By Topic

Signal Processing, IEEE Transactions on

Issue 7 • Date Jul 1992

Filter Results

Displaying Results 1 - 24 of 24
  • Adaptive multichannel lattice-escalator filter structure: an application to generalized sidelobe canceler

    Page(s): 1816 - 1819
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (280 KB)  

    An adaptive digital filter structure which can be used in multichannel noise-canceling applications is described. The proposed structure is obtained by creating a hybrid from the lattice and the escalator (Gram-Schmidt) structures, with the coefficients being updated using the least mean square (LMS) algorithm. As an application, the proposed adaptive multichannel digital filter is applied to implement an adaptive generalized sidelobe canceler View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • DOA identifiability for rotationally invariant arrays

    Page(s): 1825 - 1828
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (340 KB)  

    The problem of DOA (direction-of-arrival) estimation using sensor arrays composed of two identical uncalibrated rotated subarrays is considered. It is shown that such arrays do not provide an identifiable parameterization of the problem; i.e. unlike ESPRIT, unique estimates are not possible when more than one signal is present View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • An interference-tolerant algorithm for localization of cyclostationary-signal sources

    Page(s): 1682 - 1686
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (468 KB)  

    A method for detecting the number of cyclostationary signals radiated by remote sources and for estimating their directions of arrival by a linear and uniform array is presented. Whereas the traditional techniques exploit the spatial coherent properties, the new method locates the signal sources using the spectral coherence properties as well. This approach eliminates the need to know the characteristics of the noise and the interference, regardless of the extent of their spectral overlap. Moreover, the method applied equally well to environments containing more interferers than sensors. The conditions of applicability of the method are the existence and the knowledge of a cycle frequency at which all the signal sources exhibit spectral correlation but the noise and interference signals do not, and the existence and the knowledge of a value of the lag parameter such that the cyclic cross-correlation matrix of the desired signals has full rank View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Azimuth and elevation computation in high resolution DOA estimation

    Page(s): 1828 - 1832
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (412 KB)  

    A number of high-resolution direction finding methods for determining the two-dimensional DOA (directions of arrival) of a number of plane waves impinging on a sensor array are discussed. The array consists of triplets of sensors that are identical, as an extension of the one-dimensional ESPRIT scenario to two dimensions. Algorithms that yield the correct parameter pairs while avoiding an extensive search over the two separate one-dimensional parameter sets are devised View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • The statistical performance of some instantaneous frequency estimators

    Page(s): 1708 - 1723
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1068 KB)  

    The authors examine the class of smoothed central finite difference (SCFD) instantaneous frequency (IF) estimators which are based on finite differencing of the phase of the analytic signal. These estimators are closely related to IF estimation via the (periodic) first moment, with respect to frequency of discrete time-frequency representations (TFRs) in L. Cohen's (1966) class. The authors determine the distribution of this class of estimators and establish a framework which allows the comparison of several other estimators such as the zero-crossing estimator and one based on linear regression on the signal phase. It is found that the regression IF estimator is biased and exhibits a large threshold for much of the frequency range. By replacing the linear convolution operation in the regression estimator with the appropriate convolution operation for circular data the authors obtain the parabolic SCFD (PSCFD) estimator, which is unbiased and has a frequency-independent variance, yet retains the optimal performance and simplicity of the original estimator View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A performance analysis of subspace-based methods in the presence of model errors. I. The MUSIC algorithm

    Page(s): 1758 - 1774
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1376 KB)  

    Application of subspace-based algorithms to narrowband direction-of-arrival (DOA) estimation requires that both the array response in all directions of interest and the spatial covariance of the noise must be known. In practice, however, neither of these quantities is known precisely. Depending on the degree to which they deviate from their nominal values, serious performance degradation can result. The performance of the MUSIC algorithm is examined for situations in which the noise covariance and array response are perturbed from their assumed values. Theoretical expressions for the error in the MUSIC DOA estimates are derived and compared with simulations performed for several representative cases, and with the appropriate Cramer-Rao bound. An optimally weighted version of MUSIC is proposed for a particular class of array errors View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Operator approach to performance analysis of root-MUSIC and root-min-norm

    Page(s): 1687 - 1696
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (704 KB)  

    The authors carry out a performance analysis of two eigenstructure-based direction-of-arrival estimation algorithms, using a series expansion of projection operators (or projectors) on the signal and noise subspaces. In the interest of algebraic simplicity, an operator formalism is utilized. A perturbation analysis is performed on the projectors, the results of which are used to determine the effect on the estimated parameters. The approach makes it possible to carry out the analysis to any chosen order of expansion of the projectors by using an original recurrence formula developed for the higher-order terms in the series expansion of the projectors. This method is used to study the root-MUSIC and root-min-norm algorithms and establish the superiority of root-MUSIC in all cases. The analysis has also resulted in insightful asymptotic expressions that describe the statistical behavior of the estimated angles and radii of the signal zeros View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Using iterated function systems to model discrete sequences

    Page(s): 1724 - 1734
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (948 KB)  

    Two iterated function system (IFS) models are explored for the representation of single-valued discrete-time sequences: the self-affine fractal model and the piecewise self-affine fractal model. Algorithms are presented, one of which is suitable for a multiprocessor implementation, for identification of the parameters of each model. Applications of these models to a variety of data types are given where signal-to-noise ratios are presented, quantization effects of the model parameters are investigated, and compression ratios are computed View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A new DSP-oriented algorithm for calculation of the square root using a nonlinear digital filter

    Page(s): 1663 - 1669
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (544 KB)  

    A high-speed algorithm for calculating the square root is proposed. This algorithm, which can be regarded as calculation of the step response of a kind of nonlinear IIR filter, requires no divisions. Therefore, it is suitable for a VLSI digital signal processor (DSP) which has a high-speed hardware multiplier but does not usually have a high-speed hardware divider. The convergence properties of the algorithm are analyzed and used to develop a practical implementation of the procedure. It is implemented on the commercially available DSP (TM320C25) and is compared with the Newton-Raphson method. The proposed algorithm has two advantages over the Newton-Raphson method: higher execution speed and smaller calculation error View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • On the complexity of IQML algorithms

    Page(s): 1811 - 1813
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (232 KB)  

    The authors study the computational complexity of two methods for solving least squares and maximum likelihood modal analysis problems. In particular, they consider the Steiglitz-McBride and iterative quadratic maximum likelihood (IQML) algorithms. J.H. McClellan and D. Lee (ibid., vol.39, no.2, p.509-12, 1991) have shown the iterations of the two methods to be equivalent. However, they suggest that the Steiglitz-McBride algorithm may be computationally preferable. A method for reducing the dimension of the matrix inversion required at each iteration of IQML is provided. The resulting reduction in the computation makes the computational complexity of IQML commensurate with that of the Steiglitz-McBride algorithm View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Modeling and classification of natural sounds by product code hidden Markov models

    Page(s): 1833 - 1835
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (260 KB)  

    Linear predictive coding (LPC), vector quantization (VQ), and hidden Markov models (HMMs) are three popular techniques from speech recognition which are applied in modeling and classifying nonspeech natural sounds. A new structure called the product code HMM uses two independent HMM per class, one for spectral shape and one for gain. Classification decisions are made by scoring shape and gain index sequences from a product code VQ. In a series of classification experiments, the product code structure outperformed the conventional structure, with an accuracy of over 96% for three classes View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Adaptive inverse filters for stereophonic sound reproduction

    Page(s): 1621 - 1632
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (992 KB)  

    A general theoretical basis for the design of adaptive digital filters used for the equalization of the response of multichannel sound reproduction systems is described. The approach is applied to the two-channel case and then extended to deal with arbitrary numbers of channels. The intention is to equalize not only the response of the loudspeakers and the listening room but also the crosstalk transmission from right loudspeaker to left ear and vice versa. The formulation is a generalization of the Atal-Schroeder crosstalk canceler. However, the use of a least-squares approach to the digital filter design and of appropriate modeling delays potentially allows the effective equalization of nonminimum phase components in the transmission path. A stochastic gradient algorithm which facilitates the adaptation of the digital filters to the optimal solution, thereby providing the possibility of designing the filters in situ, is presented. Some experimental results for the two-channel case are given View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Direction-of-arrival estimation via exploitation of cyclostationary-a combination of temporal and spatial processing

    Page(s): 1775 - 1786
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (940 KB)  

    Many modulated communication signals exhibit a cyclostationarity (or periodic correlation) property, corresponding to the underlying periodicity arising from carrier frequencies or band rates. By exploiting cyclostationarity, i.e. evaluating the cyclic correlations of the received data at certain cycle frequencies, one can extract the cyclic correlations of only signals with the same cycle frequency and null out the cyclic correlations of stationary additive noise and all other cochannel interferences with different cycle frequencies. Thus, the signal detection capability can be significantly improved. An approach for exploiting cyclostationarity that is asymptotically exact for either narrowband or broadband sources, unlike previous methods, is proposed. The method also has significant implementational advantages over the earlier techniques. The simulation results indicate a significantly better performance in some environments View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A novel homomorphic processing of ultrasonic echoes for layer thickness measurement

    Page(s): 1819 - 1825
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (488 KB)  

    A homomorphic signal processing method for accurate thickness measurement of thin layers is presented. The method consists in taking the envelope of the inverse Fourier transform of a corrected logarithmic derivative of the energy spectrum (ES) of the measured signal. Simulations show excellent agreement between estimated and actual values. The method is validated by in vitro experiments at 10 MHz on Teflon films, where thicknesses down to 50 μm are recovered, and preliminary in vivo results on arterial wall thickness estimates are presented View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Some properties of the generalized time frequency representation with cone-shaped kernel

    Page(s): 1735 - 1745
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (732 KB)  

    The cone-shaped kernel generalized time-frequency representation (GTFR) of Zhao, Atlas, and Marks (ZAM) has been shown empirically to generate quite good time frequency representation in comparison to other approaches. The authors analyze some specific properties of this GTFR and compare them to other TFRs. Asymptotically, the GTFR is shown to produce results identical to that of the spectrogram for stationary signals. Interference terms normally present in many GTFRs are shown to be attenuated drastically by the use of the ZAM-GTFR. The ability of the ZAM-GTFR to track frequency hopping is shown to be close to that of the Wigner distribution. When a signal is subjected to white noise, the ZAM-GTFR produces an unbiased estimate of the ZAM-GTFR of the signal without noise. In many other GTFRs, the power spectral density of the noise is superimposed on the GTFR of the signal. It is also shown that, in discrete form, the ZAM-GTFR is generally invertible View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A new transform for time-frequency analysis

    Page(s): 1697 - 1707
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (808 KB)  

    The psi-decomposition of a signal, in which the signal is written as a weighted sum of certain elementary synthesizing functions, is described. The set S of synthesizing functions consists of dilated and translated copies of two parent functions, which are concentrated in both the time and the frequency domains. The weighting constants in the psi-decomposition define a transform called the phi-transform. The phi-transform of a signal captures both the frequency content and the temporal evolution of a nonstationary signal. The phi-transform is linear, continuous, and continuously invertible. The set S of synthesizing functions used in the psi-decomposition is nonorthogonal, hence considerable flexibility is permitted in its construction. It is shown with the help of two examples that the set S is easy to construct View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Time-scale energy distributions: a general class extending wavelet transforms

    Page(s): 1746 - 1757
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (968 KB)  

    The theory of a new general class of signal energy representations depending on time and scale is developed. Time-scale analysis has been introduced recently as a powerful tool through linear representations called (continuous) wavelet transforms (WTs), a concept for which an exhaustive bilinear generalization is given. Although time scale is presented as an alternative method to time frequency, strong links relating the two are emphasized, thus combining both descriptions into a unified perspective. The authors provide a full characterization of the new class: the result is expressed as an affine smoothing of the Wigner-Ville distribution, on which interesting properties may be further imposed through proper choices of the smoothing function parameters. Not only do specific choices allow recovery of known definitions, but they also provide, via separable smoothing, a continuous transition from Wigner-Ville to either spectrograms or scalograms (squared modulus of the WT). This property makes time-scale representations a very flexible tool for nonstationary signal analysis View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • An improved inverse filtering method for parametric spectral estimation

    Page(s): 1807 - 1811
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (364 KB)  

    For a wide-sense stationary process x(k), it is well known that its power spectrum Pxx(f) can be estimated by whitening the data with the inverse filter, V (z)=1/H(z), of the assumed minimum-phase rational model H(z) associated with x(k ). However, the initial conditions for computing the output e (k) of the recursive filter V(z) are unknown and must be preassigned. An improved inverse filtering method which simultaneously estimates the coefficients of V(z) as well as the initial conditions is proposed. The resultant power spectral estimator, with the initial conditions being estimated, outperforms that with the initial conditions wrongly set to zero as the time constant of V(z) is comparable to the number of data. Some simulation results which support the superior performance of the former are presented View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A fast quasi-Newton adaptive filtering algorithm

    Page(s): 1652 - 1662
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (884 KB)  

    The convergence rate of an adaptive system is closely related to its ability to track a time-varying optimum. Basic adaptive filtering algorithms give poor convergence performance when the input to the adaptive system is colored. More sophisticated algorithms which converge very rapidly regardless of the input spectrum algorithms typically require O(N2) computation, where N is the order of the adaptive filter, a significant disadvantage for real-time applications. Also, many of these algorithms behave poorly in finite-precision implementation. An adaptive filtering algorithm is introduced which employs a quasi-Newton approach to give rapid convergence even with colored inputs. The algorithm achieves an overall computational requirement of O(N) and appears to be quite robust in finite-precision implementations View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Pipelined recursive filter with minimum order augmentation

    Page(s): 1643 - 1651
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (604 KB)  

    Pipelining is an efficient way for improving the average computation speed of an arithmetic processor. However, for an M-stage pipeline, the result of a given operation is available only M clock periods after initiating the computation. In a recursive filter, the computation of y(n) cannot be initiated before the computations of y(n-1) through y(n-N) are completed. H.B. Voelcker and E.E. Hartquist (1970) and P.M.Kogge and H.S. Stone (1973) independently devised augmentation techniques for resolving the dependence problem in the computation of y(n). However, the augmentation required to ensure stability may be excessively high, resulting in a very complex numerator realization. A technique which results in a minimum order augmentation is presented. The complexity of the resulting filter design is very much lower. Various pipelining architectures are presented. It is demonstrated by an example that when compared to the prototype filter, the augmented filter has a lower coefficient sensitivity and better roundoff noise performance View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A double Nyquist digital product detector for quadrature sampling

    Page(s): 1670 - 1681
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1012 KB)  

    A technique for digitally obtaining the in-phase (I) and quadrature (Q) components of an IF signal is presented. Initially, the input bandpass signal is mixed to a carrier frequency that is one-fourth of the sampling rate of a single A/D converter. The digitized bandpass signal is converted into its I and Q components at one-half the A/D sample rate by a digital product detector (DPD) composed of a commutator, two sign alternators, and two FIR fractional-phase interpolator filters. This simple structure can yield image performance that is limited by A/D quantization using relatively low interpolator filter orders and IF bandwidths as large as one-half the sampling rate of the A/D converter. The DPD performs Nyquist limit demodulation of the sampled bandpass signal and, therefore, requires a minimal sampling rate. The theory of operation, an analytic proof, design methodology, and simulated performance results are presented. Simulated results show that -86 dB images can be obtained with 8-tap FIR interpolators and a 12 bit A/D converter. A VLSI implementation is also presented View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • Seismic signal understanding: a knowledge-based recognition system

    Page(s): 1787 - 1806
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (1620 KB)  

    The authors address the issue of automating routine signal analysis in the seismological domain and propose an approach that combines artificial intelligence and signal processing techniques. Distinctive features of the knowledge involved in the expert activity are investigated and used to design a knowledge-based system to support seismological interpretation. The architecture of the system, which is based on the blackboard scheme, is discussed. The implementation of a prototype (SNA2) is presented, and details are given on its hybrid problem-solving activity. Emphasis is given to the initial, selective inspection of data records, a critical aspect on the interpretive process; accurate parameter estimates are seen as subsequent, straightforward applications of well-known procedures. Several solutions are proposed to modeling the expert's focus of attention, simple but effective tools are adopted to extract relevant signal features, and a method is proposed for approximate location of events. Results of the application of the system confirm the effectiveness of the approach View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • A variable step size LMS algorithm

    Page(s): 1633 - 1642
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (696 KB)  

    A least-mean-square (LMS) adaptive filter with a variable step size is introduced. The step size increases or decreases as the mean-square error increases or decreases, allowing the adaptive filter to track changes in the system as well as produce a small steady state error. The convergence and steady-state behavior of the algorithm are analyzed. The results reduce to well-known results when specialized to the constant-step-size case. Simulation results are presented to support the analysis and to compare the performance of the algorithm with the usual LMS algorithm and another variable-step-size algorithm. They show that its performance compares favorably with these existing algorithms View full abstract»

    Full text access may be available. Click article title to sign in or learn about subscription options.
  • On fast evaluation of bivariate polynomials at equispaced arguments

    Page(s): 1813 - 1816
    Save to Project icon | Request Permissions | Click to expandQuick Abstract | PDF file iconPDF (272 KB)  

    The initial value problem arising in the recursive evaluation of a 2D polynomial at equispaced points is treated in detail; the results facilitate efficient implementation of Bose's recursive algorithm. The computational complexity is compared with that involved in a direct computation, and some general observations are made for an alternative scheme proposed by X. Nie and R. Unbehauen (1989) View full abstract»

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

Aims & Scope

IEEE Transactions on Signal Processing covers novel theory, algorithms, performance analyses and applications of techniques for the processing, understanding, learning, retrieval, mining, and extraction of information from signals

Full Aims & Scope

Meet Our Editors

Editor-in-Chief
Zhi-Quan (Tom) Luo
University of Minnesota