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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 3 • Date June 1980

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Displaying Results 1 - 21 of 21
  • [Front cover and table of contents]

    Page(s): 0
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    Freely Available from IEEE
  • Comments on "Least squares glottal inverse filtering from the acoustic speech waveform"

    Page(s): 343
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    The criticism of Wong, Markel, and Gray concerning a paper by the author is shown to be based on a misunderstanding. The author's former statements are explained and some of his results are compared with those of Wong et al. View full abstract»

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  • [Back cover]

    Page(s): c4
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    Freely Available from IEEE
  • Performance evaluation of array processors for detecting Gaussian acoustic signals

    Page(s): 328 - 333
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    The detection performance for "noise-like" underwater acoustic signals from a known direction is presented using the receiver operating characteristic (ROC). Analytical expressions for PFAand PDare derived in terms of the eigenvalues of the covariance matrices of the received data under both hypotheses. Applications of this method to linear arrays with equal or unequal spacing and planar arrays are shown. As an example, the optimal array performance for detecting Gaussian signals in both directional and independent Gaussian noise is evaluated and compared with that of a conventional beamformer envelope detector and a null steering processor. View full abstract»

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  • Response of the adaptive line enhancer to chirped and doppler-shifted sinusoids

    Page(s): 343 - 348
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    The adaptive line enhancer (ALE) was first described by Widrow et al. as a practical on-line technique for separating the coherent components from the incoherent components of an input signal. Subsequent work has shown this same adaptive filtering structure to be applicable to maximum entropy spectral estimation, predictive deconvolution, and narrow-band interference rejection. While an often cited advantage of adaptive filtering is its tolerance of slowly time-varying input statistics, the existing analyses of the ALE have concentrated on the stationary case. This correspondence extends these results, applying the theory to the case of inputs containing sinuosids whose frequencies slowly vary in time. This is approached by developing a time-varying eigenvalue-eigenvector description of the expected filter impulse response vector. These results are then used to predict the expected impulse response vector for the ALE input of stationary white noise plus a sinusoid with linearly swept frequency. The response of the ALE for this particular input signal provides useful benchmarks for dealing with more complex forms of frequency modulation. The utility of the theoretical results is demonstrated by considering the ALE's response to an input of practical interest to the sonar community, a sinusoid whose apparent frequency is shifted by Doppler effects. View full abstract»

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  • The power spectral density of a random sequence of zeros and ones

    Page(s): 335 - 337
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    The statistical properties of a randomly sampled sequence are conveniently studied by expressing the randomly sampled sequence as the term wise product of uniformly spaced signal values with a random sequence of zeros and ones. In this correspondence we derive a simple expression for the power spectral density of a class of such sequences of zeros and ones. View full abstract»

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  • Sequential regression considerations of adaptive notch filers

    Page(s): 313 - 317
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    This paper presents a sequential regression (SER) formulation for adaptive notch filters. The characteristics of this class of notch filters are compared with those obtained via the least-mean-square (LMS) algorithm. Some experimental results are also included. View full abstract»

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  • A generalized framework for power spectral estimation

    Page(s): 334 - 335
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    This correspondence presents a generalized framework for power spectral estimation and shows how three previous estimation methods fit into this framework as special cases. Further, this correspondence clarifies some recent misleading comments made on spectral estimation; in particular, specific references are given that strongly support the use of overlapped weighted segment averaging for spectral estimation. View full abstract»

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  • A stepwise approach to computing the multidimensional fast Fourier transform of large arrays

    Page(s): 280 - 284
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    We consider the problem of performing a two-dimensional fast Fourier transform (FFT) on a very large matrix in limited core memory. We propose a decomposition of the Cooley-Tukey algorithm to allow efficient utilization of core memory and mass storage. The number of input/output operations is greatly reduced, with no increase in the computational burden. The method is suitable for nonsquare matrices and arrays of three or more dimensions. View full abstract»

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  • Fourier transform phase coding of images

    Page(s): 339 - 341
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    The importance of phase has been demonstrated in previous research [1], [2] which indicates that an image constructed from only the phase of the discrete Fourier transform (DFT) of the original image is typically very intelligible. This correspondence presents a coding algorithm which codes phase and magnitude separately, and emphasizes the phase coding. Good results have been obtained with 1 bit/pixel. View full abstract»

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  • Phaselock techniques

    Page(s): 349 - 350
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    First Page of the Article
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  • An interpretation of the log likelihood ratio as a measure of waveform coder performance

    Page(s): 318 - 323
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    The log likelihood measure has been widely used in speech research for comparing speech signals. Recently, it has been proposed as a measure for assessing the quality of coded speech. In this paper we present an interpretation of the log likelihood ratio measure within the theoretical framework of a waveform coder distortion model. We then discuss the implications of this interpretation and show how it can be applied to the formulation of better objective measures of waveform coder performance. View full abstract»

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  • Reverberation characteristics of 4 kHz and 20 kHz tone bursts in a water filled tank and their effect on experimental design

    Page(s): 324 - 327
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    Although reverberation characteristics of air filled rooms have been exhaustively studied and defined, those of water filled tanks have not; water reverberation work has been concentrated on the open ocean. We have obtained reverberation data on a typical tank and applied it to the design of experiments for both human and animal subjects. Computer-programmed pulses were used with standard instrumentation to provide the data. Photographs of pulse returns and decay and their time of audibility by the dolphin are presented. View full abstract»

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  • Evaluation of quantization error in two-dimensional digital filters

    Page(s): 273 - 279
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    In the evaluation of the quantization error in two-dimensional (2-D) digital filters, a procedure for computingsummin{m=0}max{infin} summin{n=0}max{infin} y^{2}(m,n)= frac{1}{(2pij)^{2}}ointoint Y(z_{1},z_{2})Y(z_{1}^{-1},z_{2}^{-1}) frac{dz_{1}dz_{2}}{z_{1}z_{2}}T^{2} = {(z_{1},z_{2}): |z_{1}|=1, |z_{2}|=1}is required. In this paper a condition for a finite quantization error is given and a discussion on the evaluation of the integral based on the residue method is presented. Examples for such an evaluation are given. Furthermore, the salient differences between the one-dimensional (1-D) complex integral evaluation and the two-dimensional one are discussed. Notation: We note withbar{U}^{2} = {(z_{1}, z_{2}): |z_{1}| leq 1, |z_{2}| leq 1} the closed unit bidisk, withu^{2} = {(z_{1}, z_{2}): |z_{1}| < 1, |z_{2}| leq 1}the open unit bidisk, and withT_{2} = {(z_{1}, z_{2}): |z_{1}| = 1, |z_{2}| = 1}the distinguished boundary of the unit bidisk. The 2-Dz-transform is defined asY(z_{1}, z_{2} = summin{m=0}max{infin}summin{n=0}max{infin} y (m,n)z_{1}^{m}z_{2}^{n}. View full abstract»

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  • Construction of stable multidimensional recursive filters via pseudopolynomials

    Page(s): 337 - 339
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    In this correspondence it is shown that the concept of pseudopolynomials, which is well known in the theory of analytic functions of several complex variables, can be useful in finding a class of stable, nonseparable, multivariable transfer functions without using the stability tests. View full abstract»

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  • Residue number system implementations of number theoretic transforms in complex residue rings

    Page(s): 285 - 291
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    This paper discusses the implementation of number theoretic transforms defined in complex residue rings. A selection of transform parameters for lookup table and microprocessor realizations are discussed. When the length of the convolution is moderate, the lookup table approach offers greater flexibility in the choice of dynamic range. The microprocessor implementation is shown to handle a number theoretic transform of 840 complex samples. View full abstract»

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  • Integration of equally spaced sampled data using a generalized Whittaker's reconstruction formula

    Page(s): 341 - 343
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    A family of integration formulas for equally spaced sampled data are derived which can have errors of -70 dB or less over a bandwidth of 95 percent of more of the Nyquist bandwidth. The formulas are based on the integration of a reconstructed waveform using a generalized Whittaker's cardinal function. View full abstract»

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  • Disc recording and reproduction

    Page(s): 350
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    First Page of the Article
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  • An efficient lpoptimization technique for the design of two-dimensional linear-phase FIR digital filters

    Page(s): 308 - 313
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    An iterative optimization technique, based on the method of parallel tangents coupled with an efficient line searching technique, is proposed for the design of two-dimensional linear-phase FIR digital filters. The performance index for the optimization procedure is shown to be convex and hence this technique will always converge towards the global minimum. An important feature is that the design time increases slowly as the number of filter coefficients increases. To illustrate the technique several circular low-pass filters were designed. These filters compare favorably with filters designed using the minimax technique that was developed by Harris. The technique can also be used for the design of one-dimensional and multidimensional linear-phase FIR digital filters. View full abstract»

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  • Design of optimal finite wordlength FIR digital filters using integer programming techniques

    Page(s): 304 - 308
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    The application of a general-purpose integer-programming computer program to the design of optimal finite wordlength FIR digital filters is described. Examples of two optimal low-pass FIR finite wordlength filters are given and the results are compared with the results obtained by rounding the infinite wordlength coefficients. An analysis of the approach based on the results of more than 50 design cases is presented and the problem of optimal wordlength choice is discussed. View full abstract»

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  • Noise compensation for autoregressive spectral estimates

    Page(s): 292 - 303
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    The autoregressive spectral estimator possesses excellent resolution properties for time series which satisfy the "all-pole" assumption. When noise is added to the time series under analysis, the resolution of the spectral estimator decreases rapidly as the signal-to-noise ratio decreases. The usual approach to this problem is to model the resulting time series by the more appropriate autoregressive-moving average process and to use standard time series analysis techniques to identify the autoregressive parameters. This standard technique, however, does not result in a positive-definite autocorrelation matrix. As a result, it is shown that the resulting spectral estimator may exhibit a large increase in variance. An alternative approach, termed the noise compensation technique, is proposed. It attempts to correct the estimated reflection coefficients for the effect of white noise, assuming the noise variance is known or can be estimated. Simulation results indicate that a significant decrease in the degrading effects of noise may be realized using the noise compensation technique. 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