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IEEE Transactions on Acoustics, Speech, and Signal Processing

Issue 2 • April 1981

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Displaying Results 1 - 25 of 30
  • [Front cover and table of contents]

    Publication Year: 1981, Page(s): 0
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    Freely Available from IEEE
  • Design of FIR digital phase networks

    Publication Year: 1981, Page(s):171 - 176
    Cited by:  Papers (65)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (519 KB)

    The problem of the minimax design of FIR digital filters with prescribed phase characteristics and unit magnitude is a nonlinear optimization problem. In this paper it is approximated by a linear programming problem, and it is shown that the solution of this linear program is optimal to first order. That is, if δ0and ε0are optimal deviations of magnitude and phase c... View full abstract»

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  • A novel window for harmonic analysis

    Publication Year: 1981, Page(s):177 - 188
    Cited by:  Papers (25)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1052 KB)

    This paper presents a new window design technique and discusses its effect on the detection of harmonic signals in the presence of nearby strong harmonic interference. Four design parameters are introduced to independently control the pattern falloff rate, the overall sidelobe level, the near-sidelobe level and the depth of a steerable wide dip. Since the deep dip can be steered to any frequency t... View full abstract»

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  • m-adic (polyadic) invariance and its applications

    Publication Year: 1981, Page(s):207 - 213
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (588 KB)

    The concept of m-adic invariance allows approximation of a linear time-invariant (LTI) system by a linear m-adic invariant (LMI) system or, equivalently, approximation of a circulant matrix by a super-circulant matrix. The approximation reduces the number of multiplies required for computing N-point cyclic convolution to2(log_{m}N - 1)N, where N = mn. The error introduced by ... View full abstract»

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  • A parameter estimation approach to estimation of frequencies of sinusoids

    Publication Year: 1981, Page(s):214 - 219
    Cited by:  Papers (86)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (584 KB)

    It is shown that with uniform sampling, a sample of the sum of M sinusoids at time nT can be uniquely expressed as a linear combination of the 2M samples at times (n - 1)T,..., (n - 2M)T. The 2M coefficients of linear combination are also the coefficients of a 2M-order polynomial whose roots aree^{pmjomega_{i}}, omega_{i}being the frequencies of the sinusoids. Given the samples of sinus... View full abstract»

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  • On finite wordlength FIR filter design

    Publication Year: 1981, Page(s): 329
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (129 KB)

    A recent paper has investigated the use of integer linear programming for the design of Chebyshev-optimal finite wordlength FIR digital filters. We point out that for some time domain error norms, coefficient rounding provides the best approximation to the infinite precision filter. View full abstract»

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  • [Back cover]

    Publication Year: 1981, Page(s): c4
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    Freely Available from IEEE
  • Kalman filtering in two dimensions: Further results

    Publication Year: 1981, Page(s):188 - 197
    Cited by:  Papers (136)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1288 KB)

    The two-dimensional reduced update Kalman filter was recently introduced. The corresponding scalar filtering equations were derived for the case of estimating a Gaussian signal in white Gaussian noise and were shown to constitute a general nonsymmetric half-plane recursive filter. This paper extends the method to the deconvolution problem of image restoration. This paper also provides a more thoro... View full abstract»

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  • Tests of coherence unbiasing methods

    Publication Year: 1981, Page(s):321 - 323
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (344 KB)

    The causes of bias in the mean-squared coherence (MSC) estimate are discussed, assuming that the estimate is made using the FFT method. Two known unbiasing methods, prewhitening and MSC averaging, are tested in situations where the true MSC is known, and are seen to reduce bias. MSC averaging is recommended as the best unbiasing method. View full abstract»

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  • Minimum cross-entropy spectral analysis

    Publication Year: 1981, Page(s):230 - 237
    Cited by:  Papers (107)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (888 KB)

    The principle of minimum cross-entropy (minimum directed divergence, minimum discrimination information, minimum relative entropy) is summarized, discussed, and applied to the classical problem of estimating power spectra given values of the autocorrelation function. This new method differs from previous methods in its explicit inclusion of a prior estimate of the power spectrum, and it reduces to... View full abstract»

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  • Zero sets of multiparameter functions and stability of multidimensional systems

    Publication Year: 1981, Page(s):197 - 206
    Cited by:  Papers (70)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (952 KB)

    New theorems, which can be used to find the zero set of a multiparameter rational function of a complex variable where the distinguished boundary of the parameters' closed domain of definition is composed of piecewise smooth Jordan arcs or curves, are stated and proved. The results are used to derive a new simplified procedure for a multi-dimensional stability test, presented for the continuous-an... View full abstract»

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  • Quantitative determination of periodicities from short time series

    Publication Year: 1981, Page(s):329 - 331
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (304 KB)

    A new two-step least squares procedure is presented for estimating frequencies, amplitudes, and phases of sinusoids in short time series. Frequencies, revealed by autoregressive modeling algorithms in the first step, are used to determine amplitudes and phases in the second step. View full abstract»

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  • A multiresolution sampled-data spectrum analyzer for a detection system

    Publication Year: 1981, Page(s):163 - 170
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (776 KB)

    A spectrum analyzer for use with a real-time sampled-data detection system is described. At regular time intervals, it provides the detection system with an updated spectrum of the whole frequency band of the sampled data plus one or more narrower bandwidth, higher resolution (NBHR) spectra at selected frequencies. The number of NBHR spectra, their positions, bandwidths, and resolutions are input ... View full abstract»

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  • Direct Fourier reconstruction in computer tomography

    Publication Year: 1981, Page(s):237 - 245
    Cited by:  Papers (100)  |  Patents (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1296 KB)

    Direct Fourier transform (FT) reconstruction of images in computerized tomography (CT) is not widely used because of the difficulty of precisely interpolating from polar to Cartesian samples. In this paper, an exact interpolation scheme is proposed which, in practice, can be approached with arbitrary accuracy using well-conditioned algorithms. Several features of the direct FT method are discussed... View full abstract»

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  • Direct (nonrecursive) relations between cepstrum and predictor coefficients

    Publication Year: 1981, Page(s):297 - 301
    Cited by:  Papers (28)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (552 KB)

    Direct, i.e., nonrecursive, relations are derived for the cepstrum in terms of the predictor coefficients and vice versa. Connections with algebraic roots, symmetric functions, statistical moments, and cumulants are pointed out. Some implications for pitch detection are also discussed. View full abstract»

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  • CCD based adaptive filters: Realization and analysis

    Publication Year: 1981, Page(s):220 - 229
    Cited by:  Papers (14)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (2120 KB)

    In this paper the design and construction of a number of alternative adaptive filter implementations using an integrated charge-coupled device programmable transversal filter are described. Particular emphasis is placed on the analysis of errors in the system structure from both the theoretical and experimental standpoint. The principal objective was to demonstrate that an adaptive Filter having a... View full abstract»

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  • Cepstral analysis technique for automatic speaker verification

    Publication Year: 1981, Page(s):254 - 272
    Cited by:  Papers (526)  |  Patents (37)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1920 KB)

    This paper describes new techniques for automatic speaker verification using telephone speech. The operation of the system is based on a set of functions of time obtained from acoustic analysis of a fixed, sentence-long utterance. Cepstrum coefficients are extracted by means of LPC analysis successively throughout an utterance to form time functions, and frequency response distortions introduced b... View full abstract»

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  • Performance of an ADPCM/TASI system for PCM speech compression

    Publication Year: 1981, Page(s):302 - 310
    Cited by:  Papers (8)  |  Patents (11)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (984 KB)

    This paper contains a summary of performance measurements for a digital speech interpolation (DSI) system called adaptive differential PCM with time assignment speech interpolation (ADPCM/ TASI). This system is used to compress the output of two or more T1 PCM terminals into a bit rate normally requited by one such terminal. The performance measurements presented are as follows. 1) Signal-to-noise... View full abstract»

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  • An economical class of digital filters for decimation and interpolation

    Publication Year: 1981, Page(s):155 - 162
    Cited by:  Papers (598)  |  Patents (140)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (816 KB)

    A class of digital linear phase finite impulse response (FIR) filters for decimation (sampling rate decrease) and interpolation (sampling rate increase) are presented. They require no multipliers and use limited storage making them an economical alternative to conventional implementations for certain applications. A digital filter in this class consists of cascaded ideal integrator stages operatin... View full abstract»

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  • Transformed coherence functions for multivariate studies

    Publication Year: 1981, Page(s):317 - 319
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (400 KB)

    In this paper, transformed coherence function estimates are defined which display several desirable properties when compared with the conventional forms; 1) their probability distribution functions are more nearly normal, 2) their mean values are normalized to a value of unity for totally uncorrelated data, and 3) their variances are independent of the true values. View full abstract»

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  • A level building dynamic time warping algorithm for connected word recognition

    Publication Year: 1981, Page(s):284 - 297
    Cited by:  Papers (107)  |  Patents (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1432 KB)

    Dynamic time warping has been shown to be an effective method of handling variations in the time scale of polysyllabic words spoken in isolation. This class of techniques has recently been applied to connected word recognition with high degrees of success. In this paper a level building technique is proposed for optimally time aligning a sequence of connected words with a sequence of isolated word... View full abstract»

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  • Intuitive explanation for low roundoff noise in normal realizations of narrow-band digital filters

    Publication Year: 1981, Page(s):311 - 312
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (232 KB)

    The low roundoff noise performance of narrow-band low-pass digital filters realized in normal form is explained in terms of the zeros of the internal transfer functions that control the dynamic range-roundoff noise interaction. For narrow-band filters that have pole pairs close to the real axis, the internal transfer functions of normal realizations have a zero between each of these pole pairs. Th... View full abstract»

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  • New methods for the design of recursive digital filters in the time domain

    Publication Year: 1981, Page(s):245 - 254
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (856 KB)

    This paper presents new methods for the design of optimal and suboptimal recursive digital filters in the time domain. The proposed methods are based on the approximation of the desired impulse response of a digital filter. Since the duration of impulse response of a recursive digital filter is not finite, it is desirable to use an infinite interval performance index from the point of stability. F... View full abstract»

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  • Performance of transform and subband coding systems combined with harmonic scaling of speech

    Publication Year: 1981, Page(s):273 - 283
    Cited by:  Papers (19)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1232 KB)

    In this study an approach for improving the performance of waveform coders, based on coding a frequency scaled speech signal, is examined and subjectively evaluated for specific subband and transform coding systems. The recently developed simple and efficient time-domain harmonic scaling (TDHS) algorithms are used to frequency scale the speech signal. The underlying frequency-domain model of the p... View full abstract»

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  • Spectral analysis: On the usefulness of linear tapering for leakage suppression

    Publication Year: 1981, Page(s):323 - 329
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (712 KB)

    The problem of linear versus quadratic modification in power spectral analysis is discussed. It is shown that linear tapering is of great benefit in the investigation of spectra with a large dynamic range. In particular we demonstrate that a linear taper is required to prevent an undesirable leakage effect in the estimate's random error. Examples are given of a spectral analysis with and without a... 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