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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 6 • Date December 1983

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Displaying Results 1 - 25 of 34
  • [Front cover and table of contents]

    Publication Year: 1983 , Page(s): 0
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    Freely Available from IEEE
  • Comments on "A recursive Kalman window approach to image restoration"

    Publication Year: 1983 , Page(s): 1573 - 1576
    Cited by:  Papers (1)
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    In a recent paper, a recursive Kalman window estimation procedure for image restoration was claimed to be at least nearly optimal. Here, we show that it is not and point out some basic model errors. View full abstract»

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  • [Back cover]

    Publication Year: 1983 , Page(s): c4
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    Freely Available from IEEE
  • The Carathéodory-Fejér method for recursive digital filter design

    Publication Year: 1983 , Page(s): 1417 - 1426
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1000 KB)  

    A new technique for rational digital filter design is presented which is based on results in complex function theory due to Takagi, Krein, and others. Starting from a truncated or windowed impulse response, the method computes the unique optimum rational Chebyshev approximation with a prescribed number of stable poles. Both phase and magnitude are matched. Deleting the noncausal (unstable) part of the Chebyshev approximation yields a stable approximation of specified order (M,N) which is close to optimal in the Chebyshev sense. No iteration is involved except in the determination of an eigenvalue and eigenvector of the Hankel matrix of impulse response coefficients. In this paper the algorithm is specified and practical examples are discussed. View full abstract»

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  • Delay and Doppler estimation by time-space partition of the array data

    Publication Year: 1983 , Page(s): 1523 - 1535
    Cited by:  Papers (17)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1112 KB)  

    An instrumentationally attractive technique for estimating the Doppler time-compression and its time derivatives is proposed and analyzed in this paper. The basic scheme consists of (M - 1) generalized cross correlators (GCC's) in parallel (where M is the number of sensors in the receiving array), each of which is implemented successively to generate a sequence of differential delay measurements. The resulting spatial-temporal differential delay data are then compressed into the estimates of the Doppler and its derivatives using a least squares (LS) filter. It is shown that for signal-to-noise ratio (SNR) conditions above the threshold level, the indicated two-step estimation procedure is very nearly optimal. When operating near the threshold point, however, the proposed scheme is only distinctly suboptimal. View full abstract»

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  • Efficient sampling rate alteration using recursive (IIR) digital filters

    Publication Year: 1983 , Page(s): 1366 - 1373
    Cited by:  Papers (30)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (776 KB)  

    A general scheme for changing sampling rates is developed, and a method of designing recursive (IIR) filters for use in this general scheme is presented. If phase response is ignored, then this scheme provides a considerable savings in computations over nonrecursive (FIR) schemes. The use of recursive filters with approximately linear phase also led to a performance better than that using nonrecursive filters. The quantization effects in the new scheme are minimal. View full abstract»

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  • Fast Fourier transform algorithms for linear estimation, smoothing and Riccati equations

    Publication Year: 1983 , Page(s): 1435 - 1446
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (944 KB)  

    In the past two decades since the advent of Kalman's recursive filter, numerous algorithms for linear estimation have emerged. Most of these algorithms are recursive and rely on solving a Riccati equation or equivalent recursive equations. It will be shown how some of the classical problems such as linear smoothing, Riccati equations, boundary value problems, and recursive block filtering problems can be solved exactly by some new nonrecursive algorithms which are based on the fast Fourier transform (FFT). In the context of modern digital signal processing these algorithms have a highly parallel structure and are well suited for VLSI implementations. View full abstract»

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  • An algorithm for LPC synthesis gain matching

    Publication Year: 1983 , Page(s): 1566 - 1569
    Cited by:  Papers (3)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (496 KB)  

    A simple method of synthesis gain matching in a linear prediction (LP) vocoder makes use of the LP analysis residual energy. However, poor gain matching is to be expected when a low-frequency formant is in resonance with the voiced excitation impulses. Such large gain errors increase the probability of synthesis filter overflow. A simple improvement of this method is suggested, reducing these large errors substantially. The improvement makes use of the information provided by the derivative of the already synthesized signal. The method can be applied internally or externally to low-complexity real-time speech synthesizers. View full abstract»

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  • Automatic speech analysis and recognition: Proceedings of the NATO advanced study institute held at Bonas, France

    Publication Year: 1983 , Page(s): 1588 - 1589
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (288 KB)  

    First Page of the Article
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  • A fast quadratic programming algorithm for positive signal restoration

    Publication Year: 1983 , Page(s): 1337 - 1341
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (664 KB)  

    When processing a signal or picture by deconvolution, any additional a priori information is of prime interest since it can potentially lead to an improvement in results and to superresolution. In this framework, the positivity of the unknown signal is a current situation (each time this unknown is an intensity, a probability distribution, a histogram, etc.) but it involves a nonlinear constraint which is difficult to take into account. In this paper we state the problem in terms of a quadratic programming problem with positivity constraints and we propose a new algorithm derived from a conjugate gradient method, especially suited to this particular situation. It leads to a low cost solution. We then present experimental results on two-dimensional signals emphasizing relevant superresolution. View full abstract»

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  • Some aspects of band-limited signal extrapolation: Models, discrete approximations, and noise

    Publication Year: 1983 , Page(s): 1492 - 1501
    Cited by:  Papers (27)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (960 KB)  

    We present some theoretical results on the band-limited signal extrapolation problem. In Section I we describe four basic models for the extrapolation problem. These models are useful in understanding the relationship between the continuous extrapolation problem and some discrete algorithms given in [1] and [2]. One of these models was shown to approximate the continuous band-limited extrapolation problem [3]. Another model is obtained when the discrete Fourier transform (DFT) is used to implement the well-known iterative algorithm given in [4] and [5] which was designed for solving the continuous extrapolation problem; in Section II this model is related to the continuous model by means of an interesting approximation theorem. Also, an important conjecture is presented. Section III shows some approximation results. Specifically, we prove that some discrete-discrete and discrete-continuous extrapolations of noisy signals converge to solutions of a certain continuous-continuous noisy extrapolation problem when the noise η is bounded by a known number, max \eta(x)| \leq \epsilon . This convergence is obtained by using normal families of entire functions in ¢nand some other complex analysis tools. We also show that the extrapolation problem is very sensitive to noise even in cases where only small amounts of extrapolation are desired. This result indicates that in the presence of noise, extrapolation techniques should be used judiciously in order to obtain reasonable results. View full abstract»

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  • Engine noise: Excitation, vibration, and radiation

    Publication Year: 1983 , Page(s): 1586 - 1587
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (328 KB)  

    First Page of the Article
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  • A fast sequential algorithm for least-squares filtering and prediction

    Publication Year: 1983 , Page(s): 1394 - 1402
    Cited by:  Papers (138)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (800 KB)  

    A new computationally efficient algorithm for sequential least-squares (LS) estimation is presented in this paper. This fast a posteriori error sequential technique (FAEST) requires 5p MADPR (multiplications and divisions per recursion) for AR modeling and 7p MADPR for LS FIR filtering, where p is the number of estimated parameters. In contrast the well-known fast Kalman algorithm requires 8p MADPR for AR modeling and 10p MADPR for FIR filtering. The increased computational speed of the introduced algorithm stems from an alternative definition of the so-called Kalman gain vector, which takes better advantage of the relationships between forward and backward linear prediction. View full abstract»

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  • A generalization of median filtering using linear combinations of order statistics

    Publication Year: 1983 , Page(s): 1342 - 1350
    Cited by:  Papers (273)  |  Patents (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1016 KB)  

    We consider a class of nonlinear filters whose output is given by a linear combination of the order statistics of the input sequence. Assuming a constant signal in white noise, the coefficients in the linear combination are chosen to minimize the output MSE for several noise distributions. It is shown that the optimal order statistic filter (OSF) tends toward the median filter as the noise becomes more impulsive. The optimal OSF is applied to an actual noisy image and is shown to perform well, combining properties of both the averaging and median filters. A more general design scheme for applications involving nonconstant signals is also given. View full abstract»

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  • Iterative inverse filtering approach to the estimation of frequencies of noisy sinusoids

    Publication Year: 1983 , Page(s): 1456 - 1463
    Cited by:  Papers (6)
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    A signal modeling approach is proposed to reduce the problem of estimating frequencies of noisy sinusoids to the parameter estimation problem. Both white and colored noise are considered. The maximum likelihood approach leads to an algorithm computationally inefficient for low values of signal-to-noise ratio. Based on the equation-error formulation, an iterative inverse filtering algorithm is derived in the case of colored noise, and a generalized least-squares algorithm in the case of white noise. It is shown, on the basis of numerous experimental results, that the iterative inverse filtering algorithm provides highly accurate estimates of unknown frequencies for low values of signal-to-noise ratio, even in the case of a small number of sampling points. View full abstract»

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  • High-resolution emitter direction finding using a phased array antenna

    Publication Year: 1983 , Page(s): 1409 - 1416
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (760 KB)  

    Signals from several uncorrelated emitters are received by a uniformly spaced linear phased array, and the bearing angles of the emitters are determined from output power measurements only. The output power is used to estimate the correlation matrix of the signals present at the element level. The beampointing and dwell-time agilities of a phased array are exploited, such that errors in the matrix estimate are minimized, subject to a constraint on the data collection time. A simple spectral estimation technique is described which improves on the accuracy and resolution of the maximum entropy spectrum while suppressing spurious responses. Simulations demonstrate clear resolution and good accuracy for closely spaced emitters using practical SNR's and data collection time. View full abstract»

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  • Design of digital fan filters

    Publication Year: 1983 , Page(s): 1427 - 1434
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (856 KB)  

    A design technique is developed for a digital fan-pass filter. The technique involves the rotation of the magnitude response of a recursive half-plane symmetry filter which is, in turn, realized by shifting an ideal two-dimensional low-pass filter. The rotation has the effect of introducing the basic delay unit z-1/2instead of z-1. In order to overcome this effect, two realization schemes are proposed. The concepts underlying these realization schemes may be useful in their own right in realizing any other transfer function containing other basic delay units involving rational power, namely z-1/3, z-1/4, etc. The design of the fan-reject filter is an obvious parallel to that of the fan-pass filter developed in this paper. A design example is worked out, and implementational complexities are discussed. View full abstract»

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  • Speed measurement by cross correlation--Theoretical aspects and applications in the paper industry

    Publication Year: 1983 , Page(s): 1374 - 1378
    Cited by:  Papers (6)
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    Based on the cross correlation of two signals, the speed measurement method presented in this paper is to be used on materials whose structure is stable (railway, papersheet, etc.) or not (turbulent flows). There is not only a time delay between the signals, but more generally a filtering effect which may be regarded as linear. In these conditions, the theoretical characteristics (bias, variance) of the estimation are given. They are compared with simulation results. The influence of the computational rapidity of the electronic circuit on the quality of the estimation is studied. View full abstract»

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  • Estimation of time differences of arrival by pole decomposition

    Publication Year: 1983 , Page(s): 1478 - 1492
    Cited by:  Papers (19)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1616 KB)  

    Time differences of arrival (TDOA's) of emitter wave fronts to a spatially distributed array of sensors can be used to determine the source location. In this paper, we suggest a new method of TDOA estimation for multiple unknown autoregressive moving average (ARMA) sources and additive noise that may be correlated between the sensors. We derive a theoretical formula that only uses the receiver cross spectra and the source poles for the TDOA determination. The poles are estimated by a least squares technique, and two methods are suggested for the estimation of the cross spectra which allow tradeoffs between computational complexity and accuracy. A new time delay model is derived and used to show the applicability of the methods for noninteger TDOA's. Results from simulations illustrate the performance of the algorithm by Monte Carlo trials and compare it to the Cramer-Rao bound. View full abstract»

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  • Systematic design and programming of signal processors, using project management techniques

    Publication Year: 1983 , Page(s): 1536 - 1549
    Cited by:  Papers (18)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1640 KB)  

    This paper presents a systematic method of designing and microprogramming fast signal processors (SP's) while optimally utilizing the inherent parallelism of a given algorithm. The method employs graph-theoretical concepts and scheduling techniques from "project management" and permits an easy evaluation of the following key design parameters: i) the lower bound on the algorithm duration (Tmp) for given speeds of the arithmetic components in a signal processor, ii) the minimum arithmetic hardware requirements necessary for the completion of the algorithm within the time Tmp, and iii) the optimum algorithm schedule and the corresponding minimum attainable duration Tat, when constraints on the available SP resources are imposed. In combination with the data-flow analysis and resource allocation, the scheduling procedure permits the influence of the SP architecture on the schedule to be modeled and the basic architectural features of the SP to be determined. The results also provide information for automatic microprogram generation and for the assessment and comparison of signal processor performance and algorithm speeds. The synthesis of architectures is based on an "ideal, data-flow driven signal processor" that is introduced in the paper. The proposed design approach is demonstrated on various digital-filter algorithms. View full abstract»

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  • High-speed normalization and rounding circuits for pipelined floating-point processors

    Publication Year: 1983 , Page(s): 1403 - 1408
    Cited by:  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (648 KB)  

    A fast leading/trailing-zero detection circuit (LZDC/TZDC) is described, and then applied to the design of a pipelined floating-point (FLP) processor. This circuit has a total delay of 5Δ and a hardware complexity of (m + 1)(3 \log _{2}(p) + 4) where Δ, m , and p are the unit gate delay, subword number, and subword partition length, respectively. Applications of this circuit to both postnormalization and rounding are presented, including circuits for normalization, sticky bit generation, and increment-by-one. The LZDC/TZDC has two important features-modularity and expandability-which make it particularly well suited for VLSI implementation. View full abstract»

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  • Memory intensive multipliers for signal processing

    Publication Year: 1983 , Page(s): 1579 - 1582
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    Digital signal processing is an arithmetic-intensive operation. Over the past decade, several memory-intensive multiplication algorithms have been reported. They offer a high-speed alternative to traditional methods. In this correspondence, an error analysis is performed on this class of multiplier and an extended precision version is derived. The new multiplier is capable of adding several additional bits of precision to a fixed point product without significantly increasing hardware complexity. View full abstract»

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  • Roundoff noise and scaling in the digital implementation of control compensators

    Publication Year: 1983 , Page(s): 1464 - 1477
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1520 KB)  

    Researchers in digital signal processing have examined at length the effects of finite wordlength in the design of digital filters. The issues that have been considered apply to any digital system. In particular, the design of digital control systems must consider these issues. In this paper we will use, adapt, and extend the ideas developed in digital signal processing to the issue of roundoff noise in digital linear-quadratic-Gaussian (LQG) compensators. We will then examine the roundoff noise effects for a particular LQG example and several different implementation structures. View full abstract»

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  • Some experiments with a linguistic processor for continuous speech recognition

    Publication Year: 1983 , Page(s): 1549 - 1556
    Cited by:  Papers (5)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (968 KB)  

    The linguistic processor of a system for the recognition of fluently spoken Japanese is described. Input to the processor is a phoneme lattice in which scores are given for each of 27 phonemes for each segment. Phrases composed of words from a 112 word vocabulary are recognized by an error correcting parser which tolerates both classification and segmentation errors in the phoneme lattice by means of a Substitution-Insertion-Deletion Mechanism (SID). Some phrase errors are corrected at the sentence level by means of a Cartesian Product Sorting Algorithm. The processor has been tested on a total of 80 sentences from four male speakers. The sentences comprised 496 phrases with an average of 25 phonemes per phrase. The phoneme lattices had a 70 percent accuracy on phonemes with roughly equal numbers of segmentation and classification errors. Under these conditions 77 percent phrase accuracy was observed. View full abstract»

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  • Comparative study of nonlinear time warping techniques in isolated word speech recognition systems

    Publication Year: 1983 , Page(s): 1582 - 1586
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (488 KB)  

    In this paper, the effects of two major design choices on the performance of an isolated word speech recognition system are examined in detail. They are: 1) the choice of a warping algorithm among the Itakura asymmetric, the Sakoe and Chiba symmetric, and the Sakoe and Chiba asymmetric, and 2) the size of the warping window to reduce computation time. Two vocabularies were used: the digits (zero, one,..., nine) and a highly confusable subset of the alphabet (b, c, d, e, g, p, t, v, z). The Itakura asymmetric warping algorithm appears to be slightly better than the other two for the confusable vocabulary. We discuss the reasons why the performance of the algorithms is vocabulary dependent. Finally, for the data used in our experiments, a warping window of about 100 ms appears to be optimal. 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