Scheduled System Maintenance:
Some services will be unavailable Sunday, March 29th through Monday, March 30th. We apologize for the inconvenience.
By Topic

Acoustics, Speech and Signal Processing, IEEE Transactions on

Issue 4 • Date April 1987

Filter Results

Displaying Results 1 - 25 of 32
  • [Front cover and table of contents]

    Publication Year: 1987 , Page(s): 0
    Save to Project icon | Request Permissions | PDF file iconPDF (201 KB)  
    Freely Available from IEEE
  • [Back cover]

    Publication Year: 1987 , Page(s): c4
    Save to Project icon | Request Permissions | PDF file iconPDF (1283 KB)  
    Freely Available from IEEE
  • On the rate of growth of condition numbers for convolution matrices

    Publication Year: 1987 , Page(s): 471 - 475
    Cited by:  Papers (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (496 KB)  

    When analyzing linear systems of equations, the most important indicator of potential instability is the condition number of the matrix. For a convolution matrix W formed from a series w (where W_{ij} = w_{i-j+1}, 1 \leq i-j + 1 \leq k, W_{ij} = 0 otherwise), this condition number defines the stability of the deconvolution process. For the larger convolution matrices commonly encountered in practice, direct computation of the condition number (e.g., by singular value decomposition) would be extremely time consuming. However, for convolution matrices, an upper bound for the condition number is defined by the ratio of the maximum to the minimum values of the amplitude spectrum of w. This bound is infinite for any series w with a zero value in its amplitude spectrum; although for certain such series, the actual condition number for W may in fact be relatively small. In this paper we give a new simple derivation of the upper bound and present a means of defining the rate of growth of the condition number of W for a band-limited series by means of the higher order derivatives of the amplitude spectrum of w at its zeros. The rate of growth is shown to be proportional to mp, where m is the column dimension of W and p is the order of the zero of the amplitude spectrum. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A new technique for adaptive frequency estimation and tracking

    Publication Year: 1987 , Page(s): 561 - 564
    Cited by:  Papers (6)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (440 KB)  

    A new technique is presented for estimating the frequency of a sinusoid in noise. Standard techniques typically estimate the frequency of a sinusoid from an estimate of the autocorrelation function or from a filter model. These techniques require a large number of samples for an accurate estimate of the frequency. Furthermore, if the frequency is varying with time, recomputation of the autocorrelation or filter model is necessary for each estimate update. In this correspondence, we present a new technique that is based on a variable delay element. We will show that the corresponding error surface is sinusoidal, and that the first maximum of the error function occurs at one-half the period of the unknown sinusoid. We will develop an algorithm that is based on gradient techniques to find this maximum, and from this maximum we directly compute the frequency estimate. The new algorithm works in the time domain with a simple adaptive delay update computation. The technique has been tested in simulations with signal-to-noise ratios as low as 1 : 10, with excellent performance. The speed of the algorithm depends on a convergence factor that is computed from an initial estimate of the power of the input signal. Since the technique is adaptive, it can also be applied to tracking a time-varying frequency. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Theory and design of M-channel maximally decimated quadrature mirror filters with arbitrary M, having the perfect-reconstruction property

    Publication Year: 1987 , Page(s): 476 - 492
    Cited by:  Papers (179)  |  Patents (7)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1624 KB)  

    Based on the concept of losslessness in digital filter structures, this paper derives a general class of maximally decimated M-channel quadrature mirror filter banks that lead to perfect reconstruction. The perfect-reconstruction property guarantees that the reconstructed signal \hat{x} (n) is a delayed version of the input signal x (n), i.e., \hat{x} (n) = x (n - n_{0}) . It is shown that such a property can be satisfied if the alias component matrix (AC matrix for short) is unitary on the unit circle of the z plane. The number of channels M is arbitrary, and when M is two, the results reduce to certain recently reported 2-channel perfect-reconstruction QMF structures. A procedure, based on recently reported FIR cascaded-lattice structures, is presented for optimal design of such FIR M-channel filter banks. Design examples are included. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Confidence intervals for the maximum entropy spectrum

    Publication Year: 1987 , Page(s): 504 - 510
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (680 KB)  

    We present a computationally efficient method of calculating confidence intervals for the maximum entropy (ME) or autoregressive (AR) spectrum, based on new theoretical considerations. The method is consistent with the well-known asymptotic results, and it can be incorporated under certain assumptions into the Levinson-Durbin recursion. The method is also extended to ME wavenumber estimation. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Realization of power spectra from partial covariance sequences

    Publication Year: 1987 , Page(s): 438 - 449
    Cited by:  Papers (51)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1120 KB)  

    In this paper, we address the problem of realization of a spectral density function from incomplete information about the underlying stochastic process. The standing assumption is the availability of an (incomplete) partial sequence of covariance samples of the process. We study the set of rational extensions of this finite sequence to an infinite covariance function that agrees with the available samples. The classical theory of orthogonal polynomials (with respect to the unit circle) and the theory of moments have been utilized extensively in a variety of engineering problems, including the one we are dealing with. These have been known to provide a unifying framework for a variety of current spectral estimation techniques (maximum entropy method, Pisarenko's harmonic decomposition, etc.). In this work, we consider and study the set of all covariance realizations of dimension lower than or equal to the length of the partial sequence (and equal to the dimension of the maximum entropy realization). The ME solution is a point in this set. Other points correspond to "pole-zero" models. A general formula is obtained for recursively updated "pole-zero" models of dimension increasing with the data record. Information about the "zeros" is obtained from the asymptotic behavior of the "partial autocorrelation coefficients." Our approach combines techniques from the theory of orthogonal polynomials on the unit circle, the theory of moments, and also techniques from degree theory/topology. Our objective is to develop a theory that will provide a frame for constructing recursively "pole-zero" realizations of increasing dimension. View full abstract»

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

    Publication Year: 1987 , Page(s): 520 - 526
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (832 KB)  

    A direct, noniterative approach to retrieving a multidimensional complex image (i.e., its phase can vary from pixel to pixel) from the magnitude of its Fourier transform is developed. The uniqueness of the reconstruction is shown to be a direct consequence of the existence of zero surfaces or sheets in the multidimensional z transforms of the image. The analytic properties of these zero sheets enable one to distinguish the zero sheet belonging to the image from that belonging to the complex conjugate of its reflection in the coordinate origin. It is thereby possible to recover the unique, most compact "image form" of the original image. Two-dimensional examples are presented. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A new realization fo 2-D digital filters

    Publication Year: 1987 , Page(s): 533 - 542
    Cited by:  Papers (15)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (800 KB)  

    A systolic array realization for 2-D digital filters is presented. It is based on Chan's local state-space model. This structure has the minimum number of states (minimum delays). A technique is also presented to reduce the number of multipliers to its minimum. Conditions that the system eigenvalues have to satisfy to ensure low sensitivity with respect to the variations in the multipliers are developed. Several examples are included to illustrate the proposed techniques. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • On a fundamental property of the Wigner distribution

    Publication Year: 1987 , Page(s): 559 - 561
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (384 KB)  

    We discuss and illustrate the following fundamental property of the Wigner distribution: the Wigner distribution is not necessarily zero when the signal is zero. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Experimentation with synthesized speech generated from line-spectrum pairs

    Publication Year: 1987 , Page(s): 568 - 571
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (496 KB)  

    Line-spectrum pairs (LSP's) are frequency-domain parameters similar to formant frequencies. Thus, they have frequency-selective spectral-error characteristics which allow LSP quantization in accordance with auditory perception. In addition, ease of estimating the spectral-error sensitivity of each line spectrum makes possible encoding each line spectrum efficiently. This correspondence, for the first time, demonstrates that a 31 bit representation of LSP's provides similar intelligibility as a 41 bit representation of reflection coefficients in a current 2400 bit/s LPC. Even with a 12 bit quantization of LSP's, the loss of speech intelligibility is minor, only 2.4 points below that of a 41 bit quantization of reflection coefficients as measured by the diagnostic rhyme test (DRT) which tests initial-consonant discrimination. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • On optimal image digitization

    Publication Year: 1987 , Page(s): 553 - 555
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (360 KB)  

    Nielsen et al. recently addressed the problem of determining the optimal discretization grid and quantization depth when a given bivariate function f(x, y) has to be described with a predetermined number of bits. This was done under the assumption that the function value range and mean "fluctuation rates" in the x and y direction are given, and that ideal point sampling with zero-order-hold interpolation is used in reconstructing the image. This correspondence outlines an alternative approach, based on the assumption that f (x, y) is the sample function of a 2-D stationary stochastic process with a known covariance function. We use standard integral sampling and obtain closed form solutions under the assumption that f(x, y) is (the sample of) a homogeneous and separable Markov process. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • On the approximate factorization of 2-D polynomials

    Publication Year: 1987 , Page(s): 577 - 579
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (336 KB)  

    A method is presented for the approximate factorization of unfactorable bivariate polynomials. An efficient iterative algorithm is developed to minimize an appropriate criterion. The basic ideas can be easily extended to the case of polynomials in more than two independent variables. An example is presented to illustrate the utility of the method. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Resolution of signal wavefronts by eigenvector rotation

    Publication Year: 1987 , Page(s): 564 - 566
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB)  

    It is demonstrated that signal wavefronts may be resolved by eigenvector rotation. This technique relies on the source eigenvectors rather than the noise eigenvectors. It is shown that the technique is easy to perform numerically and that the resolution is superior to that obtained from the MUSIC algorithm. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A Bayesian-adaptive decision method for the V/UV/S classification of segments of a speech signal

    Publication Year: 1987 , Page(s): 556 - 559
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (424 KB)  

    In this correspondence, a method for voiced (V), unvoiced (UV), or silence (S) classification of speech segments, based on the maximum a posteriori probability criterion, is presented. The a posteriori probabilities of the three classes are determined using a vector x = ( f1,... , fL) of measurements on the segment under consideration. It is assumed that the vector x has an L-dimensional Gaussian distribution with an expected random value also characterized by an L-dimensional Gaussian distribution. In addition, it is assumed that the sequence of the classes constitutes a first-order stationary Markov chain. The initial parameters are estimated in a training phase. During the application phase, the decision method is adapted by using the previous classifications in order to update the probability density function (pdf) of the expected random values. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Improved convergence analysis of stochastic gradient adaptive filters using the sign algorithm

    Publication Year: 1987 , Page(s): 450 - 454
    Cited by:  Papers (81)  |  Patents (2)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (448 KB)  

    Convergence analysis of stochastic gradient adaptive filters using the sign algorithm is presented in this paper. The methods of analysis currently available in literature assume that the input signals to the filter are white. This restriction is removed for Gaussian signals in our analysis. Expressions for the second moment of the coefficient vector and the steady-state error power are also derived. Simulation results are presented, and the theoretical and empirical curves show a very good match. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Convergence improvement in the adaptive implementation of Pisarenko's method

    Publication Year: 1987 , Page(s): 574 - 577
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (464 KB)  

    The problems of the initialization and of the choice of the adjusting step in the adaptive implementation of Pisarenko's method have been overcome by two techniques recently proposed in the technical literature. In the present correspondence, the relation between them is investigated in detail. Moreover, some suggestions are given for improving their convergence and for reducing their computational cost. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A framework for beamforming structures

    Publication Year: 1987 , Page(s): 584 - 586
    Cited by:  Papers (1)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (384 KB)  

    A framework for describing linearly constrained minimum variance (LCMV) beamforming structures is presented. The framework results from factoring the processor impulse response into a weight vector and a matrix which describes the structure. Classes of structures with equivalent steady-state and adaptive performance are discussed. Examples are given which illustrate application of this framework to several different structures. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • The selection and application of an IIR adaptive filter for use in active sound attenuation

    Publication Year: 1987 , Page(s): 433 - 437
    Cited by:  Papers (71)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (584 KB)  

    The use of infinite impulse response (IIR) adaptive filters has lagged behind that of finite impulse response (FIR) adaptive filters. This has been due, in part, to the increased complexity of IIR filters and the potential for instability that exists due to the presence of poles in the transfer function. This paper discusses the use of adaptive filters for the active cancellation of acoustic noise. It is shown that IIR filters possess certain characteristics that are highly desirable for this problem. The selection of an appropriate IIR adaptive algorithm is discussed using observability considerations. It is shown that the recursive least mean square (RLMS) algorithm of Feintuch possesses significant advantages for use in a practical active attentuation system. Results are presented from computer simulations as well as an actual system using a TI TMS32010 digital signal processing microprocessor. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Hardware for image rotation by twice skew transformations

    Publication Year: 1987 , Page(s): 527 - 532
    Cited by:  Papers (8)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (640 KB)  

    Image processing systems in real time are going to be used in various fields, in both factories and offices. However, these performances are not sufficient when a very short processing time is required. This paper presents quite a new method of high-speed image rotation. Here an image is rotated only by twice skew transformations, whereby each pixel is shifted in parallel with each coordinate axis. With this algorithm, an image can be rotated very quickly, since such a parallel shifting is performed by simple address controlling, i.e., brief additions of address values. The processing time to rotate a gray scaled image made up of 256 × 256 pixels is about 5 s, even by an 8 bit microcomputer. Furthermore, we propose a hardware for image rotation based on this algorithm. In the case of the hardware, the processing time to rotate an image made up of 516 × 516 pixels is estimated to be about 10 ms, when a high-speed 16 bit microcomputer is used for data setting. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Closed-loop least mean square time-delay estimator

    Publication Year: 1987 , Page(s): 413 - 424
    Cited by:  Papers (8)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1080 KB)  

    An LMS closed-loop time-delay estimator is presented. It uses the error between two samples of the incoming signal (the difference between the delayed signal and the reference signal passed through a known delay) as a performance index for the estimator. The LMS algorithm adaptively controls the delay so as to minimize the mean square of this error. The controlled delay is implemented using surface acoustic wave devices. Certain design conditions are applied, resulting in a unique minimum for the performance surface. It is shown that the proposed estimator is unbiased and has a small variance if the input signal occupies most of the system bandwidth. In fact, the variance depends on the input noise power and the generalized noise-to-signal power ratio, R , as well as on the loop gain. The analysis performed also gives a bound on the loop gain required for convergence of the estimator and predicts its rate of convergence. Computer simulation results show good agreement with the theory. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Extensions of the stability criterion for ARMA filters

    Publication Year: 1987 , Page(s): 425 - 432
    Cited by:  Papers (12)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (856 KB)  

    Using the lattice representation of an ARMA filter, it is well known that the necessary and sufficient condition so that the poles are inside the unit circle is |k_{i} < 1, 1 \leq i \leq n where the ki's are the reflection coefficients. The filter is said to be wide sense stable if no pole is located outside the unit circle, and it is interesting to characterize this stability by an appropriate necessary and sufficient condition. To establish this condition, the concept of canonical reflection coefficient is introduced, which eliminates the problems appearing when the Levinson recursion is not inversible. Some examples are discussed and a simple and practical test for wide sense stability is given. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • On the probability density of a maximum likelihood mean frequency estimator

    Publication Year: 1987 , Page(s): 579 - 580
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (240 KB)  

    A simplified derivation of the probability density function of a mean frequency estimator that equates the mean to the location of the peak of the discrete power spectrum is presented. Although the model in which the signal straddles at most two spectral coefficients is emphasized, the general case of arbitrary signal leakage is also worked out in detail. Previous work on this topic implies that n! (n - 1)-tuple integrals must be evaluated for an n-sample DFT estimator. Our approach demonstrates that the derivation can proceed in a much more straightforward manner. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Modular architectures for adaptive multichannel lattice algorithms

    Publication Year: 1987 , Page(s): 543 - 552
    Cited by:  Papers (26)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1024 KB)  

    A modular architecture for adaptive multichannel lattice algorithms is presented. This architecture requires no matrix computations and has a regular structure, which significantly simplifies its implementation as compared to the multichannel (matrix) version of the same algorithms. Because the suggested architecture exhibits a high degree of parallelism and local communication, it is well suited for implementation in dedicated (VLSI) hardware. The derivation of this modular architecture demonstrates a powerful principle for modular decomposition of multichannel recursions into systolic-arraylike architectures. The scope of applicability of this principle extends beyond multichannel lattice (and related least-squares) algorithms to other algorithms involving matrix computations, such as multiplication; factorization, and inversion. View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Streaking in median filtered images

    Publication Year: 1987 , Page(s): 493 - 503
    Cited by:  Papers (38)
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1512 KB)  

    This paper presents a probabilistic analysis of the streaking or blotching effect commonly observed in median filtered signals in both one and two dimensions. The effcts are identified as runs of equal or nearly equal values which create visual impressions that have no visual correlate. For one-dimensional discrete iid random signals with continuous input probability densities, the probability of a streak of length L occurring is computed and shown to be independent of the input probability distribution. Expressions for the first and second moments of the streak length are also derived, and certain asymptotic results are given. As the analysis and definition of the analogous effect in two dimensions is less tractable, the probability that medians taken from distinct overlapping windows will take the same value is derived for various filter geometries. The analytic results are supported by examples using both one- and two-dimensional signals. 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