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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 6 • Date December 1986

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Displaying Results 1 - 25 of 42
  • [Front cover and table of contents]

    Publication Year: 1986 , Page(s): 0
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    Freely Available from IEEE
  • Recursive estimation and time-series analysis

    Publication Year: 1986 , Page(s): 1678
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (150 KB)  

    First Page of the Article
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  • [Back cover]

    Publication Year: 1986 , Page(s): c4
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    Freely Available from IEEE
  • On the complex residue arithmetic system (CRNS)

    Publication Year: 1986 , Page(s): 1675 - 1677
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (320 KB)  

    Recently, a number of papers has been published on the subject of performing complex arithmetic in the residue number system. Methods have been proposed which reduce the arithmetic complexity of a complex multiply by more than 50 percent. In this correspondence it is shown that these methods achieve Winograd's lower bound, and how these efficient mappings can be derived in terms of polynomial ring... View full abstract»

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  • An integrated-circuit-based speech recognition system

    Publication Year: 1986 , Page(s): 1465 - 1472
    Cited by:  Papers (10)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (856 KB)  

    A high-performance, flexible, and potentially inexpensive speech recognition system is described. The system is based on two special-purpose integrated circuits that perform the speech recognition algorithms very efficiently. One of these integrated circuits is the front-end processor, which computes spectral coefficients from incoming speech. The second integrated circuit computes a dynamic-time-... View full abstract»

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  • Modifications to the McClellan, Parks, and Rabiner computer program for designing higher order differentiating FIR filters

    Publication Year: 1986 , Page(s): 1671 - 1674
    Cited by:  Papers (22)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (400 KB)  

    Simple modifications to the McClellan, Parks, and Rabiner linear phase finite impulse response (FIR) filter design program are suggested to allow the design of an nth-order differentiating FIR filter of arbitrary length for any n. Two illustrative examples are also provided. View full abstract»

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  • Hyperbolic householder transformations

    Publication Year: 1986 , Page(s): 1589 - 1602
    Cited by:  Papers (43)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1576 KB)  

    A class of transformation matrices, analogous to the Householder matrices, is developed, with a nonorthogonal property designed to permit the efficient deletion of data from least-squares problems. These matrices, which we term hyperbolic Householder, are shown to effect deletion, or simultaneous addition and deletion, of data with much less sensitivity to rounding errors than for techniques based... View full abstract»

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  • On averaging burg spectral estimators for segments

    Publication Year: 1986 , Page(s): 1473 - 1484
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1120 KB)  

    The Burg spectral estimator (BSE) exhibits better peak resolution than conventional linear spectral estimators, particularly for short data records. Based on this property, the quality of the BSE is investigated with the available data record segmented and the relevant parameters or functions associated with each segment averaged. Averaging of autoregressive coefficients, reflection coefficients, ... View full abstract»

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  • Adaptive equalizer using finite-bit power-of-two quantizer

    Publication Year: 1986 , Page(s): 1603 - 1611
    Cited by:  Papers (24)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (744 KB)  

    The stochastic gradient algorithm using a simplified arithmetic is analyzed in this paper. A power-of-two quantizer is used for the input of the multiplier to reduce the multiplication to at most a simple shift. In spite of its simple implementation, the performance is shown to be comparable to the classical LMS algorithm. A linearized approximation to the quantizer is first derived, followed by t... View full abstract»

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  • Adaptive detection of transient signals

    Publication Year: 1986 , Page(s): 1410 - 1418
    Cited by:  Papers (16)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (784 KB)  

    The paper discusses the problem of detecting transient signals of unknown waveforms in white Gaussian noise. The signals are modeled as impulse responses of rational transfer functions with unknown parameters. A modified generalized likelihood ratio test (MGLRT) is proposed and its statistical properties are analyzed for both known and unknown noise variances. The MGLRT involves constrained maximu... View full abstract»

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  • A self-orthogonalizing efficient block adaptive filter

    Publication Year: 1986 , Page(s): 1573 - 1582
    Cited by:  Papers (13)  |  Patents (7)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1088 KB)  

    This paper deals with the development of a unique self-orthogonalizing block adaptive filter (SOBAF) algorithm that yields efficient finite impulse response (FIR) adaptive filter structures. Computationally, the SOBAF is shown to be superior to the least mean squares (LMS) algorithm. The consistent convergence performance which it provides lies between that of the LMS and the recursive least squar... View full abstract»

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  • The computation of line spectral frequencies using Chebyshev polynomials

    Publication Year: 1986 , Page(s): 1419 - 1426
    Cited by:  Papers (71)  |  Patents (15)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (832 KB)  

    Line spectral frequencies provide an alternate parameterization of the analysis and synthesis filters used in linear predictive coding (LPC) of speech. In this paper, a new method of converting between the direct form predictor coefficients and line spectral frequencies is presented. The system polynomial for the analysis filter is converted to two even-order symmetric polynomial with interlacing ... View full abstract»

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  • An adaptive IIR structure for sinusoidal enhancement, frequency estimation, and detection

    Publication Year: 1986 , Page(s): 1380 - 1390
    Cited by:  Papers (76)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (992 KB)  

    An adaptive IIR structure for processing a sinusoidal signal in broad-band noise is introduced. The structure contains three adaptive processors, each of which is computationally very simple. Useful features of the structure include enhancement, frequency estimation, and detection. View full abstract»

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  • Bandpass and bandstop recursive filters with low sensitivity

    Publication Year: 1986 , Page(s): 1485 - 1492
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (768 KB)  

    In a sharp cutoff bandpass or bandstop recursive digital filter, the poles and zeros are expected to be clustered around the band edges, and in close proximity of the unit circle in the z-plane. Implementation of such filters with low sensitivity could be achieved by cascading second-order sections, with appropriate low sensitivity structures chosen for individual sections depending on the locatio... View full abstract»

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  • WMMSE design of digital filter banks with specified composite response

    Publication Year: 1986 , Page(s): 1529 - 1541
    Cited by:  Papers (1)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1408 KB)  

    A new method for designing uniform and nonuniform digital filter banks with a specified composite response is presented. The composite response of the filter bank can be met either exactly or to within a given tolerance. We focus on filter banks in which the individual filters are finite impulse response (FIR) digital filters of possibly nonequal length, although the new method is applicable even ... View full abstract»

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  • Parametric projection filter for image and signal restoration

    Publication Year: 1986 , Page(s): 1643 - 1653
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1608 KB)  

    A class of linear algebraic restoration filters is derived for the linear degradation model with additive noise. The filters are based on an optimization criterion involving two effects: first, the error in the restored noiseless image compared to the original image, and second, the energy of additive noise passed through the restoration. These two effects can be balanced with a scalar parameter. ... View full abstract»

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  • Fast Hadamard transform based on a simple matrix factorization

    Publication Year: 1986 , Page(s): 1666 - 1667
    Cited by:  Papers (26)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (224 KB)  

    This correspondence presents a simple factorization of the Hadamard matrix which is used to develop a fast algorithm for the Hadamard transform. The matrix decomposition is of the form of the Kronecker products of identity matrices and successively lower order Hadamard matrices. This decomposition leads very clearly to a sparse-matrix factorization of the Hadamard matrix. View full abstract»

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  • On the number of signals resolvable by a uniform linear array

    Publication Year: 1986 , Page(s): 1361 - 1375
    Cited by:  Papers (51)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1448 KB)  

    An algebraic limitation on the maximum number of directions of arrival of plane waves that can be resolved by a uniform linear sensor array is studied. Achievable lower and upper bounds are derived on that number as a function of the number of elements in the array, number of snapshots, and the rank of the source sample-correlation matrix. The signals are assumed narrow-band and of identical and k... View full abstract»

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  • On an adaptive noise cancellation application for radar

    Publication Year: 1986 , Page(s): 1654 - 1655
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (208 KB)  

    The purpose of this correspondence is to introduce a novel application of adaptive noise cancellation related to a class of radar signals. View full abstract»

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  • A new algorithm for real-time median filtering

    Publication Year: 1986 , Page(s): 1674 - 1675
    Cited by:  Papers (28)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (232 KB)  

    An algorithm is presented to determine the median of a given set of samples W. The median is determined by making r comparisons of the elements of the "m-array," where r is the number of bits in the radix-2 representation of the largest element of W. The m-array has 2relements, and mkequals the number of samples of W greater than or equal to (k - 1). The method is suitable to... View full abstract»

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  • On the relation between the maximum entropy probability density function and the autoregressive model

    Publication Year: 1986 , Page(s): 1659 - 1661
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (368 KB)  

    The problem of maximizing the entropy of an n-variate random vector subject to constraints on the first p + 1 autocovariance terms is examined. It is shown that the maximum is achieved by the Gaussian autoregressive process of order p satisfying the autocovariance constraints. This solution provides Burg's theorem about the maximum entropy spectral density as a special case. View full abstract»

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  • Approximate realization algorithms for truncated impulse response data

    Publication Year: 1986 , Page(s): 1583 - 1588
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (464 KB)  

    The objectives of this paper are: 1) to present a novel derivation of an approximate realization algorithm using singular value decomposition proposed by Kung; 2) to explain the difference between two versions of this algorithm which have been used interchangeably in the literature; and 3) to introduce a new version of this algorithm which exhibits the accuracy of the original algorithm and the th... View full abstract»

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  • Speech transformations based on a sinusoidal representation

    Publication Year: 1986 , Page(s): 1449 - 1464
    Cited by:  Papers (49)  |  Patents (39)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1528 KB)  

    In this paper a new speech analysis/synthesis technique is presented which provides the basis for a general class of speech transformations including time-scale modification, frequency scaling, and pitch modification. These modifications can be performed with a time-varying change, permitting continuous adjustment of a speaker's fundamental frequency and rate of articulation. The method is based o... View full abstract»

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  • A convergence analysis of a passive underwater tracking system with nonlinear feedback

    Publication Year: 1986 , Page(s): 1401 - 1409
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (832 KB)  

    The basic signal processing structure of a new passive underwater tracking system incorporating nonlinear feedback [1], [2] is modeled, and its ability to converge to an unbiased estimate of target range is examined. Target measurements are derived using difference of arrival times between passive sensor systems geometrically separated. The range tracking system utilizes a nonlinear signal process... View full abstract»

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  • Zero-tracking adaptive filters

    Publication Year: 1986 , Page(s): 1566 - 1572
    Cited by:  Papers (18)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (712 KB)  

    A new type of adaptive filter is proposed which can directly estimate and track its own zeros. The adaptation algorithm adapts the zeros of the filter and hence, indirectly, the filter coefficients. To first order in the adaptation parameter, the new algorithm is equivalent to the usual LMS algorithm, and thus it shares the same convergence properties with the latter. The cases of adaptive predict... 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