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IEEE Transactions on Acoustics, Speech, and Signal Processing

Issue 1 • Date Jan. 1990

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Displaying Results 1 - 25 of 31
  • Comments on "An axiomatic construction of quadratic NAR power estimators for adaptive detection" [with reply]

    Publication Year: 1990, Page(s):173 - 175
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (347 KB)

    An unstated axiom in the abovementioned correspondence by M.H. El-Ayadi (see ibid., vol.ASSP-35, p.1347-50, Sept.1987) on quadratic noise estimators requires the kernel matrix to be nonnegative definite. The commenter shows that this assumption is not required for the noise-alone-reference (NAR) estimator; it limits the performance of the generalized NAR estimator in the sense of a minimum-varianc... View full abstract»

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  • Comments on "On the shift property of DCTs and DSTs

    Publication Year: 1990, Page(s):186 - 188
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (182 KB)

    In the abovementioned correspondence P. Yip and K.R. Rao (see ibid., vol.ASSP-35, p.404-6, Mar.1987) presented a shift property for even discrete cosine transforms (EDCT) and sine transforms (EDST). The commenter points out that the odd counterparts (ODCT and ODST) also exist in the family. He extends Yip and Rao's results to these odd triangular transforms and makes several corrections to their w... View full abstract»

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  • An encoding procedure and a decoding procedure for a new modified Huffman code

    Publication Year: 1990, Page(s):128 - 136
    Cited by:  Papers (4)  |  Patents (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (860 KB)

    The conventional Huffman table, which stores a set of code words with different lengths, is difficult to construct. An easily constructed table that includes a code-length array and an encoding mapping array can accompany an encoding procedure to replace the conventional Huffman table. The code-length array records the number of code words of the same length, and the index of this array represents... View full abstract»

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  • Seismic deconvolution by multipulse methods

    Publication Year: 1990, Page(s):156 - 160
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (516 KB)

    A multiple method used in speech coding is applied to the seismic deconvolution problem. The advantage of this method is that the source wavelet and reflectivity series representing the layered Earth structure are simultaneously estimated. Results of this investigation show that the method shows promise for all-pole models of the input wavelet. An application of the method to actual seismic data i... View full abstract»

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  • String processing on the hypercube

    Publication Year: 1990, Page(s):160 - 164
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (500 KB)

    Parallel algorithms are given for solving some string-comparison problems on the hypercube. These algorithms are widely applicable to the problems of speech and signal processing. For strings x and y with length (x)=m, length (y)=n, and assuming nm, the authors show that the substring problem can be solved in O(... View full abstract»

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  • A high-resolution postbeamforming method based on semiinfinite linear optimization

    Publication Year: 1990, Page(s):16 - 22
    Cited by:  Papers (2)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (596 KB)

    A method for enhancing the bearing resolution obtained by conventional beamforming is described. Using the beamformer power outputs, the beam response functions, and optionally a background-noise spatial power spectral density (PSD) as input, the method gives a point-mass estimate of the signal PSD as function of bearing. The estimate is defined from the solution of a semiinfinite linear optimizat... View full abstract»

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  • Upper and lower bounds of the misadjustment in the LMS algorithm

    Publication Year: 1990, Page(s):164 - 166
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (200 KB)

    The misadjustment is a dimensionless measure of the difference between the optimal Weiner performance and the actual performance of the LMS (least-mean-squares) algorithm. This difference is attributed to the gradient noise in the adaptive process. Present estimates relate the misadjustment to the convergence gain and the trace of the autocorrelation matrix. The authors establish tighter lower bou... View full abstract»

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  • Fixed-point error analysis of stochastic gradient adaptive lattice filters

    Publication Year: 1990, Page(s):70 - 80
    Cited by:  Papers (11)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (776 KB)

    A theoretical analysis is presented of the stochastic gradient adaptive lattice filter used as a linear, one-step predictor, when the effects of finite-precision arithmetic are taken into account. Only the fixed-point implementation is considered. Both the unnormalized and normalized adaptation algorithms are analyzed. Expressions of the steady-state mean-squared values of the accumulated numerica... View full abstract»

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  • Kernel-factorization deconvolution method

    Publication Year: 1990, Page(s):166 - 168
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (232 KB)

    The deconvolution problem for a cycle convolution equation with the input (or the deconvolution kernel) spectrum containing zeros is discussed. It is shown that through factorization of the kernel the impulse can be obtained by combined use of the two conventional methods: the Z-transform division method and DFT (discrete Fourier transform) division method, if the length of the impulse re... View full abstract»

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  • Predictive trellis coded quantization of speech

    Publication Year: 1990, Page(s):46 - 55
    Cited by:  Papers (38)  |  Patents (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (908 KB)

    Trellis-coded quantization (TCQ) is incorporated into a predictive coding structure for encoding sampled speech. The modest complexity of the resulting structure is seen to be a direct consequence of the TCQ formulation. Simulation results are presented for systems using fixed-prediction/fixed-residual encoding, fixed-prediction/adaptive-residual encoding, and adaptive-prediction/adaptive-residual... View full abstract»

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  • Pipeline architecture for block adaptive LS FIR filtering and prediction

    Publication Year: 1990, Page(s):81 - 90
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (692 KB)

    Consideration is given to the development of a highly parallel, block-type, order-recursive algorithm for least-squares finite-impulse-response (LS FIR) filter identification and prediction. The computation of the required reflection coefficients is achieved by a technique that does not involve inner vector product operations. Its origin can be traced to the classical Schur algorithm for Toeplitz ... View full abstract»

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  • Beamforming in the presence of correlated arrivals using structured correlation matrix

    Publication Year: 1990, Page(s):1 - 15
    Cited by:  Papers (22)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (780 KB)

    Expressions are derived for the SNR (signal-to-noise-ratio) at the output of two antenna array processors in the presence of correlated arrivals, and the effect on the output SNR of the magnitude and the phase of the correlation, the number of elements in the array, the direction of the level of the interference source, and the level of the uncorrelated noise is investigated. The two processes con... View full abstract»

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  • New FFT structures based on the Bruun algorithm

    Publication Year: 1990, Page(s):188 - 191
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (288 KB)

    In some signal processing applications, the input data are real. In this case, the Bruun algorithm for computation of the discrete Fourier transform (DFT) is attractive. The author offers a pipeline and a recirculated shuffle network implementation of the Bruun algorithm. The implementation of the parallel pipeline and recirculated FFT structures is based on the modified perfect shuffle network View full abstract»

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  • Optimal fault-tolerant signal detection

    Publication Year: 1990, Page(s):179 - 180
    Cited by:  Papers (5)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (164 KB)

    G.G.L. Meyer and H.L. Weinert (see ibid., vol.ASSP-34, p.973-8, Aug. 1986) use the majority rule for fault-tolerant detection. The authors extend this and examine a redundant system with a general voting rule. They present an optimal design for each individual processor and the optimal selection of the k-out-of-M voting rule. They give an analytic solution to the optimal voting p... View full abstract»

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  • DCT algorithms for VLSI parallel implementations

    Publication Year: 1990, Page(s):121 - 127
    Cited by:  Papers (32)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (556 KB)

    Two algorithms are presented for computing the discrete cosine transform (DCT) on existing VLSI structures. First, it is shown that the N-point DCT can be implemented on the existing systolic architecture for the N-point discrete Fourier transform (DFT) by introducing some modifications. Second, a new prime factor DCT algorithm is presented for the class of DCTs of length ... View full abstract»

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  • On asymmetrical performance of discrete cosine transform

    Publication Year: 1990, Page(s):154 - 156
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (188 KB)

    The discrete cosine transform (DCT) is considered as a suboptimum transform for many practical source-coding applications. Autoregressive order 1 [AR(1)] source models are good first approximations to several natural signals. It is known that the performance of the DCT depends on the sign of the autocorrelation coefficient of the AR(1) source. The authors explain theoretically this asymmetrical pe... View full abstract»

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  • Phase error estimation and correction in acoustic microscopy

    Publication Year: 1990, Page(s):171 - 173
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (204 KB)

    Consideration is given to the phase error estimation and correction problem in tomographic acoustic microscopy. In a recent paper, Z.C. Lin and G. Wade (ibid., vol.ASSP-35, p.1037-45, July 1987) modeled the phase error as a constant and proposed a single-sideband technique for its correction. The present authors point out that the phase error is not a constant, and that the single-sideband techniq... View full abstract»

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  • VLSI median filters

    Publication Year: 1990, Page(s):145 - 153
    Cited by:  Papers (94)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (812 KB)

    Median filters have been proposed for the analysis of speech data and in image processing to enhance the data by smoothing the signal and removing noise. Many designs for median filters have been suggested in the literature. The author reviews these designs (most originally given as software filters) and compare them on the basis of suitability for implementation in VLSI. Each design is analyzed i... View full abstract»

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  • An approach to co-channel talker interference suppression using a sinusoidal model for speech

    Publication Year: 1990, Page(s):56 - 69
    Cited by:  Papers (30)  |  Patents (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1092 KB)

    The technique fits a sinusoidal model to additive vocal speech segments so that the least-mean-squared error between the model and the summed waveforms is obtained. Enhancement is achieved by synthesizing a waveform from the sine waves attributed to the desired speaker. Least-squares estimation is applied to obtain sine-wave amplitudes and phases of both talkers, based on either a priori sine-wave... View full abstract»

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  • An overview of the SPHINX speech recognition system

    Publication Year: 1990, Page(s):35 - 45
    Cited by:  Papers (96)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1016 KB)

    A description is given of SPHINX, a system that demonstrates the feasibility of accurate, large-vocabulary, speaker-independent, continuous speech recognition. SPHINX is based on discrete hidden Markov models (HMMs) with LPC- (linear-predictive-coding) derived parameters. To provide speaker independence, knowledge was added to these HMMs in several ways: multiple codebooks of fixed-width parameter... View full abstract»

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  • A Fourier-domain formula for the least-squares projection of a function onto a repetitive basis in N-dimensional space

    Publication Year: 1990, Page(s):114 - 120
    Cited by:  Papers (1)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (520 KB)

    A theorem concerning the least-squares projection of an arbitrary function onto an infinite basis of translated function is given. The theorem provides an explicit formula for the Fourier transform, of the projected function. The formula has the advantage of being valid for least-squares projections in any N-dimensional space. The expression for the projected function can be approximately... View full abstract»

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  • Adaptive beamforming for correlated signal and interference: a frequency domain smoothing approach

    Publication Year: 1990, Page(s):193 - 195
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (276 KB)

    An approach is suggested to wideband adaptive beamforming for correlated signal and interference. Unlike other approaches, such as spatial smoothing, the new approach solves the signal cancellation problem by employing the idea of frequency-domain smoothing. Advantages of frequency-domain smoothing over spatial smoothing are identified. Preliminary performance studies of a simple frequency-domain ... View full abstract»

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  • Computational aspects of quadratic signal processing

    Publication Year: 1990, Page(s):137 - 144
    Cited by:  Papers (5)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (564 KB)

    Quadratic signal processing is often computationally more intense than the linear case, and hence attention to concurrent computation is an important concern, particularly in real-time applications. The author addresses the concurrent computation issue in the context of systolic array processors. As a prototype the data processing requirements inherent in the calculation of discrete Wigner distrib... View full abstract»

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  • Adaptive image restoration with artifact suppression using the theory of convex projections

    Publication Year: 1990, Page(s):181 - 185
    Cited by:  Papers (25)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (720 KB)

    The authors develop an iterative, adaptive, space-variant restoration incorporating both regularization and artifact-suppression constraints. The algorithm is formulated using the method of projections onto convex sets (POCS). They introduce a closed-convex regularization constraint set called the partial Wiener solution set. Projection onto this set forces the solution to be equal to the Wiener s... View full abstract»

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  • Robust estimation of sinusoidal signals with colored noise using decentralized processing

    Publication Year: 1990, Page(s):91 - 104
    Cited by:  Papers (6)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1060 KB)

    A technique is developed for the estimation of the number of signals and their central frequencies using decentralized processing, when it is known a priori that the observations consist of a finite number of sinusoidal signals corrupted by an additive colored random noise process with unknown correlations. Such a noise sequence may be caused by jamming from a hostile agent. The authors' decentral... View full abstract»

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Aims & Scope

This Transactions ceased production in 1990. The current retitled publication is IEEE Transactions on Signal Processing.

Full Aims & Scope