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Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 3 • Date June 1983

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Displaying Results 1 - 25 of 48
  • [Front cover and table of contents]

    Publication Year: 1983 , Page(s): 0
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    Freely Available from IEEE
  • Rational modeling by pencil-of-functions method

    Publication Year: 1983 , Page(s): 564 - 573
    Cited by:  Papers (25)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (820 KB)  

    Pole-zero modeling of signals, such as an electromagnetic-scatterer response, is considered in this paper. It is shown by use of the pencil-of-functions theorem that a) the true parameters can be recovered in the ideal case [where the signal is the impulse reponse of a rational function H(z)], and b) the parameters are optimal in the functional dependence sense when the observed data are corrupted by additive noise or by systematic error. Although the computations are more involved than in all-pole modeling, they are considerably less than those required in iterative schemes of pole-zero modeling. The advantages of the method are demonstrated by a simulation example and through application to the electromagnetic response of a scatterer. The paper also includes very recent and promising results on a new approach to noise correction. In contradistinction with spectral subtraction techniques, where only amplitude information is emphasized (and phase is ignored), we propose a method that a) estimates the noise spectral density for the data frame, and then b) performs the subtraction of the noise correlation matrix from the Gram matrix of the signal. View full abstract»

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  • Comments on "The reconstruction of a multidimensional sequence from the phase or magnitude of its Fourier transform"

    Publication Year: 1983 , Page(s): 738 - 739
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (259 KB)  

    When one imposes a nonnegativity constraint, one usually can reconstruct a two-dimensional sequence of finite support from the modulus of its Fourier transform using an iterative algorithm, even when file initial estimate is an array of random numbers. View full abstract»

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  • [Back cover]

    Publication Year: 1983 , Page(s): c4
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    Freely Available from IEEE
  • Stability of 2-D causal digital filters using the residue theorem

    Publication Year: 1983 , Page(s): 774 - 775
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (248 KB)  

    In this correspondence stability of the 2-D causal digital filter is proven directly in terms of its inverse Z-transform. As a by-product one gets exponential bounds on the rate of convergence to zero of the filters impulse response. Using known transformations, these results may be extended to general 2-D recursive digital filters. View full abstract»

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  • A formant estimating equation useful in a graphically controlled speech synthesizer

    Publication Year: 1983 , Page(s): 736 - 738
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (384 KB)  

    A problem encountered in controlling a digital speech synthesizer using a digitizing surface is mapping the necessary controls into a two-dimensional space. The first two formant frequencies are readily associated with the x and y dimensions of the tablet surface, but higher formants must either be fixed or estimated based on the first two. This correspondence describes a method for estimating the third formant frequency given the first two. In comparison to a fixed estimate, the estimator is shown to reduce the worst case error by a factor of three. View full abstract»

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  • A stochastic approach to optimal linear digital equalizers

    Publication Year: 1983 , Page(s): 775 - 780
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (592 KB)  

    This correspondence presents, from the stochastic point of view, a new method for the optimal design of linear digital equalizers using a quadratic performance index. It is well known that a Kalman filter can be used as an equalizer for a digital communication system. If the Kalman filter is applied, then it is useful that an optimal linear equalizer, which minimizes the mean-square error, can be obtained. However, the most disadvantageous point in the Kalman filter is that a great deal of computation is needed to obtain the estimates. Furthermore, the order of the Kalman filter is always equal to the sample number of the impulse response of a transmission channel. Therefore, a high-order equalizer becomes necessary if the sample number is large. Usually, however, it is not necessary to use such a high-order equalizer. Considering this fact, an optimal equalizer is presented under the condition that the order can be assigned arbitrarily by the designer. The proposed equalizer is optimal in the sense that it minimizes the mean-square error subject to this condition. View full abstract»

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  • Nonstationary spectral modeling of voiced speech

    Publication Year: 1983 , Page(s): 664 - 678
    Cited by:  Papers (34)  |  Patents (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1600 KB)  

    The main purpose of this paper is to present a novel model for voiced speech. The classical model, which is being used in many applications, assumes local stationarity, and consequently imposes a simple and well known line structure to the short-time spectrum of voiced speech. The model derived in this paper allows for local non-stationarities not only in terms of pitch perturbations, but in terms of vocal tract variations as well. The resulting structure of the short-time spectrum becomes more complex, but can still be interpreted in terms of generalized lines. The proposed model supports new forms of spectral prediction, which can be put to advantage in speech coding applications. Experimental results are presented supporting the validity of both the model itself and the prediction relationships. Finally, a new class of speech coders, denoted harmonic coders, based on the presented model, is proposed, and a specific implementation is presented. View full abstract»

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  • An adaptive approach for time delay estimation of band-limited signals

    Publication Year: 1983 , Page(s): 780 - 784
    Cited by:  Papers (9)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (496 KB)  

    This correspondence concerns the use of the Widrow's adaptive least-mean-square (LMS) adaptive filter algorithm [1] for estimating the time delay between two-sensor outputs. Theoretical bias and estimation error are discussed. This analytical study provides a clue for determining the most effective way of processing the received signals through the adaptive filter. Computer simulation results are also included. View full abstract»

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  • Approximate expressions for the output signal-to-noise ratios for two types of cross correlators with correlated bandpass inputs

    Publication Year: 1983 , Page(s): 740 - 742
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (360 KB)  

    Thus far, analytical expressions have not been developed for the output signal-to-noise ratio of a polarity coincidence correlator when the noise is correlated between the input channels. Approximate expressions are developed here for the output signal-to-noise ratios for two types of cross-correlator systems- one being the polarity coincidence correlator with a hard limiter in each input channel and the other variation of this correlator having a hard limiter in only one input channel. The approximation scheme is based upon the known output signal-to-noise characteristics of a limiter and that of the standard analog correlator with no limiters. The derived expressions are applicable for the case of deterministic sine waves plus narrow-band Gaussian noise in each input channel, where the noise between channels may be correlated For the case of uncorrelated noise between input channels, good agreement is found between the approximate formulas and known exact results. Finally, asymptotic expressions are developed for certain limiting cases of input signal-to-noise ratios having extreme values in one or both input channels. View full abstract»

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  • Deconvolution of nonstationary seismic data using adaptive lattice filters

    Publication Year: 1983 , Page(s): 591 - 598
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1416 KB)  

    This paper examines the results of the application of two lattice algorithm to the problem of adaptive deconvolution on non-stationary seismic data. A comparative study of the deconvolution performance of the recently proposed gradient lattice and least-squares lattice algorithms is made with the help of experiments on simulated and real seismic data. We show that the gradient lattice algorithm is computationally superior, but it suffers from a possible slow rate of convergence, while the least-squares lattice has better convergence properties and is more robust numerically. We also show that both algorithms can yield equally good deconvolution results with a moderate amount of computation. Finally we indicate that a modified deconvolved output, derived as a linear combination of the forward and backward residuals, improves the performance without involving any additional computational burden. View full abstract»

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  • A statistical approach to the design of an adaptive self-normalizing silence detector

    Publication Year: 1983 , Page(s): 678 - 684
    Cited by:  Papers (7)  |  Patents (13)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1040 KB)  

    Silence detection is an important aspect of automatic speech recognition. In this paper a silence detector is described which classifies each frame of an utterance as being silent or nonsilent. No attempt is made to determine the endpoints of words. The silence detector is based upon a statistical test and has the properties that it is adaptive, self-normalizing, speaker independent, and script independent. No training sets are required, although it is assumed that the first part of a signal is silent. An experiment is described in which the silence detector was successfully applied to more than 2 h of isolated-word speech using a frame length of 1 cs. View full abstract»

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  • Iterative time-limited signal restoration

    Publication Year: 1983 , Page(s): 643 - 650
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (800 KB)  

    The purpose of this paper is to show that time-limited restoration of shift-invariant blurred signals can be done by means of fixed-point solutions of contraction mappings, under rather general conditions for the distortion operator. All our results are valid for multidimensional signals. An application of these results to the iterative extrapolation of band-limited discrete images is shown. View full abstract»

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  • Electroacoustic analysis and enhancement of alaryngeal speech

    Publication Year: 1983 , Page(s): 785
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    First Page of the Article
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  • A particular example of phase unwrapping using noisy experimental data

    Publication Year: 1983 , Page(s): 742 - 744
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (352 KB)  

    Obtaining a smooth continuous function of phase from a distorted, sampled set of phase measurements is addressed. Provided the smooth continuous function can be written as a power series, all the important coefficients can be derived from phase difference measurments which have proved to be less affected by phase noise than the phase measured directly. This simple technique is shown to have advantages in some experimental configurations. View full abstract»

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  • A note on stability and lattice filter relations

    Publication Year: 1983 , Page(s): 770 - 772
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (328 KB)  

    In this note we draw attention to an earlier result by Marden [2] for establishing the stability of linear discrete-time systems. The stability algorithm then provides a simple constructive realization of stable filters in lattice form. The class of unstable filters which cannot be represented in lattice form is also partially characterized. View full abstract»

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  • Minimum-variance and maximum-likelihood recursive waveshaping

    Publication Year: 1983 , Page(s): 599 - 604
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (608 KB)  

    In this paper, we develop optimal recursive waveshaping filters in the framework of estimation theory and state-variable models. We develop a linear minimum-variance waveshaper and a nonlinear maximum-likelihood waveshaper. Both waveshapers consist of two components: 1) stochastic inversion and 2) waveshaping. The former is performed by means of minimum-variance deconvolution. Simulations are given which illustrate results that can be obtained by both waveshapers. In retrospect, we view the minimum-variance results of this paper as the recursive counterparts to those presented by Treitel and Robinson [14], which are for finite-impulse response waveshaping. View full abstract»

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  • Design of circularly symmetric two-dimensional FIR digital filters employing transformations with variable parameters

    Publication Year: 1983 , Page(s): 637 - 642
    Cited by:  Papers (24)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (632 KB)  

    A modification of the trasformation method used for the design of two-dimensional (2-D) circularly symmetric finite impulse response (FIR) digital filters from one-dimensional (1-D) filters is described. This modification entails the embedding of variable parameters in the different transformations applied to the different factors of the 1-D reference function. This new method results in the design of 2-D FIR filters whose frequency response characteristics meet the cutoff boundary specifications more closely than the transformations without the modification. This method is quite useful for the design of 2-D FIR filters with multiple cutoff boundaries such as bandpass filters. View full abstract»

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  • Digital integrators using optimal FIR compensators

    Publication Year: 1983 , Page(s): 726 - 729
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (488 KB)  

    An important class of digital integration algorithms is examined. The complete integration filter is partitioned into a feedback filter combined with a finite duration impulse response (FIR) part which may be thought of as a cascaded compensator subfilter. New integrator designs are obtained using linear programming to achieve minimax optimization of the linear-phase FIR compensator, constrained to have no dc error. Comparisons with other integration algorithms are presented. A tabulation of some optimal integrator filter coefficients is included. View full abstract»

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  • Observers for a class of 2-D filters

    Publication Year: 1983 , Page(s): 557 - 563
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (680 KB)  

    In this paper, a 2-D observer is proposed for 2-D systems of the Fornasini-Marchesini type. Conditions are given for the existence of the proposed observer where both the case of known and unknown boundary conditions are considered. Design equations are developed for calculation of the observer matrices. Conditions are also given for being able to choose the form of the observer in the simple form for 2-D systems proposed by Attasi. Finally, it is shown how the proposed observer can be used in a feedback scheme to stabilize a 2-D filter having a state-space model of the Attasi type. View full abstract»

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  • Representation of glottal shape data for signal processing

    Publication Year: 1983 , Page(s): 766 - 769
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (720 KB)  

    A new representation of glottal-edge data derivable from high-speed laryngeal films is given. The resulting simple curves represent the glottal shape during phonation and are so arranged that various one-and two-dimensional discrete Fourier transforms can be used to test glottal symmetry and vocal fold vibration rates and phases. Possible discriminant features are suggested and illustrated. View full abstract»

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  • An approximation theoretic approach to maximum entropy spectral analysis

    Publication Year: 1983 , Page(s): 734 - 736
    Cited by:  Papers (2)
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    It is shown that the theory of best approximation in weighted mean square can be used to provide a mathematical foundation for the analysis and extension of the maximum entropy method of spectral analysis. View full abstract»

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  • Comparison of the characteristics of linear least squares and orthonormal expansion in estimation

    Publication Year: 1983 , Page(s): 755 - 759
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    It is frequently assumed in signal processing applications that expansion of a sequence by a weighted sum of mutually orthonormal sequences yields weighting coefficients that are identical to the estimates of the parameters which maximize the likelihood function if the linear sum of sequences is chosen as a model. Although this may be a valid approximation when signal-to-noise ratios are large, it is not generally the case and may lead to erroneous results when substantial noise exists. This paper explores the relationship between orthonormal expansion and linear least squares estimation. In doing so, the conditions under which orthonormal expansion coefficients are maximum likelihood estimates are identified. Several interesting properties related to both techniques are also revealed. The results are relevant to a wide range of signal processing applications such as the discrete Fourier transform and linear prediction theory and can be extended to non-linear least squares estimation. This should make the results of interest to those involved with the analysis of noisy data. View full abstract»

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  • Critical band analysis-synthesis

    Publication Year: 1983 , Page(s): 656 - 663
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (736 KB)  

    The formal derivation of a transformation which models the frequency selective properties (critical bandwidths) of the auditory system is developed. A parameterized family of constant-Q analysis-synthesis transform pairs is developed from a property of homogeneous functions. This formulation allows for a wide choice of selections for center frequencies, bandwidths, and filter shapes. A particular member of the transform family is implemented to model the frequency selective properties of the peripheral auditory system. With this transform, short-time spectral analysis using critical band filter shapes can be implemented. In the absence of spectral modification, the analysis-synthesis transform can be made arbitrarily close to an identity system. This new approach to analysis-synthesis provides the necessary mathematical support needed to design and optimize both constant-Q and critical band analysis-synthesis transforms. View full abstract»

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  • On using symmetry properties for selecting transform components for image coding

    Publication Year: 1983 , Page(s): 749 - 752
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (408 KB)  

    A method based on the symmetry property of the transform operators to compute the transform domain covariances is discussed. A selection scheme which uses the transform domain covariance and also symmetry properties of a given transform is proposed. Some experimental results are also presented. 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