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IEEE Transactions on Acoustics, Speech, and Signal Processing

Issue 1 • February 1986

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Displaying Results 1 - 25 of 32
  • [Front cover and table of contents]

    Publication Year: 1986, Page(s): 0
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    Freely Available from IEEE
  • Authors' reply to "Comments on 'comparison of the convergence of two algorithms for adaptive FIR digital filters'"

    Publication Year: 1986, Page(s):202 - 203
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (264 KB)

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  • On an approximate subspace method for eigenfilter computation

    Publication Year: 1986, Page(s):205 - 207
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (377 KB)

    A method has been proposed for estimating the frequencies of sinusoids in additive white Gaussian noise. The method is an approximation to Kumaresan's "improved Pisarenko" method, which can be thought of as the computation of an FIR notch filter with zeros at the signal frequencies. The new algorithm consists of subtracting a small ε from the View full abstract»

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  • Fast algorithms for digital signal processing

    Publication Year: 1986, Page(s): 221
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (171 KB)

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  • [Back cover]

    Publication Year: 1986, Page(s): c4
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    Freely Available from IEEE
  • The block-processing FTF adaptive algorithm

    Publication Year: 1986, Page(s):77 - 90
    Cited by:  Papers (38)  |  Patents (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1352 KB)

    A new block-type adaptive-filtering algorithm is presented. This new block adaptive filter differs from the frequency-domain block adaptive filters of Ferrara (1980), and of Clark, Mitra, and Parker (1980), in that the new method applies a deterministic time-domain least-squares criteria within each of the data blocks. Information is carried from block to block via a weighted initial condition. Th... View full abstract»

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  • On the optimum data nonlinearity in LMS adaptation

    Publication Year: 1986, Page(s):69 - 76
    Cited by:  Papers (43)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (712 KB)

    The effect of an arbitrary nonlinear operation on the data input to the weight update equation in the LMS adaptive algorithm is investigated for a Gaussian data model. Exact difference equations are derived for the weight first and second moments that include the effects of an arbitrary nonlinear operation on the data sequence. The difference equations are used to obtain expressions for the transi... View full abstract»

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  • Computation of the exact information matrix of Gaussian time series with stationary random components

    Publication Year: 1986, Page(s):118 - 130
    Cited by:  Papers (75)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (976 KB)

    The paper presents an algorithm for efficient recursive computation of the Fisher information matrix of Gaussian time series whose random components are stationary, and whose means and covariances are functions of a parameter vector. The algorithm is first developed in a general framework and then specialized to the case of autoregressive moving-average processes, with possible additive white nois... View full abstract»

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  • Higher radix aperiodic-convolution algorithms

    Publication Year: 1986, Page(s):60 - 68
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (944 KB)

    Two alternative methods are proposed which can be used to derive efficient aperiodic-convolution algorithms for all filter lengths up to 36. A nesting technique is then described by which algorithms of this class can be used to derive efficient aperiodic-convolution algorithms for over 60 percent of the filter lengths in the range 37-1296. The computational complexity of the new algorithms is stud... View full abstract»

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  • On the automatic training of phonetic units for word recognition

    Publication Year: 1986, Page(s):209 - 213
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (688 KB)

    In this correspondence, we present some preliminary results on using phonetic subword units in word recognition as compared to whole word templates. The phonetic subword units are specified as either phonelike units with and without temporal structure or as diphonelike units. The determination of these subword units requires segmentation, labeling, and parameter estimation at the same time, and is... View full abstract»

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  • On the number of multiplications necessary to compute a length-2nDFT

    Publication Year: 1986, Page(s):91 - 95
    Cited by:  Papers (40)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (616 KB)

    The number of multiplications necessary and sufficient to compute a length-2nDFT is determined. The method of derivation is shown to apply to the multiplicative complexity results of Winograd for a length-pnDFT, for p an odd prime number. The multiplicative complexity of the one-dimensional DFT is summarized for many possible lengths. View full abstract»

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  • Block realization of multirate adaptive digital filters

    Publication Year: 1986, Page(s):105 - 117
    Cited by:  Papers (24)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1160 KB)

    In this paper, we study multirate transversal adaptive digital filters (ADF's) based on the block least-mean-square (BLMS) algorithms. These include an ADF with decimation (ADFD) and an ADF with interpolation (ADFI). We first formulate these filters based on the block mean-squared error (BMSE) criterion and derive BLMS multi-rate weight-adjustment algorithms, and then study efficient realization o... View full abstract»

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  • Estimating the covariance matrix by signal subspace averaging

    Publication Year: 1986, Page(s):8 - 12
    Cited by:  Papers (95)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (536 KB)

    An efficient algorithm is presented for estimating a covariance matrix consisting of a low-rank signal term and a full-rank noise term, known apart from a scalar factor. For each sample of the vector of sensor outputs, the algorithm approximates, in the least-squares sense, a rank-one update of the covariance matrix, under the side condition that the rank of the signal term remains bounded. If the... View full abstract»

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  • A zero crossing-based spectrum analyzer

    Publication Year: 1986, Page(s):96 - 104
    Cited by:  Papers (45)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (792 KB)

    A band-limited signal can be represented by the real and complex zeros of the signal. By simple invertible techniques, it is possible to convert the signal into another one which has only real (physical) zeros. Based on a zero-based representation, a spectrum analyzer is proposed. The spectrum analyzer samples the zero crossings of the signal waveform and uses this information to determine the dis... View full abstract»

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  • Estimating the steady-state frequency of a sine-wave burst with extremely short record length

    Publication Year: 1986, Page(s):213 - 215
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (368 KB)

    We present a technique based on modified linear prediction for accurate estimation of the steady-state frequency of a sine-wave burst with extremely short record length. The estimation procedure presented here makes use of automatic windowing of the burst location within the observation interval. Such a windowing technique eliminates the transition effects which accompany the estimators derived us... View full abstract»

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  • Nonlinear quantization effects in the frequency domain complex scalar LMS adaptive algorithm

    Publication Year: 1986, Page(s):140 - 151
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1048 KB)

    Nonlinear quantization effects in the frequency domain complex scalar LMS adaptive algorithm are analyzed by using conditional expectations. The probability density function of the quantizer input, conditioned on the weight, is derived. The density is applied to finding the conditional characteristic function and the Mth conditional moment at the quantizer output. The first and second conditional ... View full abstract»

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  • Glottal inverse filtering by joint estimation of an AR system with a linear input model

    Publication Year: 1986, Page(s):28 - 42
    Cited by:  Papers (31)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1584 KB)

    Glottal inverse filtering is a technique by which the flow past the time varying glottal constriction is estimated by a filtering operation on the acoustic signal in human speech. The filtering operation removes the effects of the vocal tract resonances to reveal the underlying voice source signal. A linear model of a glottal pulse waveform is described along with a procedure for jointly determini... View full abstract»

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  • On the role of input signals in deterministic linear filtering

    Publication Year: 1986, Page(s):203 - 205
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (408 KB)

    This correspondence shows that typical frequency domain design formulations of deterministic linear filters without explicit formal relationship with input signals, can be interpreted as worst-case designs with respect to normed spaces of input signals. Filter design formulations optimal only with respect to subsets of input signals are discussed. The interesting similarities with the minimax form... View full abstract»

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  • A class of rank order smoothers

    Publication Year: 1986, Page(s):157 - 165
    Cited by:  Papers (12)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (960 KB)

    Smoothers or nonlinear filters that make estimates purely on the basis of rank and temporal order determinations among members of the input data are inherently robust and compatible with VLSI implementations. The classic example is the moving median smoother, but it is possible to define others which have significantly different attributes both in relation to the statistical estimation problem and... View full abstract»

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  • Optimum binary windows for discrete Fourier transforms

    Publication Year: 1986, Page(s):216 - 220
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (336 KB)

    A class of windows, called binary windows, for the frequency-domain implementation of the discrete Fourier transform is proposed. While the conventional time-domain windowing requires N/2 stored values of the data window and N multiplications, the binary windows do not require multiplications or stored constants; rather, they are replaced by shift and add operations. Typical hardware structures ar... View full abstract»

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  • Complex digital signal processing using quadratic residue number systems

    Publication Year: 1986, Page(s):166 - 177
    Cited by:  Papers (25)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1000 KB)

    Recently, the quadratic residue number system (QRNS) has been introduced [6], [7] which allows the multiplication of complex integers with two real multiplications. The restriction is that the number system has either all prime moduli of the form 4K + 1, or composite numbers with prime factors of that form. If an increase in real multiplications from two to three can be tolerated, then the restric... View full abstract»

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  • A new application of adaptive noise cancellation

    Publication Year: 1986, Page(s):21 - 27
    Cited by:  Papers (78)  |  Patents (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (896 KB)

    A new application of Widrow's adaptive noise cancellation (ANC) is presented in this paper. Specifically, the method is applied to the case where an acoustic barrier exists between the primary and reference microphones. By updating the coefficients of the noise estimation filter only during silence, it is shown that ANC can provide substantial noise reduction with little speech distortion even whe... View full abstract»

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  • Equalization with few adaptive frequency gains for rapidly evolutive selective fading

    Publication Year: 1986, Page(s):208 - 209
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    In this correspondence, we present a new equalizer with a reduced number of adaptive taps for rapid selective fading in data transmission channels. The equalization is performed in the frequency domain with a bank of frequency sampling filters. Only the gains of the filters whose passbands lie in the fading area are updated. The fading area is adaptively identified by an auxiliary coarse equalizer... View full abstract»

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  • A low-complexity echo canceller performing the channel phase acquisition

    Publication Year: 1986, Page(s):200 - 202
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (360 KB)

    An algorithm with low complexity (multiplication free) is presented for the purpose of echo cancellation at a data rate of up to 4800 bit/s. It is derived from the sign algorithm, and takes into account only the component of the error that is not corrupted by the far-end data signal. View full abstract»

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  • Time delay estimation by combining efficient algorithms and generalized cross-correlation methods

    Publication Year: 1986, Page(s):1 - 7
    Cited by:  Papers (26)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (584 KB)

    This paper analyzes the performance of the two efficient algorithms (presented by Stein and Cabot) for time delay of arrival estimation (TDE) between two signals. It is shown that these estimators are unbiased, and explicit expressions for the TDE mean-square error (MSE) are presented. It is also shown how to improve the performance of these algorithms by combining them with generalized cross-corr... 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