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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 4 • Date August 1978

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Displaying Results 1 - 25 of 27
  • [Front cover and table of contents]

    Publication Year: 1978 , Page(s): 0
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    Freely Available from IEEE
  • Editorial thanks for support

    Publication Year: 1978 , Page(s): 277
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    Freely Available from IEEE
  • Fortran control of real-time signal processing with high-speed processors

    Publication Year: 1978 , Page(s): 278 - 284
    Cited by:  Papers (1)
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  • A note on "An lp design technique for two-dimensional digital recursive filters"

    Publication Year: 1978 , Page(s): 371 - 372
    Cited by:  Papers (4)
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  • [Back cover]

    Publication Year: 1978 , Page(s): c4
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    Freely Available from IEEE
  • Design and implementation of two-dimensional FIR digital filters with nonrectangular arrays

    Publication Year: 1978 , Page(s): 314 - 318
    Cited by:  Papers (2)
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    A new technique for the design of two-dimensional FIR digital filters is presented in this paper. A high-order impulse response array of a filter may be optimally truncated to give a low-order impulse response array. In general, such low-order arrays are nonrectangular. A method for implementing the filters is presented. Examples of the design and implementation techniques are included. View full abstract»

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  • Speech intelligibility enhancement in a power generating noise environment

    Publication Year: 1978 , Page(s): 378 - 380
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (344 KB)  

    A previous paper described a method for processing speech which resulted in the enhancement of speech intelligibility in the presence of high noise levels at the listener. This processing technqiue consists of high-pass filtering followed by rapid amplitude compression. A significant intelligibility gain was achieved over unprocessed speech for signal-to-noise ratios from -10 to +10 dB (noise at 90 dB). In this previous work, only white noise was considered. In the present paper, results are presented for this processing technique for a listener in the environment of power generating noise. View full abstract»

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  • Real-time harmonic pitch detector

    Publication Year: 1978 , Page(s): 358 - 365
    Cited by:  Papers (14)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (888 KB)  

    A real-time harmonic pitch detection algorithm has been developed on the Lincoln Digital Voice Terminal (LDVT). The algorithm was designed to be fast and to perform well when the input speech is degraded (i.e., telephone quality) or corrupted with acoustically coupled noise. The algorithm determines the fundamental frequency from the spacing between harmonics in a selected portion of the spectrum. The algorithm was incorporated into a real-time linear prediction vocoder and compared favorably in informal listening tests with the Gold-Rabiner time-domain detector under a variety of adverse conditions. View full abstract»

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  • An efficient algorithm for bilinear transformation of multivariable polynomials

    Publication Year: 1978 , Page(s): 380 - 381
    Cited by:  Papers (13)
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    An algorithm for bilinear-transforming multivariable polynomials is presented. It is similar to but simpler and computationally more efficient than existing algorithms. View full abstract»

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  • On the optimal number of finite-duration impluse-response (FIR) filter coefficients under a memory size constraint

    Publication Year: 1978 , Page(s): 366 - 367
    Cited by:  Papers (1)
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    A method is given for efficiently assessing the performance of a finite-duration impulse-response (FIR) digital filter under the constraint of fixed (number of coefficients) × (word length) product. The tradeoff of impulse-response truncation error against coefficient quantization error is readily evaluated. The method yields a quantitative estimate of the optimal solution and of the sensitivity of this solution without the need for extensive filter simulations. View full abstract»

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  • More accurate interpolation using discrete Fourier transforms

    Publication Year: 1978 , Page(s): 368 - 369
    Cited by:  Papers (5)
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    It is shown that substantial errors in interpolation by means of existing discrete Fourier transform (DFT) techniques are generally present for finite sequences, even if the function sampled is band-limited. A modified approach is proposed which provides significantly more accurate information. View full abstract»

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  • Optimum element placement for passive bearing estimation in unequal signal-to-noise ratio environments

    Publication Year: 1978 , Page(s): 365 - 366
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (248 KB)  

    Consider that each of a fixed number of receiving hydrophones can be placed at one of two locations to estimate the bearing to an acoustic source. For high-output signal-to-noise ratio (SNR) the hydrophones should be divided to balance the output voltage SNR. However, for low-output SNR the hydrophones should be equally divided. View full abstract»

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  • p-normalized digital two-pairs

    Publication Year: 1978 , Page(s): 374 - 376
    Cited by:  Papers (1)
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    A digital two-pair constrained by any unity magnitude all-pass transfer function and exhibiting unity Lp-scaling norms of the transfer functions at all pertinent nodes is defined to be a p-normalized digital two-pair. Several such first-order two-pairs are developed, and the advantage of using these two-pairs in the realization of an arbitrary infinite-duration impulse-response (IIR) digital transfer function is pointed out. View full abstract»

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  • Receiver operating characteristic of a thresholded sum-squared coherence detector

    Publication Year: 1978 , Page(s): 369 - 371
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    Given two signals x(t), y(t) observed on the time interval [0, t], a detector is desired which will decide whether y(t) contains only noise uncorrelated with x(t) (hypothesis H1or noise plus signal linearly related to x(t) (hypothesis H0}). Previous results have given the receiver operating characteristic (ROC) of a coherence detector at each point in frequency. A natural extension of this "point" detector is to band detection, whereby all coherence information over a test band is used to make a decision as to whether or not there is a signal present. The sum of the magnitude squared coherence over the band of test provides a statistic for decision. Under the assumption that this statistic is Gaussian, the ROC is found, compared for the special case to the one-point result, and plotted for typical multipoint situations. View full abstract»

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  • A class of all-zero lattice digital filters: Properties and applications

    Publication Year: 1978 , Page(s): 304 - 314
    Cited by:  Papers (115)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1016 KB)  

    A class of minimum- or maximum-phase all-zero lattice digital filters, based on the two-multiplier lattice of Itakura and Saito, is developed. Different lattice forms with different numbers of multipliers are derived, including two one-multiplier forms. Many of the properties of these lattice filters are given, including the important orthogonalization and decoupling properties of successive stages in optimal inverse filtering of signals. These properties lead to important applications in the areas of adaptive linear prediction and adaptive Wiener filtering. As a specific example, the design of a new fast start-up equalizer is presented. View full abstract»

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  • Evaluation of an adaptive comb filtering method for enhancing speech degraded by white noise addition

    Publication Year: 1978 , Page(s): 354 - 358
    Cited by:  Papers (38)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (528 KB)  

    An intelligibility test was performed to evaluate an adaptive comb filtering method proposed by Frazier [2] for enhancement of degraded speech due to additive white noise. Results indicate that independent of S/N ratio the adaptive comb filtering scheme does not increase speech intelligibility. View full abstract»

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  • Fast algorithms for computing the powers of a primitive element in GF(q2)

    Publication Year: 1978 , Page(s): 376 - 378
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    In this correspondence a new algorithm for computing efficiently multiplications by powers of a primitive element in the finite field GF(q2), where q is a Mersenne prime, is described. This algorithm is applicable to transforms over GF(q2) which is used to implement fast circular convolutions without roundoff error. View full abstract»

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  • Transform-domain digital filtering with number theoretic transforms and limited word lengths

    Publication Year: 1978 , Page(s): 284 - 290
    Cited by:  Papers (5)
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    While the discrete Fourier transform (DFT) is defined in the field of complex numbers, number theoretic transforms (NTT's) operate in finite rings and fields. Some of these NTT's have a fast-transform structure similar to that of the fast Fourier transform (FFT) and can be used for fast digital signal processing. Both the computational effort and the signal-to-noise ratio (SNR) performance of transform-domain signal processing with NTT's are investigated in this paper. In particular, the effect of limited word lengths, i.e., b \leq 16 , and long transform lengths on the SNR of NTT filtering is analyzed. For small word lengths and/or moderate to large transform lengths, NTT filtering is shown to achieve a better SNR than FFT filtering with fixed-point arithmetic. Finally, new NTT's with a single- or mixed-radix fast-transform structure are presented. While these NTT's require efficient implementations of modulo arithmetic operations, their transform length is optimum for any given work length b in the range 8 \leq b \leq 16 . View full abstract»

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  • A design technique for circularly symmetric low-pass filters

    Publication Year: 1978 , Page(s): 290 - 304
    Cited by:  Papers (28)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1184 KB)  

    The design of two-dimensional (2-D) circularly-symmetric low-pass digital filters by cascading several rotated filters (a rotated filter is defined to be one produced by rotating a one-dimensional (1-D) continuous filter into a two-dimensional continuous filter which is in turn bilinearly transformed into a two-dimensional digital filter) is a well-known and useful technique. An alternate approach which is an extension of the above technique is presented. This new method is based on a spectral transformation from the one-dimensional discrete domain into the two-dimensional discrete domain. This approach retains most of the advantages of the original technique while permitting design entirely in the discrete domain, yielding filters with better stability characteristics, and facilitating frequency response optimization via nonlinear programming. View full abstract»

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  • Applications of short-time homomorphic signal analysis to seismic wavelet estimation

    Publication Year: 1978 , Page(s): 343 - 353
    Cited by:  Papers (3)
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    The use of homomorphic systems to deconvolve seismic reflection and teleseismic data has been proposed and explored by a number of researchers with varying success. A careful study of the methods employed reveals a number of problems involving both the class of characteristic systems used and their numerical implementation. Furthermore, the analysis strategies used introduce deterministic constraints on the estimation of the seismic wavelet which may lead to serious problems in determining the earth impulse response. Several novel results are presented in this paper. A class of homomorphic systems matched to the bandpass nature of seismic signals is discussed and improved and more reliable implementation algorithms are developed. The concept of short-time wavelet estimation by homomorphic filtering is introduced. This technique takes into account the specific time-varying characteristics of seismic traces. Strategies for homomorphic wavelet estimation are proposed and illustrated. The recovery of the earth impulse response may then be accomplished by combining homomorphic wavelet estimation with parametric inverse filtering. View full abstract»

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  • Truncation error bounds for sampling expansions of sequency-band-limited signals

    Publication Year: 1978 , Page(s): 372 - 374
    Cited by:  Papers (4)
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    This paper discusses truncation error bounds for sampling expansions of finite-energy, sequency-band-limited signals. Results are compared with those obtained for frequency-band-limited signals which utilize cardinal series expansions. View full abstract»

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  • An efficient method for generating unaliased samples of certain signals

    Publication Year: 1978 , Page(s): 338 - 342
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (520 KB)  

    In simulating a continuous-time system being driven by a continuous-time signal f(t), we may wish to generate a discrete-time signal whose spectrum agrees with that of f(t) for frequencies up to the Nyquist frequency. One important example is provided by the simulation of the vocal tract driven by a triangularly shaped glottal pulse, although a similar problem arises whenever we simulate a known continuous-time input to a system. We define classes of signals, called H-synchronous signals, which, after sampling, can be "de-aliased" with a fixed digital filter depending on H. The McClellan-Parks-Rabiner algorithm is used to find FIR designs for such de-aliasing filters for the important special cases of piecewise-constant and piecewise-linear signals. View full abstract»

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  • Digital filter realizations without overflow oscillations

    Publication Year: 1978 , Page(s): 334 - 338
    Cited by:  Papers (91)
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    Most of the literature dealing with overflow oscillation in fixed-point arithmetic digital filters has considered the direct form exclusively. It is possible to eliminate overflow oscillations, regardless of pole locations, by considering more general forms. A sufficient condition is given for a two's complement state variable realization of any order to be free of overflow oscillation. A simple characterization of the condition is given for second-order filters. Among those second-order forms which meet the condition are normal forms, and forms which minimize output roundoff noise. View full abstract»

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  • Signal analysis

    Publication Year: 1978 , Page(s): 386
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  • FIR system modeling and identification in the presence of noise and with band-limited inputs

    Publication Year: 1978 , Page(s): 319 - 333
    Cited by:  Papers (27)
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    System identification, that is, the modeling and identification of a system from knowledge of its input and output signals, is a subject that is of considerable importance in many areas of signal and data processing. Because of the diversity of applications, a number of different methods for system identification with different advantages and disadvantages have been described and used in the literature. In this paper we investigate the performance of three well-known system identification methods based on an FIR (finite impulse response) model of the system. The methods will be referred to in this paper as the least squares analysis (LSA) method, the least mean squares adaptation algorithm (LMS), and the short-time spectral analysis (SSA) procedure. Our particular interest in this paper concerns the performance of these algorithms in the presence of high noise levels and in situations where the input signal may be band-limited. Both white and nonwhite random noise signals as well as speech signals are used as test signals to measure the performance of each of the system identification techniques as a function of the signal-to-noise ratio of the systems output. Quantitative results in terms of an accuracy measure of system identification are presented and a simple analytical model is used to explain the measured results. 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