By Topic

Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 6 • Date December 1983

Filter Results

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

    Publication Year: 1983 , Page(s): 0
    Save to Project icon | Request Permissions | PDF file iconPDF (226 KB)  
    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)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (453 KB)  

    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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • [Back cover]

    Publication Year: 1983 , Page(s): c4
    Save to Project icon | Request Permissions | PDF file iconPDF (3778 KB)  
    Freely Available from IEEE
  • Speed measurement by cross correlation--Theoretical aspects and applications in the paper industry

    Publication Year: 1983 , Page(s): 1374 - 1378
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (536 KB)  

    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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Iterative inverse filtering approach to the estimation of frequencies of noisy sinusoids

    Publication Year: 1983 , Page(s): 1456 - 1463
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (856 KB)  

    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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Two-dimensional root structures and convergence properties of the separable median filter

    Publication Year: 1983 , Page(s): 1350 - 1365
    Cited by:  Papers (35)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1896 KB)  

    The root (signals invariant to filtering) structures of the two-dimensional separable median filter are derived and presented. In addition, it is proved that with rare exception, after repetitive passes of the separable median filter any two-dimensional signal will be reduced to a signal containing only root structures. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Minimal delay realization of first order 2-D all-pass digital filters

    Publication Year: 1983 , Page(s): 1577 - 1579
    Cited by:  Papers (11)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB)  

    A method is proposed to realize a first-order 2-D all-pass transfer function with two delays and six multipliers. Stability constraints on the coefficients of first-order 2-D filters are used to guarantee the existence of real-gain multipliers for the realized filter. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • An error analysis for 2-D block implemented digital filters

    Publication Year: 1983 , Page(s): 1570 - 1573
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (376 KB)  

    This correspondence considers error analysis of block-implemented 2-D digital filters. Expressions for error bound and mean-square error for roundoff error accumulation are derived using fixed-point arithmetic, and compared with the results obtained using ordinary 2-D difference equations. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Derivative constraints for broad-band element space antenna array processors

    Publication Year: 1983 , Page(s): 1378 - 1393
    Cited by:  Papers (121)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1160 KB)  

    In this paper a class of linear constraints, also termed as derivative constraints, which is applicable to broad-band element space antenna array processors, is presented. The performance characteristics of the optimum processor with derivative constraints are demonstrated by computer studies involving two types of array geometries, namely linear and circular arrays. As a consequence of derivative constraints, the beam width in the look direction can be made as broad as desired and the beam spacings can be selected without fear of substantial signal suppression in the event of signal arrivals between beams. However, this increased beam width is achieved at the price of reducing array gain. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Memory intensive multipliers for signal processing

    Publication Year: 1983 , Page(s): 1579 - 1582
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (408 KB)  

    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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Pole-zero modeling of speech based on high-order pole model fitting and decomposition method

    Publication Year: 1983 , Page(s): 1556 - 1565
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1080 KB)  

    In this paper four pole-zero modeling algorithms of clean and noisy speech have been studied in a unified approach that is based on high-order pole model fitting and decomposition method. They are autocorrelation prediction (AP), modified Yule-Walker (MYW), modified least square (MLS), and modified least square with autocorrelation compensation (MLSAC) methods. They involve only linear equations, and therefore are computationally efficient. Among these algorithms, the MLSAC method appears to be the most effective in spectral envelope estimation of noisy as well as clean speech. According to our simulation results, the improvement resulting from the use of the MLSAC pole-zero model for noisy speech is equivalent to increasing signal-to-noise ratio (SNR) by about 5 dB when SNR of input speech is 10 dB or less. The use of a pole-zero model in multirate vocoding is also discussed. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • The signal subspace approach for multiple wide-band emitter location

    Publication Year: 1983 , Page(s): 1502 - 1522
    Cited by:  Papers (57)  |  Patents (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (2056 KB)  

    The rational vector space generalization of the signal subspace approach is presented and applied to the estimation of multiple wide-band emitter locations from the signals received at multiple sensors. The signal subspace and array manifold concepts first introduced by Schmidt are generalized to rational vector space. These concepts are used to develop the rational signal subspace theory and prove the signal subspace theorem, on which signal subspace algorithms are based. The theory is applied in discrete time to derive a class of rational signal subspace algorithms for source location and spectral estimation using unit circle eigendecomposition of multivariate rational models of sensor outputs. Simulation results are presented for an algorithm in this class, including sample statistics from Monte Carlo trials and comparisons with the Cramer-Rao bound. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • The noise gain of a digital butterworth ladder filter

    Publication Year: 1983 , Page(s): 1576 - 1577
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (208 KB)  

    We show that any digital Butterworth ladder filter has a noise gain of N/2, where N is the filter order. The result is obtained by linking the noise gain to the average time delay of an analog Butterworth filter, which we show to be N/4E. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Computer analysis and perception, vol. II: Auditory signals

    Publication Year: 1983 , Page(s): 1587 - 1588
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (344 KB)  

    First Page of the Article
    View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • 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»

    Full text access may be available. Click article title to sign in or learn about subscription options.

Aims & Scope

This Transactions ceased production in 1990. The current retitled publication is IEEE Transactions on Signal Processing.

Full Aims & Scope