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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 6 • Date December 1981

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Displaying Results 1 - 25 of 28
  • [Front cover and table of contents]

    Publication Year: 1981 , Page(s): 0
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    Freely Available from IEEE
  • Comment on "A useful property of the coefficients of a Walsh - Hadamard transform

    Publication Year: 1981 , Page(s): 1202
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  • Rebuttal to "Observations on the extrapolation of a band-limited signal problem"

    Publication Year: 1981 , Page(s): 1209
    Cited by:  Papers (1)
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  • [Back cover]

    Publication Year: 1981 , Page(s): c4
    Cited by:  Papers (2)
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    Freely Available from IEEE
  • Analysis of the steady-state performance in adaptive echo cancelers with correlated data and correlated received signal

    Publication Year: 1981 , Page(s): 1222 - 1225
    Cited by:  Papers (1)
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    Adaptive echo cancelers are used to allow full duplex data transmission on two-wire circuits. Analyses so far have assumed that both the transmitted data and the received signal consisted of independent samples. In this contribution the steady state performance is analyzed when both the data and the received signal have arbitrary correlations. Computer simulation results are presented to confirm the analytical results. View full abstract»

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  • Maximum entropy power spectrum estimation of signals with missing correlation points

    Publication Year: 1981 , Page(s): 1215 - 1217
    Cited by:  Papers (4)
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    A computationally simple algorithm has been recently proposed by Lim and Malik [1] to solve the two-dimensional (2-D) maximum entropy (ME) power spectrum estimation (PSE) problem. In this note, we illustrate that this algorithm also solves the ME PSE problem for both 1-D and 2-D signals when the region in which the correlation function is known has any arbitrary shape that includes the origin. View full abstract»

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  • A pulse driving function generator for LPC synthesis of voiced segments of speech

    Publication Year: 1981 , Page(s): 1113 - 1116
    Cited by:  Papers (2)  |  Patents (2)
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    Most of the pitch detection schemes for voiced segments of speech measure the pitch period as an integer multiple of the sampling period. Averaging techniques on more than one frame, for smoothing purposes, will result in pitch periods which are noninteger multiples of the sampling period. The same result is expected when interpolation takes place by changing the pitch several times within the same frame. Finally, a similar effect is observed, even without smoothing or interpolation, in those pitch detection schemes that use averaging of distances within the same frame to calculate the final pitch period for the frame. The algorithm described here permits the synthesis of male, female, and children's voices using noninteger pitches. The impact of this technique on current LPC implementations is marginal at low pitch frequencies while, at higher ones, the improvement is detectable. It is believed that as other sources of error in LPC systems are reduced, this technique will prove to be useful in improving synthetic speech quality. This paper deals with two items. It discusses the effects of truncation and rounding of noninteger pitch periods and describes the design of a pulse driving function generator, whose pitch period is a noninteger multiple of a fixed sampling period. View full abstract»

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  • Rotation of a two-dimensional sampling set using one-dimensional resampling

    Publication Year: 1981 , Page(s): 1218 - 1222
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    Generating intermediate sample points from a two-dimensional sample set generally requires two-dimensional filtering. Because true two-dimensional filtering requires a large number of computations, alternative filtering methods may be attractive. One alternative is to perform a series of one-dimensional filtering operations on individual lines of samples within the two-dimensional set. As long as the Nyquist sampling rate in both dimensions is satisfied at each intermediate step, valid sample values result. This paper analyzes a one-dimensional interpolation procedure for resampling a two-dimensional set of samples taken on an orthogonal grid into a second set of samples on an orthogonal grid which is rotated with respect to the first. The mathematical analysis quantifies the minimum sampling rate requirements, for a function with a rectangular band limit, as a function of the two-dimensional bandwidths and the angle of rotation between the two sampling grids. View full abstract»

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  • A simple method for determining the computable ordering of a digital network

    Publication Year: 1981 , Page(s): 1226
    Cited by:  Papers (1)
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    An algorithm is described for use in renumbering the nodes of a digital network to allow computation of the network equations in order. The method is not only easily programmed, but is simple and straightforward to perform by hand. View full abstract»

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  • Cubic convolution interpolation for digital image processing

    Publication Year: 1981 , Page(s): 1153 - 1160
    Cited by:  Papers (421)  |  Patents (63)
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    Cubic convolution interpolation is a new technique for resampling discrete data. It has a number of desirable features which make it useful for image processing. The technique can be performed efficiently on a digital computer. The cubic convolution interpolation function converges uniformly to the function being interpolated as the sampling increment approaches zero. With the appropriate boundary conditions and constraints on the interpolation kernel, it can be shown that the order of accuracy of the cubic convolution method is between that of linear interpolation and that of cubic splines. A one-dimensional interpolation function is derived in this paper. A separable extension of this algorithm to two dimensions is applied to image data. View full abstract»

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  • Corrections to "Fortran subroutines for the solution of Toeplitz sets of linear equations"

    Publication Year: 1981 , Page(s): 1212
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    Corrections and improvements to previously published subroutines are presented. View full abstract»

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  • The design of an optimal first-order quantizer structure for use in a recursive digital filter

    Publication Year: 1981 , Page(s): 1209 - 1210
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    Presented in [1] is a structure useful to increase SNR of a recursive filter with poles close to the unit circle. Presented within this note is a procedure to choose the quantizer coefficient optimally for arbitrarily located poles. The procedure is optimal in the sense that a system loss in SNR is not possible for any pole locations when compared to the Structure without the quantizer. View full abstract»

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  • Stability and overflow oscillations in 2-D state-space digital filters

    Publication Year: 1981 , Page(s): 1161 - 1171
    Cited by:  Papers (60)
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    An important theorem relating to the Lyapunov stability of two-dimensional discrete systems is proven. Using this theorem it is shown that for any 2-D digital filter satisfying Shanks' criterion there exists a realization that cannot support overflow oscillations. In the process of proving the theorem some interesting results on the multi-dimensional bilinear transformation are developed. One of these results yields a simple test that can be used to check the stability of a 2-D discrete transfer function that has been obtained from the bilinear transform of a 2-D continuous transfer function with a 2-D Hurwitzian denominator polynomial. A technique is given for determining whether a normal realization exists for a given 2-D discrete system. Also, a theorem is presented that allows the determination of the norm of the minimum norm realization of a given transfer function. A noniterative technique for obtaining a low norm realization and an iterative technique for obtaining a minimum norm realization are developed. View full abstract»

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  • A theoretical analysis of the properties of median filters

    Publication Year: 1981 , Page(s): 1136 - 1141
    Cited by:  Papers (296)  |  Patents (7)
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    Necessary and sufficient conditions for a signal to be invariant under a specific form of median filtering are derived. These conditions state that a signal must be locally monotone to pass through a median filter unchanged. It is proven that the form of successive median filtering of a signal (i.e., the filtered output is itself again filtered) eventually reduces the original signal to an invariant signal called a root signal. For a signal of length L samples, a maximum of frac{1}{2}(L - 2) repeated filterings produces a root signal. View full abstract»

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  • A parametric representation and a clustering method for phoneme recognition--Application to stops in a CV environment

    Publication Year: 1981 , Page(s): 1117 - 1127
    Cited by:  Papers (8)
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    A new method of representing phonemic categories and determining their standard values from a training sample distribution is presented. It is an essential part of a phoneme recognition system aiming at speaker-independent speech recognition. The phonemic value of a short-duration speech signal of up to 50 ms is represented by a matrix composed of acoustic parameters. Standard phonemic categories (SPC's) are defined by a combination of several simple potential functions in this matrix space. The potential function set, as well as its number, is determined automatically by the proposed method. Processing is primarily by algebraic operation and is formulated according to an analogy to particle dynamics. The method is applied to voiceless and voiced stop consonant sets spoken by twelve speakers. The relationship between the classification rate and the number of SPC's is investigated under several initial conditions. Stop consonant recognition tests in CV-syllables are made using derived SPC sets irrespective of following vowels. Recognition rates for the utterances of four speakers not included among the twelve speakers used for training were 84 percent for voiceless and 81 percent for voiced stops. View full abstract»

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  • On FIR delta modulation digital filters

    Publication Year: 1981 , Page(s): 1194 - 1201
    Cited by:  Papers (14)
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    The performance of a finite impulse response (FIR) delta modulation digital filter (DMDF) that requires no multiplication operations has been studied in two important aspects. First, an analytical expression for the threshold word length at which the DMDF performance begins to deteriorate rapidly has been obtained explicitly in terms of sampling frequency and input signal level. Second, three approximate formulas for the optimum LDM step size of DMDF in the region where the word length is greater, equal to, or less than the threshold word length, respectively, have been obtained. These analyses have been done assuming a band-limited Gaussian input, and have been verified by computer simulation. A design example of an FIR DMDF is also given. View full abstract»

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  • Maximum entropy and spectral estimation: A review

    Publication Year: 1981 , Page(s): 1176 - 1186
    Cited by:  Papers (40)
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    The method of maximum entropy is reviewed with emphasis on its relationship to entropy rate, Wiener filters, autoregressive processes, extrapolation, the Levinson algorithm, lattice, all-pole and all-pass filters, and stability. View full abstract»

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  • Adaptive prediction for DPCM coding of TV signals

    Publication Year: 1981 , Page(s): 1142 - 1147
    Cited by:  Papers (2)
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    An adaptive minimum mean-squared error prediction algorithm has been derived for DPCM coding of video signals. Its parallel structure guarantees the possibility of high-speed implementation. Simulations show that there is only little gain in terms of signal-to-noise ratio as compared to optimum fixed prediction. However, if the autocorrelation function of the input is not known a priori and there is a mismatch in the prediction coefficients, the gain due to a fixed predictor experiences a severe dropoff, and adaptive prediction proves superior. View full abstract»

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  • Estimating three-dimensional motion parameters of a rigid planar patch

    Publication Year: 1981 , Page(s): 1147 - 1152
    Cited by:  Papers (104)  |  Patents (11)
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    We present a new direct method of estimating the three-dimensional motion parameters of a rigid planar patch from two time-sequential perspective views (image frames). First, a set of eight pure parameters are defined. These parameters can be determined uniquely from the two given image frames by solving a set of linear equations. Then, the actual motion parameters are determined from these pure parameters by a method which requires the solution of a sixth-order polynomial of one variable only, and there exists a certain efficient algorithm for solving a sixth-order polynomial. Aside from a scale factor for the translation parameters, the number of real solutions never exceeds two. In the special case of three-dimensional translation, the motion parameters can be expressed directly as some simple functions of the eight pure parameters. Thus, only a few arithmetic operations are needed. View full abstract»

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  • Observations on the extrapolation of a band-limited signal problem

    Publication Year: 1981 , Page(s): 1208 - 1209
    Cited by:  Papers (5)
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    In this correspondence, the problem of extrapolating a time-truncated band-limited signal shall be considered with two objectives in mind: 1) to respond to the Sabri and Steenaart [1] correspondence relative to the author's extrapolation method [2], and 2) to comment on the numerical difficulties which individuals have experienced when using various extrapolation procedures. In the discussion that follows, the terminology used in [2] shall be adopted. View full abstract»

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  • Intermodulation in terms of the harmonic output of a nonlinearity

    Publication Year: 1981 , Page(s): 1202 - 1205
    Cited by:  Papers (3)
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    Measurement of pth-harmonic output of a memoryless nonlinearity with a single sinusoid as input provides enough information to compute all intermodulation products falling in the same harmonic zone. For the case of two input sinusoids of arbitrary amplitudes, an integral expression (representing a weighted average of the measured characteristic) is derived for the amplitude of each output signal and any intermodulation product. Simpler expressions are obtained for the case of equal input amplitudes and for the case where one input amplitude is small. The results are applicable to bandpass nonlinearities that produce AM-to-PM as well as AM-to-AM conversion. They are applied to the cases of hard and soft limiters and to polynomial nonlinearities. View full abstract»

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  • Sufficient excitation and stable reduced-order adaptive IIR filtering

    Publication Year: 1981 , Page(s): 1212 - 1215
    Cited by:  Papers (9)
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    A perturbed error system is used to describe the parameter and output error behaviors of reduced-order application of adaptive identifier/filters such as the hyperstable adaptive recursive filter (HARF). Given satisfaction of a sufficient excitation condition, this error system is shown to exhibit a bounded-input, bounded-state (BIBS) property. This implies that, despite order insufficiency, the output and parameter estimates of HARF (and similar adaptive identifier/filter algorithms) remain bounded. View full abstract»

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  • A unilateral representation for autoregressive random field models

    Publication Year: 1981 , Page(s): 1227 - 1228
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    This paper discusses autoregressive random field (ARF) models and derives a unilateral representation driven by uncorrelated noise. The representation is in terms of a stochastic differential equation involving the spectral representation of the model. View full abstract»

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  • MFIR filters: Properties and applications

    Publication Year: 1981 , Page(s): 1128 - 1136
    Cited by:  Papers (22)  |  Patents (1)
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    The class of multiplicative FIR (MFIR) filters is considered and some properties and applications of MFIR filters are investigated. MFIR filter approximation of a given IIR filter is shown to reduce the number of multiplications and additions logarithmically, in comparison to the corresponding FIR filter in direct form. The pure multiplicity property is introduced and is shown to apply to a class of MFIR filters. This property results in a criterion for optimal ordering and expressions for roundoff noise when no scaling is used and also results in the invariance of roundoff noise output under l2-scaling. Linear phase MFIR filter realization of a desired low-pass frequency response magnitude |H_{d}| with centered transition band is shown to require 0(n \log _{2} N) each multipliers and adders. N is the order of the min-max FIR filter design of |H_{d}| and n is the order of the elliptic IIR filter design of |H_{d}|^{1/2} . Several design examples of linear phase low-pass filters are used to compare MFIR filter designs versus those of min-max FIR filters in direct form. Comb filters of order N are shown to have an exact MFIR realization that requires fewer than 2 \log _{2} N additions. Suggestions for further research and applications conclude the paper. View full abstract»

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  • Iterative techniques for minimum phase signal reconstruction from phase or magnitude

    Publication Year: 1981 , Page(s): 1187 - 1193
    Cited by:  Papers (33)  |  Patents (22)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (832 KB)  

    In this paper, we develop iterative algorithms for reconstructing a minimum phase sequence from the phase or magnitude of its Fourier transform. These iterative solutions involve repeatedly imposing a causality constraint in the time domain and incorporating the known phase or magnitude function in the frequency domain. This approach is the basis of a new means of computing the Hilbert transform of the log-magnitude or phase of the Fourier transform of a minimum phase sequence which does not require phase unwrapping. Finally, we discuss the potential use of this iterative computation in determining samples of the unwrapped phase of a mixed phase sequence. 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