of the model for that signal is normally achieved by appropriate normalization of each vector of the signal prior to both training and recognition. This approach, however, is not applicable when only noisy signals are available for recognition. A unified appr... View full abstract»

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  • Performance analysis of a split-path LMS adaptive filter for AR modeling

    Publication Year: 1992, Page(s):1375 - 1382
    Cited by:  Papers (18)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (580 KB)

    A split-path adaptive filter is proposed for extracting the model parameters of an autoregressive process. The structure is composed of two linear phase filters connected in parallel, one antisymmetric and the other symmetric. The two filters are adapted independently on a sample-by-sample basis using the least-mean-square (LMS) algorithm. The performance of the system in terms of convergence spee... View full abstract»

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  • Newton algorithms for conditional and unconditional maximum likelihood estimation of the parameters of exponential signals in noise

    Publication Year: 1992, Page(s):1528 - 1534
    Cited by:  Papers (27)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (560 KB)

    The authors present polynomial-based Newton algorithms for maximum likelihood estimation (MLE) of the parameters of multiple exponential signals in noise. This formulation can be used in the estimation, for example, of the directions of arrival of multiple noise-corrupted narrowband plane waves using uniform linear arrays and the frequencies of multiple noise-corrupted complex sine waves. The algo... View full abstract»

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  • Excursions of adaptive algorithms via the Poisson clumping heuristic

    Publication Year: 1992, Page(s):1443 - 1451
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (704 KB)

    The authors detail the application of the Poisson clumping heuristic (PCH) to the least mean square (LMS) adaptive algorithm and its signed variants. Under certain conditions on the input and disturbance statistics, the parameter estimate errors form a Markov process. The PCH asserts that large excursions of the parameter estimates occur in clumps, and that these clumps are distributed in a Poisso... View full abstract»

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  • Least squared error FIR filter design with transition bands

    Publication Year: 1992, Page(s):1327 - 1340
    Cited by:  Papers (50)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1136 KB)

    The authors propose the use of transition bands and transition functions in the ideal amplitude frequency response to allow the analytical design of optimal least-squared-error FIR digital filters with an explicit control of the transition band edges. Design formulas are derived for approximations to ideal frequency responses which use pth-order spline transition functions. A mixed analyt... View full abstract»

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  • On prime factor mapping for the discrete Hartley transform

    Publication Year: 1992, Page(s):1399 - 1411
    Cited by:  Papers (11)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (808 KB)

    The authors propose a new prime factor mapping scheme, which requires no extra arithmetic operations for the realization of prime factor mapping, for the computation of the discrete Hartley transform (DHT). It is achieved by embedding all the extra arithmetic operations into the subsequent short-length computations, with the computational complexities of these embedded short lengths remaining unch... View full abstract»

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  • On unique localization of constrained-signals sources

    Publication Year: 1992, Page(s):1542 - 1547
    Cited by:  Papers (17)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (428 KB)

    Conditions for unique localization of radiating sources by passive sensor arrays are presented. Unlike in previous analyses, the case wherein the signals are constrained to certain loci in the complex plane is addressed. Two types of constraints are considered. The first includes the case of constant-amplitude signals and the second includes the case of complex-exponential signals. The conditions ... View full abstract»

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  • On focusing matrices for wide-band array processing

    Publication Year: 1992, Page(s):1295 - 1302
    Cited by:  Papers (81)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (672 KB)

    A general class of transformation matrices for coherent signal-subspace processing is presented. These signal-subspace transformation (SST) matrices are shown to generate a sufficient statistic for maximum-likelihood bearing estimation. Two general forms for calculating SST matrices are presented, and the rotational signal-subspace (RSS) focusing matrices proposed by H. Hung and M. Kaveh (1988) ar... View full abstract»

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  • Efficient order recursive algorithms for multichannel least squares filtering

    Publication Year: 1992, Page(s):1354 - 1374
    Cited by:  Papers (22)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1412 KB)

    Four efficient order-recursive algorithms for least-squares (LS) multichannel FIR filtering and multivariable system identification are developed. The need for such algorithms arises when the system model assigns an unequal number of delay elements to each input channel. All proposed schemes provide considerable improvements over overparametrization or the zero padding approach. First, a block-str... View full abstract»

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  • Two methods for Toeplitz-plus-Hankel approximation to a data covariance matrix

    Publication Year: 1992, Page(s):1490 - 1498
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (572 KB)

    Recently, fast algorithms have been developed for computing the optimal linear least squares prediction filters for nonstationary random processes (fields) whose covariances have (block) Toeplitz-Hankel form. If the covariance of the random process (field) must be estimated from the data, the following problem is presented: given a data covariance matrix, computer from the available data, find the... View full abstract»

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  • Linear bispectrum of signals and identification of nonminimum phase FIR systems driven by colored input

    Publication Year: 1992, Page(s):1469 - 1479
    Cited by:  Papers (11)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (872 KB)

    The identification of non-minimum-phase finite-impulse-response (FIR) systems driven by third-order stationary colored signals that are not linear processes is addressed. Modeling the linear part of the bispectrum of a signal is discussed. The bispectrum of a signal is decomposed into two multiplicative factors. The linear bispectrum is defined as the factor of the bispectrum that can exactly be m... View full abstract»

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  • Fully sigma-delta modulation encoded FIR filters

    Publication Year: 1992, Page(s):1605 - 1610
    Cited by:  Papers (34)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (464 KB)

    The author suggests an FIR filter structure where both the input signal and the impulse response are encoded using sigma-delta modulation. As a result, the convolution can be obtained by summing a collection of binary or ternary numbers, depending on the quantizer in the sigma-delta modulator. This filter structure can accept analog input signals directly and can be built using modular hardware. T... View full abstract»

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  • An adaptive array robust to beam pointing error

    Publication Year: 1992, Page(s):1582 - 1584
    Cited by:  Papers (22)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (252 KB)

    The authors propose an algorithm that is robust to beam pointing error. In the algorithm, the weight vector of a beam steered adaptive array corresponds to the maximum eigenvector of a data covariance matrix which is filtered by the projection operator composed of the eigenvectors of noise subspace of the covariance matrix in which the desired signal is removed by subtractive preprocessing. The re... View full abstract»

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  • Uniform approximation with doubly finite Volterra series

    Publication Year: 1992, Page(s):1438 - 1442
    Cited by:  Papers (22)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (316 KB)

    The assumption that a system possesses a certain discrete-time Volterra series representation frequently forms the basis for studies in the areas of signal processing and communication theory. A further assumption often made, always without discussion, is that the representation can be suitably approximated by a corresponding `double finite' series. It is shown that, for a very large class of nonl... View full abstract»

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  • Recursive autoregressive spectral estimation by minimization of the free energy

    Publication Year: 1992, Page(s):1518 - 1527
    Cited by:  Papers (10)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (700 KB)

    A digital signal processing technique applicable to power spectrum estimation, designated as the minimum free energy method, is described. With no a priori model assumption and no attempt to extract special features such as sinusoids, one can obtain high resolution even with high noise contamination of the measured signal. The technique is demonstrated by modification of the Burg recursive method ... View full abstract»

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  • Design of windows with steerable sidelobe dips

    Publication Year: 1992, Page(s):1452 - 1459
    Cited by:  Papers (11)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (504 KB)

    Based on the frequency sampling method, a novel and simple window design technique was established upon three theorems. The technique produces a set of windows which have simple expressions and the steerable sidelobe-dip (SSLD) property. By steering the dips in the sidelobes, it is easy to reduce the edge of the mainlobe, or to minimize the height of sidelobes, or to create a `hole' in the sidelob... View full abstract»

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  • Suppression of limit cycles in the first-order two-dimensional direct form digital filter with a controlled rounding arithmetic

    Publication Year: 1992, Page(s):1599 - 1601
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (240 KB)

    The first-order two-dimensional direct-form digital filter with magnitude truncation is known to be free from limit cycles for a limited range of allowed filter coefficients. In this correspondence, the quantization technique of controlled rounding is used to extend the region of filter coefficients for which suppression of limit cycles can be proved. With controlled rounding a signal is quantized... View full abstract»

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  • Spectral distortion in sampling rate conversion by zero-order polynomial interpolation

    Publication Year: 1992, Page(s):1576 - 1579
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (336 KB)

    For such applications as digital beamforming in sonar and sampling rate conversion in digital audio systems, there is interest in fast, approximate interpolation techniques. Computationally, the simplest of these is to approximate the desired time series sample by the available sample which is nearest in time. This procedure is variously known as sample-and-hold interpolation, zero-order polynomia... View full abstract»

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  • Adaptive code excited predictive coding

    Publication Year: 1992, Page(s):1317 - 1326
    Cited by:  Papers (7)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (740 KB)

    A novel way to use the code excited linear prediction (CELP) concept that decreases the processing load while keeping the same speech quality is discussed. Rather than performing individual weighting of each candidate sequence, a global implementation of the perceptual weighting function at the codebook level is proposed. As a result, the analysis-by-synthesis procedure does not require the proces... View full abstract»

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  • Joint blind equalization, carrier recovery and timing recovery for high-order QAM signal constellations

    Publication Year: 1992, Page(s):1383 - 1398
    Cited by:  Papers (119)  |  Patents (48)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1160 KB)

    Two existing blind equalization tap update recursions for 64-point and greater QAM (quadrature amplitude modulation) signal constellations are studied, along with existing and novel carrier and timing recovery techniques. It is determined that the superior tap update recursion is the one known as the constant modulus algorithm. Carrier recovery requires a modified second-order decision-directed di... View full abstract»

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  • An updating algorithm for subspace tracking

    Publication Year: 1992, Page(s):1535 - 1541
    Cited by:  Papers (173)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (492 KB)

    In certain signal processing applications it is required to compute the null space of a matrix whose rows are samples of a signal with p components. The usual tool for doing this is the singular value decomposition. However, the singular value decomposition has the drawback that it requires O(p3) operations to recompute when a new sample arrives. It is shown t... View full abstract»

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  • Tracking of a time-varying acoustic impulse response by an adaptive filter

    Publication Year: 1992, Page(s):1285 - 1294
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (748 KB)

    A simple model is set up describing roughly the change of an acoustic impulse response if an object moves through a room. From this model, specifications can be formed regarding the length and convergence rate of an adaptive filter trying to track this change. It is shown that in many practical applications an adaptive filter would not perform better than the optimal fixed filter. Next, a more acc... View full abstract»

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  • Calculating the FHT in hardware

    Publication Year: 1992, Page(s):1341 - 1353
    Cited by:  Papers (21)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1004 KB)

    A parallel, pipelined architecture for calculating the fast Hartley transform (FHT) is discussed. Hardware implementation of the FHT introduces two challenges: retrograde indexing and data scaling. A novel addressing scheme that permits the fast computation of FHT butterflies is proposed, and a hardware implementation of conditional block floating point scaling that reduces error due to data growt... View full abstract»

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  • Time-domain filter bank analysis: a new design theory

    Publication Year: 1992, Page(s):1412 - 1429
    Cited by:  Papers (123)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1440 KB)

    The authors present a new time-domain approach for the analysis and design of a broad class of general analysis/synthesis systems based on M-band filter banks. They derive a set of time-domain conditions for reconstruction which can be used directly in a filter bank design procedure. The general and unrestricted nature of this framework allows for the design of many useful banks. In addit... View full abstract»

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Aims & Scope

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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Meet Our Editors

Editor-in-Chief
Sergios Theodoridis
University of Athens