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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 1 • Date Jan. 1989

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Displaying Results 1 - 24 of 24
  • Forward/backward spatial smoothing techniques for coherent signal identification

    Publication Year: 1989 , Page(s): 8 - 15
    Cited by:  Papers (275)  |  Patents (10)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (667 KB)  

    A spatial smoothing scheme is further investigated in the context of coherent signal classification. It is shown that by making use of a set of forward and complex conjugated backward subarrays simultaneously, it is always possible to estimate any K directions of arrival using at most 3K/2 sensor elements. This is achieved by creating a smoothed array output covariance matrix that is structurally identical to a covariance matrix in some noncoherent situation. By incorporating eigenstructure-based techniques on this smoothed covariance matrix, it then becomes possible to correctly identify all directions of arrival irrespective of their correlation.<> View full abstract»

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  • Line-array beamforming using linear prediction for aperture interpolation and extrapolation

    Publication Year: 1989 , Page(s): 16 - 30
    Cited by:  Papers (39)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1633 KB)  

    It is demonstrated that linear prediction can be successfully utilized in conjunction with otherwise conventional narrow-band line-array beamforming for aperture extrapolation (APEX), where there are significant SNR (signal-to-noise ratio) and resolution advantages compared to the conventional approach, aperture interpolation, where faulty sensor data are replaced, and a combined technique (ALPINEX). The performance of all three methods is investigated in connection with both simulated and real data. The APEX technique is shown to be amenable to a simple theoretical treatment. Further, the approach is demonstrated to offer a practical performance comparable to the well-known high-resolution technique, MUSIC, when the observation time is short.<> View full abstract»

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  • Entropy-constrained vector quantization

    Publication Year: 1989 , Page(s): 31 - 42
    Cited by:  Papers (298)  |  Patents (24)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1315 KB)  

    An iterative descent algorithm based on a Lagrangian formulation for designing vector quantizers having minimum distortion subject to an entropy constraint is discussed. These entropy-constrained vector quantizers (ECVQs) can be used in tandem with variable-rate noiseless coding systems to provide locally optimal variable-rate block source coding with respect to a fidelity criterion. Experiments on sampled speech and on synthetic sources with memory indicate that for waveform coding at low rates (about 1 bit/sample) under the squared error distortion measure, about 1.6 dB improvement in the signal-to-noise ratio can be expected over the best scalar and lattice quantizers when block entropy-coded with block length 4. Even greater gains are made over other forms of entropy-coded vector quantizers. For pattern recognition, it is shown that the ECVQ algorithm is a generalization of the k-means and related algorithms for estimating cluster means, in that the ECVQ algorithm estimates the prior cluster probabilities as well. Experiments on multivariate Gaussian distributions show that for clustering problems involving classes with widely different priors, the ECVQ outperforms the k-means algorithm in both likelihood and probability of error.<> View full abstract»

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  • On the probability density function of the LMS adaptive filter weights

    Publication Year: 1989 , Page(s): 43 - 56
    Cited by:  Papers (26)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (962 KB)  

    The joint probability density function of the weight vector in least-mean-square (LMS) adaptation is studied for Gaussian data models. An exact expression is derived for the characteristic function of the weight vector at time n+1, conditioned on the weight vector at time n. The conditional characteristic function is expanded in a Taylor series and averaged over the unknown weight density to yield a first-order partial differential-difference equation in the unconditioned characteristic function of the weight vector. The equation is approximately solved for small values of the adaption parameter and the weights are shown to be jointly Gaussian with time-varying mean vector and covariance matrix given as the solution to well-known difference equations for the weight vector mean and covariance matrix. The theoretical results are applied to analyzing the use of the weights in detection and time delay estimation. Simulations that support the theoretical results are also presented.<> View full abstract»

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  • Split-radix algorithms for length-p/sup m/ DFT's

    Publication Year: 1989 , Page(s): 57 - 64
    Cited by:  Papers (19)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (776 KB)  

    The split-radix algorithm for the discrete Fourier transform (DFT) of length-2/sup m/ is considered. First, the reason why the split-radix algorithm is better than any single-radix algorithm on length-2/sup m/ DFTs is given. Then, the split-radix approach is generalized to length-p/sup m/ DFTs. It is shown that whenever a radix-p/sup 2/ outperforms a radix-p algorithm, then a radix-p/p/sup 2/ algorithm will outperform both of them. As an example, a radix-3/9 algorithm is developed for length-3/sup m/ DFTs.<> View full abstract»

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  • Detail-preserving ranked-order based filters for image processing

    Publication Year: 1989 , Page(s): 83 - 98
    Cited by:  Papers (116)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (2693 KB)  

    The theoretical analysis of multistage median filters is developed. It is shown that multistage median filters are a combination of max/median and min/median filters. Since multistage median filters belong to the class of two-dimensional stack filters, they have threshold decomposition attributes making their theoretical analysis simple. Statistical threshold decomposition is applied to derive the statistical characteristics of these filters, and the results are used to evaluate the performance of these two types of multistage filters. Finally, a quantitative and qualitative comparison of the multistage filters and of other efficient detail-preserving filters is presented. The comparisons are made using the mean-squared-error and the mean-absolute-error criteria.<> View full abstract»

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  • An efficient algorithm for two-dimensional autoregressive spectrum estimation

    Publication Year: 1989 , Page(s): 106 - 117
    Cited by:  Papers (16)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (958 KB)  

    An algorithm similar to the Burg algorithm is given for autoregressive estimation in one dimension. Coefficients of the filter used to decorrelate the signal are found recursively from lower-order filters. The coefficients needed to get the larger-sized filters can be found at each stage of the recursion from the correlations of lower-order filter outputs. Thus, the resulting estimate does not assume knowledge of ensemble averages. It offers a computational saving of 25% over the commonly used method, in which an estimate of the correlation matrix is first computed and then inverted.<> View full abstract»

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  • A test procedure for 2-D discrete scattering Hurwitz polynomials

    Publication Year: 1989 , Page(s): 118 - 120
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (300 KB)  

    A procedure to test whether a given two-variable polynomial is a discrete scattering Hurwitz polynomial is described. In this procedure, Schur-Cohn matrices associated with the given polynomial are used to test the set of necessary and sufficient conditions established by S. Basu and A. Fettweis (Circuits, Syst., Signal Processing, vol.3, no.2, p.225-42, 1984). The procedure is illustrated by examples.<> View full abstract»

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  • Classification of natural textures by means of two-dimensional orthogonal masks

    Publication Year: 1989 , Page(s): 125 - 128
    Cited by:  Papers (5)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (691 KB)  

    An approach to texture classification in which each local neighborhood is transformed reversibly into a minimal set of uncorrelated features is described. These features are found to represent local average, as well as first- and second-order derivative information of the texture. Comparison to other techniques of similar complexity, using identical classification procedures, shows the superiority of the proposed approach. Moreover, this approach leads directly to a real-time parallel implementation.<> View full abstract»

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  • An efficient FIR adaptive filter using DPCM and the sign algorithm

    Publication Year: 1989 , Page(s): 128 - 133
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (575 KB)  

    An adaptive filter structure that requires zero multiplications for its implementation is introduced. The primary input signal is quantized using differential pulse code modulation (DPCM), and the DPCM outputs are processed by a conventional adaptive filter. The filter coefficients are updated using the sign algorithm. Under the assumption that the primary and reference input signals to the adaptive filter are zero-mean, wide-sense stationary, and Gaussian, theoretical results for the mean and mean-squared behaviour of the filter are also derived. A simulation example demonstrating the usefulness of the filter, as well as verifying the theoretical results and comparing the multiplication-free adaptive digital filter to the sign algorithm, is also presented.<> View full abstract»

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  • Coherent signal classification using symmetry considerations

    Publication Year: 1989 , Page(s): 135 - 138
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (430 KB)  

    The problem of locating the directions of arrival of a set of coherent/correlated signals in an arbitrary noise field using multiple sensor elements is considered. It is shown that by making use of a symmetric array, it is possible to perform a preprocessing on the array output correlations to generate a set of new quantities that are functionally similar to the actual correlations in an incoherent environment. The orthogonality of the direction vectors corresponding to the true angles of arrival and a set of eigenvectors of the Toeplitz matrix generated from these new quantities is then made use of in estimating the actual multipath arrival angles.<> View full abstract»

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  • A comparison of two quantized state adaptive algorithms

    Publication Year: 1989 , Page(s): 138 - 143
    Cited by:  Papers (23)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (699 KB)  

    Quantized state (QS) adaptive algorithms reduce the numerical complexity and dynamic range requirements of least mean squares (LMS) adaptation by replacing multiplications with shifts, bit comparisons, or table lookups. A theoretical foundation with which to distinguish two primary QS algorithm forms and to predict which algorithm is most appropriate in a given context is presented. An extended Lyapunov approach is used to derive a persistence of excitation (PE) condition which guarantees linear stability of the quantized error (QE) form. Averaging theory is then used to derive PE condition which guarantees exponential stability of the quantized regressor (QReg) form. Failure to meet this latter condition (which is not equivalent to the spectral richness PE condition for LMS) can result in exponential instability. The QE and QReg algorithms are then compared in terms of conditions for stability, convergence properties of the prediction and parameter errors, convergence rates, and steady-state errors.<> View full abstract»

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  • Source location bias in the coherently focused high-resolution broad-band beamformer

    Publication Year: 1989 , Page(s): 143 - 145
    Cited by:  Papers (17)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (320 KB)  

    A simple expression for the source located (angle of arrival) bias is developed for H. Wang and M. Kaveh's (ibid., vol.33, no.4, p. 823-31, 1985) focused broadband beamformer. It is shown to depend on the source temporal frequency spectrum only through its centroidal frequency. The bias is zero if the angle of arrival is aligned with the primary steering angle (i.e. focusing angle), or, more interestingly, if the source centroidal frequency equals the focusing frequency.<> View full abstract»

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  • Window functions represented by B-spline functions

    Publication Year: 1989 , Page(s): 145 - 147
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (277 KB)  

    Characteristics of window functions provided by B-spline functions are analyzed in comparison to other known windows. For the same highest-sidelobe level, those windows have a mainlobe bandwidth between those of Gaussian and cos/sup a/ x windows are as good as those of cos/sup a/ x ones in the sidelobe falloff.<> View full abstract»

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  • Significance of initial interpolation in band-limited signal interpolation

    Publication Year: 1989 , Page(s): 151 - 152
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (204 KB)  

    An improved version of the Papoulis algorithm for bandlimited signal interpolation is presented. This algorithm uses the concept of initial interpolation. The justification for initial interpolation is developed only through experimental studies. It is shown that the performance of the interpolation scheme depends on the number and distribution of the known data samples.<> View full abstract»

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  • Estimating two-dimensional angles of arrival in coherent source environment

    Publication Year: 1989 , Page(s): 153 - 155
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (303 KB)  

    The concept of spatial smoothing is first extended to a rectangular planar array and a two-dimensional (2-D) search function is formed to estimate the source of directions. To avoid performing a 2-D search, an approach based on two one-dimensional (1-D) searches is also discussed. This approach uses rows and columns of the rectangular array to perform 1-D searches. To match the data obtained, a 2-D verification is then performed. Computer simulation results for both approaches, based on the MUSIC method, are presented.<> View full abstract»

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  • A VLSI implementation of a correlator/digital-filter based on distributed arithmetic

    Publication Year: 1989 , Page(s): 156 - 160
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (376 KB)  

    The design of a VLSI chip which applies the ideas of distributed arithmetic to a nonrecursive digital filter is described. The main features of this design are very high precision and a high degree of flexibility, which allows one chip to implement a large number of quite different digital filters and correlators View full abstract»

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  • Comparison of noisy speech enhancement algorithms in terms of LPC perturbation

    Publication Year: 1989 , Page(s): 121 - 125
    Cited by:  Papers (5)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (408 KB)  

    A number of noisy speech enhancement algorithms are experimentally compared in terms of linear predictive coding (LPC) perturbations. The enhancement algorithms considered are simple spectral subtraction, spectral oversubtraction with use of a spectral floor, spectral subtraction with residual noise removal, and time-domain and frequency-domain adaptive minimum mean-square-error filtering. LPC perturbations considered are LPC cepstral distance, log likelihood ratios, and weighted likelihood ratio View full abstract»

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  • The generalized H-lattice filter and estimation of the ARMA models

    Publication Year: 1989 , Page(s): 147 - 151
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (400 KB)  

    A class of lattice filters, the generalized H-lattice filters is introduced. The generalized H-lattice filters can represent the autoregressive moving-average (ARMA) models and they can be used for the modeling and analysis of the ARMA stochastic process View full abstract»

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  • Signal reconstruction from noisy partial information of its transform

    Publication Year: 1989 , Page(s): 65 - 72
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (752 KB)  

    The usual assumption that the available partial information is noiseless is replaced by a more realistic statistical model which compensates for the presence of noise. The signal reconstruction is thus viewed as a parameter estimation problem, for which the EM iterative algorithm of A.P. Dempster, N.M. Laird, and D.B. Rubin (J. Roy. Stat. Soc., B, vol.39, p.1-37, 1977) is especially suitable. The posterior probability of the signal increases from iteration to iteration, till the signal converges to a stationary point of the posterior distribution. Each iteration involves one transformation followed by an inverse transformation (usually discrete Fourier transformation (DFT) and inverse DFT). Algorithms for reconstruction of both one- and two-dimensional signals from their spectral magnitude, spectral phase, or modified short time Fourier transform are typical examples of the proposed scheme View full abstract»

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  • Signal detectors for deformable hydrophone arrays

    Publication Year: 1989 , Page(s): 1 - 7
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (504 KB)  

    Passive signal detection using a deformable hydrophone array, in which the position of each sensor deviates randomly from a nominal or estimated location, is examined. A general detector, which adds the power of R arbitrary linear combinations of all hydrophone outputs, is proposed. This configuration permits division of the array into arbitrary sections or partitions. When a single section is used, the detector reduces to the standard form used with no uncertainty. Linear filters in the processor are chosen to optimize the deflection criterion. Using the same criterion, performance of optimal and several suboptimal detectors is examined as a function of uncertainty variance. Numerical results presented for a large planar array confirm the existence of practical detectors that are nearly optimal and clearly superior to the standard detector in this application View full abstract»

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  • Prony, Pisarenko, and the matrix pencil: a unified presentation

    Publication Year: 1989 , Page(s): 133 - 134
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (504 KB)  

    In the past, the Prony and Pisarenko algorithms have been widely used in the pole zero identification of linear systems as well as in the problem of extracting harmonics embedded in noise. These techniques have also been applied successfully to the direction of arrival (DOA) problem. It is shown that these algorithms are equivalent to a recently proposed DOA estimation technique called the matrix pencil approach View full abstract»

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  • Estimating the basis functions of the Karhunen-Loeve transform

    Publication Year: 1989 , Page(s): 99 - 105
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (548 KB)  

    A procedure for estimating the basis functions of the Karhunen-Loeve (KL) transform and the associated eigenvalues is presented. This estimator differs from previous results in the literature in that the continuous KL basis functions are estimated, allowing the direct evaluation of the effects of sampling on estimator performance. Theorems describing the convergence of the estimated eigensolution are given. These theorems provide a sound theoretical foundation for the estimation procedure and add insight into the quality of the estimates and the effects of sampling. The previous result of estimating the KL transform using the eigensolution of the sample covariance matrix can be formulated as a special case of the proposed procedure, proving the convergence of this previous result. An example that illustrates the procedure and provides a comparison to previous results is presented View full abstract»

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  • Maximum likelihood estimation of a class of non-Gaussian densities with application to Ip deconvolution

    Publication Year: 1989 , Page(s): 73 - 82
    Cited by:  Papers (16)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (596 KB)  

    The properties of the maximum likelihood estimator of the generalized p-Gaussian (GPG) probability density function from N independent identically distributed samples is investigated, especially in the context of the deconvolution problem under GPG white noise. Specifically, the properties in the estimator are first described independently of the application. Then the solution of the above-mentioned deconvolution problem is obtained as the solution of a minimum norm problem in an lp normed space. It is shown that such minimum norm solution is the maximum-likelihood estimate of the system function parameters and that such an estimate is unbiased, with the lower bound of the variance of the error equal to the Cramer-Rao lower bound, and the upper bound derived from the concept of a generalized inverse. The results are illustrated by computer simulations 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