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IEEE Transactions on Acoustics, Speech, and Signal Processing

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
    Request permission for commercial reuse | 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
  • A Kalman filtering approach to short-time Fourier analysis

    Publication Year: 1986, Page(s):1493 - 1501
    Cited by:  Papers (49)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1048 KB)

    The problem of estimating time-varying harmonic components of a signal measured in noise is considered. The approach used is via state estimation. Two methods are proposed, one involving pole-placement of a state observer, the other using quadratic optimization techniques. The result is the development of a new class of filters, akin to recursive frequency-sampling filters, for inclusion in a para... 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)
    Request permission for commercial reuse | 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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  • On averaging burg spectral estimators for segments

    Publication Year: 1986, Page(s):1473 - 1484
    Cited by:  Papers (4)
    Request permission for commercial reuse | 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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  • Zero-tracking adaptive filters

    Publication Year: 1986, Page(s):1566 - 1572
    Cited by:  Papers (18)
    Request permission for commercial reuse | 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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  • The relation between maximum likelihood estimation of structured covariance matrices and periodograms

    Publication Year: 1986, Page(s):1661 - 1662
    Cited by:  Papers (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (272 KB)

    A generalized Burg technique has been developed recently by Burg, Luenberger, and Wegner for maximum likelihood estimation of structured covariance matrices. In this correspondence, the unique solution for the positive definite estimate over a class of nonnegative definite, symmetric matrices with known eigenvectors is presented. This solution coincides with the Karhunen-Loève expansion, and ... View full abstract»

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  • An integrated-circuit-based speech recognition system

    Publication Year: 1986, Page(s):1465 - 1472
    Cited by:  Papers (15)
    Request permission for commercial reuse | 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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  • Rotational search methods for adaptive Pisarenko harmonic retrieval

    Publication Year: 1986, Page(s):1550 - 1565
    Cited by:  Papers (30)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1272 KB)

    Two adaptation algorithms for adaptive Pisarenko harmonic retrieval are described. They are derived by considering the associated minimum eigenvalue problem as an optimization problem which seeks the minimum of a quadratic cost function given a hyperspherical constraint. An iterative search procedure is used in which each search path is constrained to lie on the unit hypersphere. Computational com... 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 (4)
    Request permission for commercial reuse | 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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  • Speech transformations based on a sinusoidal representation

    Publication Year: 1986, Page(s):1449 - 1464
    Cited by:  Papers (51)  |  Patents (44)
    Request permission for commercial reuse | 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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  • Performance analysis of the LMS algorithm with a tapped delay line (two-dimensional case)

    Publication Year: 1986, Page(s):1542 - 1549
    Cited by:  Papers (17)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (552 KB)

    This paper presents exact analytical results of the LMS performances on correlated data vectors. The case addressed here is of data vectors constructed from a tapped delay line. While the results are for the two-dimensional case, they do provide indications of the effect this structure of data vectors, so common to LMS applications, has on LMS performance. They show conclusively the possibility of... View full abstract»

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  • A rearranged DFT algorithm requiring N2/6 multiplications

    Publication Year: 1986, Page(s):1658 - 1659
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (256 KB)

    We consider the problem of computing the DFT and present two reductions over the standard formula. In the special case of an N-point sequence with N = 2l, the number of multiplications per output point required by this algorithm is, at most, N/4 - 1 and, on the average, N/6 - 1. Each output point requires no more than N - 1 additions. In applications requiring only some of the output po... View full abstract»

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  • Waveform substitution techniques for recovering missing speech segments in packet voice communications

    Publication Year: 1986, Page(s):1440 - 1448
    Cited by:  Papers (79)  |  Patents (68)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1024 KB)

    Packet communication systems cannot, in general, guarantee accurate and prompt delivery of every packet. The effect of network congestion and transmission impairments on data packets is extended delay; in voice communications these problems lead to lost packets. When some speech packets are not available, the simplest response of a receiving terminal is to substitute silence for the missing speech... View full abstract»

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  • On the complex residue arithmetic system (CRNS)

    Publication Year: 1986, Page(s):1675 - 1677
    Cited by:  Papers (5)
    Request permission for commercial reuse | 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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  • WMMSE design of digital filter banks with specified composite response

    Publication Year: 1986, Page(s):1529 - 1541
    Cited by:  Papers (1)  |  Patents (2)
    Request permission for commercial reuse | 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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  • Fast Hadamard transform based on a simple matrix factorization

    Publication Year: 1986, Page(s):1666 - 1667
    Cited by:  Papers (42)
    Request permission for commercial reuse | 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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  • Adaptive microphone-array system for noise reduction

    Publication Year: 1986, Page(s):1391 - 1400
    Cited by:  Papers (143)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1072 KB)

    This paper introduces a new adaptive microphone-array system for noise reduction (AMNOR system). It is first shown that there exists a tradeoff relationship between reducing the output noise power and reducing the frequency response degradation of a microphone-array to a desired signal. It is then shown that this tradeoff can be controlled by the introduction of a fictitious desired signal. A new ... 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 (33)
    Request permission for commercial reuse | 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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  • Form-invariant linear filtering: Theory and applications

    Publication Year: 1986, Page(s):1612 - 1628
    Cited by:  Papers (13)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (2840 KB)

    The form-invariant filters are, by definition, those shift-variant filters such that their output, for any given input, turns out to be linearly scaled (implying that its "form" does not change) whenever the input is linearly scaled. In this paper the most general classes of 1- D and 2-D linear form-invariant filters are derived and their properties are discussed, together with their implementatio... View full abstract»

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  • Sequences with positive semidefinite Fourier transforms

    Publication Year: 1986, Page(s):1502 - 1510
    Cited by:  Papers (8)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (816 KB)

    A sequence is said to be positive if its Fourier transform exclusively takes on real nonnegative values as a function of frequency. Positive sequences play a prominent role in contemporary signal processing and system theory. To illustrate this point, it is well known that the factorization theorem is extensively used in studies related to wide-sense stationary random signals and linear systems. T... View full abstract»

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  • A self-orthogonalizing efficient block adaptive filter

    Publication Year: 1986, Page(s):1573 - 1582
    Cited by:  Papers (12)  |  Patents (8)
    Request permission for commercial reuse | 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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  • Bias in the cross spectrum and time delay estimates due to misalignment

    Publication Year: 1986, Page(s):1662 - 1665
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (440 KB)

    The effect of misalignment in the weighted, overlapped segment averaging method used for estimating time delay is investigated. With nonwhite input, it is shown that the phase of the cross spectrum and the location of the peak in the cross correlator are biased. 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 (104)  |  Patents (19)
    Request permission for commercial reuse | 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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Aims & Scope

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

Full Aims & Scope