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IEEE Transactions on Acoustics, Speech, and Signal Processing

Issue 6 • Date December 1982

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

    Publication Year: 1982, Page(s): 0
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    Freely Available from IEEE
  • [Back cover]

    Publication Year: 1982, Page(s): c4
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    Freely Available from IEEE
  • Fast algorithms for linear prediction and system identification filters with linear phase

    Publication Year: 1982, Page(s):942 - 953
    Cited by:  Papers (57)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1088 KB)

    A general finite impulse response (FIR) filter can be used as a linear prediction filter, if given only an input sample sequence, or as a system identification model, if given the input and output sequences from an unknown system. With known correlation, the coefficients of the FIR filter that minimize the mean square error in both applications are found by solution of a set of normal equations wi... View full abstract»

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  • Direct sample interpolation (DSI) speech synthesis: An interpolation technique for digital speech data compression and speech synthesis

    Publication Year: 1982, Page(s):825 - 832
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (904 KB)

    Direct transcription of the waveform is a potentially simple method to generate speech. However, it has a poor reputation due to the enormous data store required, although many digital encoding techniques have been suggested which reduce the data store significantly. Digital encoding is not invoked but a method called direct sample interpolation (DSI) is described which will compute bridging secti... View full abstract»

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  • Magnitude of the truncation error in sampling expansions of band-limited signals

    Publication Year: 1982, Page(s):906 - 912
    Cited by:  Papers (31)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (680 KB)

    Bounds are established for the truncation error in the sampling theorem for signals with radian bandwidth πW, W > 0. The assumptions are either differenfiability conditions upon the Fourier transform of the signal or on the rate of decay of the first derivative of the signal itself. Finally, our theorems are applied to two new examples, worked out in detail. View full abstract»

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  • Phase unwrapping by factorization

    Publication Year: 1982, Page(s):984 - 991
    Cited by:  Papers (54)  |  Patents (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (800 KB)

    An algorithm for the numerical factorization of very high degree but well-conditioned polynomials is developed. This is used to factor the z-transform of finite-length signals, and the zeros are used to calculate the unwrapped phase. The method has been tested on signals up to 512 points in length. A complete Fortran 77 program is given for the case of a real-valued signal. Two related analytical ... View full abstract»

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  • An interpretation of error spectrum shaping in digital filters

    Publication Year: 1982, Page(s):1013 - 1015
    Cited by:  Papers (10)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (392 KB)

    Error spectrum shaping is used in digital filters to reduce roundoff noise. There are several other filter structures which can also reduce roundoff noise. Are these methods of noise reduction in any way related? This paper shows that optimal error spectrum shaping is a method of increasing the word length of accumulators and internal variables, i.e., it is, in its optimum form, extended precision... View full abstract»

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  • Restoration of continuously sampled band-limited signals from aliased data

    Publication Year: 1982, Page(s):937 - 942
    Cited by:  Papers (6)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (640 KB)

    A continuously sampled signal is obtained by periodically placing a signal to zero. A straightforward closed form method is presented for restoration of continuously sampled bandlimited signals-even when the data is aliased. The sampled signal is simply multiplied by a periodic function specified by the duty cycle of the degradation and the severity of aliasing. This product is then placed through... View full abstract»

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  • An explicit estimate of time delay between two signals, with an unknown relative phase shift

    Publication Year: 1982, Page(s):1006 - 1007
    Cited by:  Papers (5)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (192 KB)

    This correspondence presents an explicit method for evaluating the time delay between two signals with a random relative phase shift. The analysis applies for Gaussian narrow-band signals. The method is derived from the ML estimator [1], but is much faster than the ML estimator since a search for a maximum is replaced by an explicit formula. View full abstract»

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  • State description for the root-signal set of median filters

    Publication Year: 1982, Page(s):894 - 902
    Cited by:  Papers (79)  |  Patents (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (744 KB)

    Median filtering is a simple digital technique for smoothing signals. One main characteristic of the filter is that it maps the input signal space into a root signal space, where signals invariant to median filters are called roots of the signal. In this paper, we develop the theory for the root signal set of median filters. A tree structure for the root signal set is obtained for binary signals. ... View full abstract»

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  • Direct form expansion of the transfer function for a digital Butterworth low-pass filter

    Publication Year: 1982, Page(s):1004 - 1006
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (248 KB)

    For a digital Butterworth low-pass general-order filter, generalized expressions for the filter coefficients of the transfer function are derived, with representation in direct form as a single high-order filter, rather than as a combination of second-order and first-order subfilters. By using double precision to overcome the usual problems of numerical accuracy with the direct form, the advantage... View full abstract»

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  • Helium speech enhancement using the short-time Fourier transform

    Publication Year: 1982, Page(s):841 - 853
    Cited by:  Papers (9)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1744 KB)

    Speech produced in a hyperbaric helium-oxygen atmosphere suffers a variety of distortions which render it virtually unintelligible. This paper describes a new system for helium speech enhancement based on a short-time Fourier transform signal representation. The algorithm is robust, allows nonlinear warping of the spectral envelope, and includes provisions for generating the enhanced speech at a r... View full abstract»

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  • The application of random reference sequences to the reconstruction of clipped differentiable signals

    Publication Year: 1982, Page(s):953 - 963
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1032 KB)

    The estimation of analog waveforms s(t) from their clipped contaminated samples is considered. For differentiable signals s(t), appropriate estimates are established and their performance is investigated: Error bounds and convergence rates are derived which improve with the degree of differentiability of the signal s(t). The analytical results are supplemented by a simulation study. View full abstract»

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  • The margin of stability of 2-D linear discrete systems

    Publication Year: 1982, Page(s):869 - 873
    Cited by:  Papers (30)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (488 KB)

    In this paper the margin of stability for a 2-D discrete system is considered. The definition of the margin of stability for the 1-D case is extended to the 2-D case in terms of analytic regions of rational functions in two variables. The relationship between the newly defined stability margin and the impulse response is established. A method to compute the stability margin is presented and illust... View full abstract»

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  • FIR filter structures having low sensitivity and roundoff noise

    Publication Year: 1982, Page(s):913 - 920
    Cited by:  Papers (10)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (800 KB)

    A class of structures for FIR filters is presented, which exhibits reduced coefficient sensitivity and superior roundoff noise properties as compared to the direct form realization. It is shown that using fixed-point arithmetic, these structures achieve the same accuracy and about the same roundoff noise as those obtained in the floating-point implementationn. The optimum structure to achieve the ... View full abstract»

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  • Time-varying wave digital filters for modeling analog systems

    Publication Year: 1982, Page(s):864 - 868
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (592 KB)

    The theory of wave digital filters, which are well suited to model analog networks including those containing transmission lines, is extended to the time-varying case. This allows simulation of the moving vocal tract or parametric amplifiers. A representation consistent with the physics of time-varying reactances and transmission lines requires not only time-varying filter coefficients, but also s... View full abstract»

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  • Fast polynominal transforms for two-dimensional convolution

    Publication Year: 1982, Page(s):1007 - 1010
    Cited by:  Papers (2)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (376 KB)

    In a recent paper [1], Truong et al. have presented a new method for computing two-dimensional convolutions. It is shown that a modification of their algorithm can lead to a slightly more efficient method, i.e., a gain in efficiency of 5-10 percent can be attained. View full abstract»

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  • Unstable digital filters and their equivalent stable filters

    Publication Year: 1982, Page(s):903 - 906
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (512 KB)

    Unstable digital filters have divergent (or nonconvergent) z transforms. The Mellin integral representation of such series can be written down by inspection. A change in the path of integration gives the equivalent convergent series, and hence the stable filter. To carry out this procedure the analytic form of the original filter coefficients is required. If the analytic form is not known directly... View full abstract»

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  • Estimating the parameters of exponentially damped sinusoids and pole-zero modeling in noise

    Publication Year: 1982, Page(s):833 - 840
    Cited by:  Papers (529)  |  Patents (7)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (720 KB)

    We have presented techniques [1] - [6] based on linear prediction (LP) and singular value decomposition (SVD) for accurate estimation of closely spaced frequencies of sinusoidal signals in noise. In this note we extend these techniques to estimate the parameters of exponentially damped sinusoidal signals in noise. The estimation procedure presented here makes use of "backward prediction" in additi... View full abstract»

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  • Noise reduction strategies for digital filters: Error spectrum shaping versus the optimal linear state-space formulation

    Publication Year: 1982, Page(s):963 - 973
    Cited by:  Papers (56)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1032 KB)

    The noise reduction performance of error spectrum shaping (ESS) structures and of the optimal linear state-space (LSS) structure are compared for second-order digital filter sections. It is shown that optimal direct form 1 and direct form 2 ESS realizations have a higher signal-to-noise ratio than the optimal LSS structure. In practice, suboptimal ESS structures with simple hardware implementation... View full abstract»

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  • Recursive lattice forms for spectral estimation

    Publication Year: 1982, Page(s):920 - 930
    Cited by:  Papers (12)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (912 KB)

    A class of lattice prediction filters is proposed for high resolution spectral estimation. The square-root normalized lattice recursions are used to estimate a set of reflection coefficients from the data. The lattice variables determine the coefficients of a least-squares predictor, from which the spectrum can be evaluated. The prewindowed and sliding window (covariance) cases are considered for ... View full abstract»

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  • A conversational test for comparing voice systems using working two-way communication links

    Publication Year: 1982, Page(s):853 - 863
    Cited by:  Papers (3)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1224 KB)

    A conversational test using live two-way communications provides a measure of the actual usability of voice systems, especially when voice quality is degraded. A conversational test developed at NRL was compared with two other communicability tests in a series of experiments using a variety of digital voice processors with data rates from 800 to 32 000 bits/s. All three tests ranked the voice proc... View full abstract»

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  • Comparison of the effects of quantization on digital filters

    Publication Year: 1982, Page(s):1010 - 1013
    Cited by:  Papers (1)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (448 KB)

    The fixed-point roundoff noise and coefficient sensitivity properties of the cascade canonic, Sedlmeyer-Fettweiss wave digital filter, the digital generalized immittance converter filter, and the Swamy-Thyagarajan wave digital filter are examined. The relative power spectral density and the statistical wordlength of each structure are computed. View full abstract»

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  • Generation of a random sequence having a jointly specified marginal distribution and autocovariance

    Publication Year: 1982, Page(s):973 - 983
    Cited by:  Papers (60)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (1088 KB)

    We consider the problem of generating a random sequence with a specified marginal distribution and autocovariance. The proposed scheme for generating such a sequence consists of a white Gaussian noise source input to a linear digital filter followed by a zero-memory nonlinearity (ZMNL). The ZMNL is chosen so that the desired distribution is exactly realized and the digital filter is designed so th... View full abstract»

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  • Minimum sensitivity realization of second order recursive digital filter

    Publication Year: 1982, Page(s):930 - 937
    Cited by:  Papers (9)
    Request permission for commercial reuse | Click to expandAbstract | PDF file iconPDF (832 KB)

    In the fixed point second order digital filter, the sensitivity problem which arises from the coefficient truncation is studied. Based on the definition of state space eigenvalues sensitivity function, a new low-sensitivity realization is found. It is applicable to both complex conjugate poles and real poles filters. The dynamic range of the implemented output coefficients and the number of multip... 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