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IEEE Transactions on Acoustics, Speech, and Signal Processing

Issue 2 • April 1982

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  • [Front cover and table of contents]

    Publication Year: 1982, Page(s): 0
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    Freely Available from IEEE
  • High-speed coherence processing using the sectionalized Fourier transform

    Publication Year: 1982, Page(s):189 - 205
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1630 KB)

    The sectionalized Fourier transform of a band-limited signal (defined as a Fourier transform which is computed over incremented temporal sections of the function) is equivalent to basebanding, filtering, and sampling the signal in time domain. Spectral windowing is employed, through appropriately summing a sequence of the Fourier transform bins, to control the passband and leakage characteristics ... View full abstract»

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  • A fast digital method of estimating the autocorrelation of a Guassian stationary process

    Publication Year: 1982, Page(s): 329
    Cited by:  Papers (19)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (95 KB)

    Given a zero mean stationary Gaussian process{x_{n}}, it is shown that the autocorrelation can be estimated byhat{R}_{xx}(j) = C_{N} summin{i=1}max{N} x_{i} sign(x_{i+j})whereC_{N}=frac{pi}{2N^{2}} summin{i=1}max{N}|x_{i}|. This method is attractive sinceC_{N}needs to he Computed only once, andsummin{i=1}max{N} x_{i} sign (x_{i+j})can be compu... View full abstract»

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  • How to make an all-pass filter with a desired impulse response

    Publication Year: 1982, Page(s):336 - 337
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (233 KB)

    lt is shown that, and how for any sequence ofn + 1numbers, an all-pass filter of orderleq ncan be constructed whose impulse response begins with the given sequence up to a gain factor. An application is pulse expansion, for instance, speech prefiltering in ADPCM systems in order to avoid quantizer overload due to pitch pulses. View full abstract»

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  • Correction to "Sequential convolution techniques for image filtering"

    Publication Year: 1982, Page(s): 346
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (368 KB)

    First Page of the Article
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  • [Back cover]

    Publication Year: 1982, Page(s): c4
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    Freely Available from IEEE
  • Reverberation between concentric spheres

    Publication Year: 1982, Page(s):162 - 166
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (496 KB)

    Reverberation time T is widely used in architectural acoustics as a figure of merit for an auditorium. Only approximate formulas have been available for calculating T from auditorium dimensions. Recently H. Kuttruff and W. B. Joyce showed that T could be obtained exactly from a difficult integral equation. A solution is given here for a sphere containing a smaller concentric sphere. The radii and ... View full abstract»

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  • The reconstruction of a multidimensional sequence from the phase or magnitude of its Fourier transform

    Publication Year: 1982, Page(s):140 - 154
    Cited by:  Papers (231)  |  Patents (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (2464 KB)

    This paper addresses two fundamental issues involved in the reconstruction of a multidimensional sequence from either the phase or magnitude of its Fourier transform. The first issue relates to the uniqueness of a multidimensional sequence in terms of its phase or magnitude. Although phase or magnitude information alone is not sufficient, in general, to uniquely specify a sequence, a large class o... View full abstract»

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  • Hardware for two-dimensional digital filtering using Fermat number transforms

    Publication Year: 1982, Page(s):155 - 162
    Cited by:  Papers (6)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (944 KB)

    Two-dimensional convolution computed using minicomputers is very time-consuming when Fourier transform techniques are used because of the large number of complex multiplications required. The Fermat number transform (FNT) has been proposed as an efficient and exact method of implementing one-dimensional and two-dimensional convolutions. Nonetheless, certain inherent restrictions of this transform ... View full abstract»

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  • Bias and variance of spectral estimates from an all-pole digital filter

    Publication Year: 1982, Page(s):322 - 329
    Cited by:  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (536 KB)

    An adaptive digital filter, equivalent to a one step linear predictor, may be employed to determine spectral estimates closely related to the linear prediction all-pole model spectrum of the input signal. This paper presents an analysis of the bias and variance of this estimate for a filter using the Widrow-Hoff LMS algorithm. Results are compared with computer simulations for two four pole signal... View full abstract»

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  • Robust detection by autoregressive spectrum analysis

    Publication Year: 1982, Page(s):256 - 269
    Cited by:  Papers (33)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1096 KB)

    The problem of detecting a signal with an unknown Doppler shift and random phase in white noise is essentially a problem in spectral analysis. This paper examines the merits of a detector based upon the autoregressive spectral estimator. Some advantages of the auto-regressive detector are that the detection performance is independent of Doppler shift and phase and the false alarm rate is independe... View full abstract»

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  • Order selection for and design of IIR filters

    Publication Year: 1982, Page(s):211 - 216
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (624 KB)

    The design of infinite impulse response digital filters can be viewed as a spectral factorization problem. When one knows what the numerator and denominator orders are, this design problem is greatly simplified. Selection of the appropriate orders has long been a significant problem in recursive filter design. Recent work by Gray et al. suggests a general method by which the orders may be selected... View full abstract»

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  • On a new adaptive lattice algorithm for recursive filters

    Publication Year: 1982, Page(s):316 - 319
    Cited by:  Papers (18)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (288 KB)

    A new and simplified adaptive algorithm is presented for lattice structured recursive filters implementing the two-multiplier form based on Itakura and Saito. A comparison is also presented between the results obtained from this new algorithm and the algorithm introduced recently by Parikh, Ahmed, and Stearns. View full abstract»

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  • A recursive Kalman window approach to image restoration

    Publication Year: 1982, Page(s):125 - 140
    Cited by:  Papers (14)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1928 KB)

    Recursive restoration of blurred and noisy images using Kalman algorithms is hindered by the correlated nature of the two-dimensional data and the excessive computing requirements of the very long state vectors. This paper uses a semicausal model for image representation to account for the correlated nature of the data. Such a model is subsequently used to develop a discrete linear imaging system ... View full abstract»

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  • LPC distance measures and statistical tests with particular reference to the likelihood ratio

    Publication Year: 1982, Page(s):304 - 315
    Cited by:  Papers (25)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1160 KB)

    Several LPC distance measures and statistical tests have been proposed for use in speech processing, the most popular of which is Itakura's log likelihood ratio statistic, and some simple variants thereof. In this paper it is shown that these statistics share some undesirable properties. It is argued that there are more tractable and more sensitive measures available including other relevant likel... View full abstract»

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  • Two maximum entropy beamforming algorithms for equally spaced line arrays

    Publication Year: 1982, Page(s):175 - 189
    Cited by:  Papers (12)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1232 KB)

    This paper presents two previously unpublished maximum entropy beamforming algorithms applicable to equally spaced line arrays. Both algorithms, the first due to John P. Burg and the second due to the author, modify the Burg technique to permit simultaneous time averaging and spatial averaging for quasi-time-stationary array data. View full abstract»

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  • The λ²-LMS algorithm for separating sinusoids using the intermediate-converged adaptive filter solution

    Publication Year: 1982, Page(s):246 - 256
    Cited by:  Papers (4)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1056 KB)

    A new procedure is proposed for automatically separating sinusoids of different, but unknown, frequencies, based only on the fact that they differ in amplitude. The procedure involves taking the intermediate solution from the LMS adaptive solution rather than the converged steady-state solution. In addition, a new algorithm which converges at a rate proportional to the squares of the signal eigenv... View full abstract»

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  • System identification techniques for adaptive signal processing

    Publication Year: 1982, Page(s):240 - 246
    Cited by:  Papers (35)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (784 KB)

    Many problems in adaptive filtering can be approached from the point of view of system identification. The recursive maximum likelihood algorithm is proposed for estimating the parameters of the signal model. The parameter estimates are then used to form an adaptive infinite impulse response filter. Several examples are discussed including: adaptive line enhancement, adaptive deconvolution, adapti... View full abstract»

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  • Signal averaging by parallel digital filters

    Publication Year: 1982, Page(s):338 - 346
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1360 KB)

    Signal averaging for signal-to-noise ratio improvement is modified by using parallel digital filters instead of the classical block averager. The digital filter averager allows the user to specify the signal-to-noise ratio improvement, and it computes a continuously updated average. The steady-state and transient performance of the averagers are demonstrated by equations and by simulation. View full abstract»

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  • A design algorithm for constrained equiripple digital filters

    Publication Year: 1982, Page(s):206 - 211
    Cited by:  Papers (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (656 KB)

    The large scale integration (LSI) of systems imposes several new requirements on a digital filter. Typically, these requirements can be met by designing equiripple filters with arbitrary passband or stop-band shaping, with prescribed zeros or poles, and with different, both in number and magnitude, ripples in passband and stopband. The design of such filters requires constrained equiripple approxi... View full abstract»

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  • Permuted difference coefficient realization of FIR digital filters

    Publication Year: 1982, Page(s):269 - 278
    Cited by:  Papers (15)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (856 KB)

    A new realization for FIR digital filters, using permuted difference coefficients, is proposed in this paper. Its coefficients are obtained as the difference between the successive values of the original coefficients reordered in a sequence with falling magnitude. The proposed realization can hold desirable properties in FIR filters, such as an exactly linear phase characteristic and stable implem... View full abstract»

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  • Distortion performance of vector quantization for LPC voice coding

    Publication Year: 1982, Page(s):294 - 304
    Cited by:  Papers (53)  |  Patents (9)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1168 KB)

    The distortion performance of the vector quantization approach for LPC voice coding is examined both analytically and experimentally. Analytically, interpretations of the interparameter coupling effects of a distortion measure and the clustering nature of the algorithm for LPC vector quantization are obtained to show its relationship with the residual minimization process in LPC analysis. Experime... View full abstract»

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  • A note on Maqusi's proofs and truncation error bounds in the dyadic(Walsh)sampling theorem

    Publication Year: 1982, Page(s):334 - 335
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (216 KB)

    In [1] and [2] bounds for the truncation error are presented which arises when a sequency-limited signal function is reconstructed by finite Walsh sampling sums. The knowledge of the properties that sequency-limited functions have leads to an immediate proof of the dyadic sampling theorem and truncation turns out to be a trivial affair. View full abstract»

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  • On recursive discrete Fourier transformation

    Publication Year: 1982, Page(s):319 - 322
    Cited by:  Papers (24)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (464 KB)

    It is shown that the recent deadbeat state-observer approach to discrete Fourier transform evaluation is exactly equivalent to the nonrecursive frequency-sampling filters of finite impulse response digital filter design. The approach is then extended to consider the Kalman filter derived from the signal model and this is demonstrated to be nonrobust and pathological. The application of state obser... View full abstract»

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  • Noise figures of differential-pair input-stage equalizer preamplifiers

    Publication Year: 1982, Page(s):167 - 174
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (608 KB)

    Noise characteristics of audio amplifiers with field-effect (FET) and bipolar-junction transistor (BJT) differential-pair (DP) input-stage and inductive source impedance are reviewed. The signal-to-noise ratio (SNR) for RIAA phono equalizers is evaluated, considering the influence of the source impedance and different ear-response weighting. 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