Record: 18 AUTHOR: Keerthi, Arvind V.; Mathur, Amit.; Shynk, John J. TITLE: Misadjustment and tracking analysis of the constant modulus array. SOURCE: IEEE Transactions on Signal Processing v. 46 (Jan. '98) p. 51- 8 bibl diag. STANDARD NO: 1053-587X DATE: 1998 PLACE: United States RECORD TYPE: art CONTENTS: feature article ABSTRACT: The convergence and tracking properties of the constant modulus (CM) array were analyzed using a least-mean-square approximation. The CM array is a blind adaptive beamformer that can separate cochannel signals. Direction finding of the source can be achieved using a follow-on adaptive signal canceler. Expressions for the misadjustment of the adaptive algorithms were derived and a tracking model was developed to predict the behavior of the system during fades. The adaptive signal canceler was found to contribute more to the overall misadjustment than the adaptive CM beamformer. SUBJECT: Adaptive signal processing. Tracking algorithms. Beamforming.

Record: 23 AUTHOR: Minkoff, J. TITLE: The operation of multichannel feedforward adaptive systems. SOURCE: IEEE Transactions on Signal Processing v. 45 (Dec. '97) p. 2993-3005 bibl diags. STANDARD NO: 1053-587X DATE: 1997 PLACE: United States RECORD TYPE: art CONTENTS: feature article
ABSTRACT: The author presents a generalized formulation, applicable to both random and deterministic signals, to describe the operation of multichannel feedforward adaptive systems. Expressions for the optimum multichannel adaptive-filter transfer functions and for the minimized cost function were derived. In addition, the maximum achievable error-reduction performance of the systems was obtained. SUBJECT: Cost functions. Adaptive signal processing.

Record: 25 AUTHOR: Schodorf, Jeffrey B.; Williams, Douglas B. TITLE: Array processing techniques for multiuser detection. SOURCE: IEEE Transactions on Communications v. 45 (Nov. '97) p. 1375- 8 bibl diags. STANDARD NO: 0090-6778 DATE: 1997 PLACE: United States RECORD TYPE: art CONTENTS: feature article
ABSTRACT: Techniques employed in the area of adaptive array signal processing were applied to the multiuser detection problem. A robust detector that is suitable for use in the presence of modeling errors and a reduced-rank detector with improved transient behavior relative to full-rank detectors were derived. Bit-error-rate curves and least-mean-square learning curves were utilized to show algorithm performance. SUBJECT: Multiuser detection. Constrained optimization methods. Adaptive signal processing.

Record: 26 AUTHOR: Feng, Jinwei.; Gan, Woon-Seng. TITLE: A broadband self-tuning active noise equaliser. SOURCE: Signal Processing v. 62 no2 (Oct. '97) p. 251-6 bibl diags. STANDARD NO: 0165-1684 DATE: 1997 PLACE: Netherlands RECORD TYPE: art CONTENTS: feature article
ABSTRACT: The authors propose a broadband self-tuning active noise equalizer (SANE) to extend the applicability of the active noise equalizer (ANE) technique to the practical situations where the primary noise power varies with time. Using a variable gain factor, the proposed SANE can shape the residual noise spectrum as the existing ANE does and can also automatically adjust the residual noise power. Computer simulations are used to validate the proposed algorithm. SUBJECT: Active noise control. Adaptive signal processing. Self-tuning controllers.

Record: 27 AUTHOR: Li, Qiang.; Fan, H. (Howard) TITLE: On properties of information matrices of delta-operator based adaptive signal processing algorithms. SOURCE: IEEE Transactions on Signal Processing v. 45 (Oct. '97) p. 2454-67 bibl. STANDARD NO: 1053-587X DATE: 1997 PLACE: United States RECORD TYPE: art CONTENTS: feature article ABSTRACT: The authors investigated the properties of information matrices of the difference operator or the so-called delta (d) operator-based algorithms for adaptive signal processing. It was demonstrated that the conditioning of a transformed information matrix in the d domain is always better than that of the original information matrix in the conventional q domain for ill-conditioned problems. The results provide an explanation of the advantages of using delta operator algorithms for adaptive signal processing. The problem of the effect of D in the delta operator algorithms is also discussed. SUBJECT: Operators (Mathematics). Adaptive signal processing. Matrices (Mathematics). Record:

28 AUTHOR: Charleston, Sonia.; Azimi-Sadjadi, Mahmood R.; Gonzalez-Camarena, Ramon. TITLE: Interference cancellation in respiratory sounds via a multiresolution joint time-delay and signal-estimation scheme. SOURCE: IEEE Transactions on Biomedical Engineering v. 44 (Oct. '97) p. 1006-19 bibl diags. STANDARD NO: 0018-9294 DATE: 1997 PLACE: United States RECORD TYPE: art CONTENTS: feature article
ABSTRACT: The authors address the problem of separating out heart sounds from acquired respiratory sounds using a new joint time-delay and signal estimation (JTDSE) procedure. The primary characteristic of the JTDSE method is that it is based on the common information about the time delay between reference and acquired signals that is shared by all the levels and information-bearing subbands. Other advantages of the proposed method include robustness in the presence of noise and accuracy in time-delay estimation. SUBJECT: Adaptive filters - Mathematical models. Wavelet transforms. Time delay estimation. Signal processing.

Record: 29 AUTHOR: Reif, Konrad.; Luo, Fa-Long.; Unbehauen, Rolf. TITLE: The exponential stability of the invariant-norm PCA algorithm. SOURCE: IEEE Transactions on Circuits and Systems. Part II, Analog and Digital Signal Processing v. 44 (Oct. '97) p. 873-6 bibl. STANDARD NO: 1057-7130 DATE: 1997 PLACE: United States RECORD TYPE: art CONTENTS: feature article
ABSTRACT: A recently proposed principal component analysis algorithm was investigated. The solutions of the corresponding differential equations were shown to converge to the principal eigenvectors of the autocorrelation matrix. An exponential decaying bound for the error was computed. SUBJECT: Principal components analysis. Adaptive signal processing. Unsupervised learning.

Record: 30 AUTHOR: Omoto, A.; Takashima, K.; Fujiwara, K. TITLE: Active suppression of sound diffracted by a barrier: an outdoor experiment. SOURCE: The Journal of the Acoustical Society of America v. 102 (Sept. '97) p. 1671-9 bibl il diags. STANDARD NO: 0001-4966 DATE: 1997 PLACE: United States RECORD TYPE: art CONTENTS: feature article
ABSTRACT: The active control method was used to suppress the sound diffracted by an outdoor barrier. This method operated by the cancellation of the sound pressure at the diffraction edge of the barrier, which normally behaves like the virtual source of the diffracted field. The results of two experiments are shown in this paper. In the first experiment, we employed two independent controllers that utilized multi-channel adaptive signal processing to minimize the sum of the mean square of the sound pressure at four and six points along the diffraction edge. Measurement of sound pressure levels at various distances from the barrier showed effective sound suppression, with about 6-dB excess attenuation over the barrier's insertion loss at the receiver at a distance of 50 m. A practically realistic noise source, a fan blower, was used as a primary source in the second experiment and the moderate attenuation could be achieved at almost all the receiver points. The results obtained in these two experiments provided the experimental verification of the strategy for the active suppression of sound diffracted by a noise barrier. Reprinted by permission of the publisher. SUBJECT: Active noise control. Noise barriers. Adaptive signal processing.

Record: 36 AUTHOR: Strobach, Peter. TITLE: Fast recursive orthogonal iteration subspace tracking algorithms and applications. SOURCE: Signal Processing v. 59 no1 (May '97) p. 73-100 bibl. STANDARD NO: 0165-1684 DATE: 1997 PLACE: Netherlands RECORD TYPE: art CONTENTS: feature article
ABSTRACT: The author introduces a class of fast recursive subspace tracking algorithms that is based on the orthogonal iteration principle. This includes the derivation of realizations with O(Nr2) and O(Nr) complexity, where N is the model order and r(less than or equal)N is the rank of the underlying data covariance matrix. Comparisons indicate that the algorithms require fewer operations and offer a better angle performance than the recently introduced transposed QR singular value decomposition subspace tracker. Also described are complete quasi-code tables of the algorithms and applications to adaptive frequency estimation and rank adaptive subspace filtering. SUBJECT: Adaptive signal processing. Iterative algorithms. Tracking algorithms.

Record: 39 AUTHOR: Goldstein, J. Scott.; Reed, Irving S. TITLE: Reduced-rank adaptive filtering. SOURCE: IEEE Transactions on Signal Processing v. 45 (Feb. '97) p. 492- 6 bibl diags. STANDARD NO: 1053-587X DATE: 1997 PLACE: United States RECORD TYPE: art CONTENTS: feature article
ABSTRACT: A new rank reduction scheme for adaptive filtering problems is presented. The scheme utilizes a cross-spectral metric for selecting the optimal lower dimensional subspace for reduced- rank adaptive filtering as a function of the basis vectors of the full-rank space. It is found that the steady-state solution in the subspace opted for by the cross-spectral metric provides an upper bound for the solution found in the space spanned by the eigenvectors that correspond with the largest eigenvalues of the full-rank correlation matrix. SUBJECT: Adaptive signal processing.

Record: 51 AUTHOR: Bollini, P.; Chisci, L.; Farina, A. TITLE: QR versus IQR algorithms for adaptive signal processing: performance evaluation for radar applications. SOURCE: IEE Proceedings. Radar, Sonar and Navigation v. 143 (Oct. '96) p. 328-40 bibl diags. STANDARD NO: 1350-2395 DATE: 1996 PLACE: United Kingdom RECORD TYPE: art CONTENTS: feature article SUBJECT: Adaptive signal processing. Beamforming. CORDIC algorithms.

Record: 53 AUTHOR: Liu, Shih-Chii.; Mead, Carver. TITLE: Continuous-time adaptive delay system. SOURCE: IEEE Transactions on Circuits and Systems. Part II, Analog and Digital Signal Processing v. 43 (Nov. '96) p. 744-51 bibl diags. STANDARD NO: 1057-7130 DATE: 1996 PLACE: United States RECORD TYPE: art CONTENTS: feature article SUBJECT: Adaptive signal processing. Delay lines - Mathematical models. Phase locked loops.

Record: 57 AUTHOR: Rabideau, Daniel J. TITLE: Fast, rank adaptive subspace tracking and applications. SOURCE: IEEE Transactions on Signal Processing v. 44 (Sept. '96) p. 2229-44 bibl diags. STANDARD NO: 1053-587X DATE: 1996 PLACE: United States RECORD TYPE: art CONTENTS: feature article SUBJECT: Tracking algorithms. Subspaces. Adaptive signal processing.

Record: 58 AUTHOR: Guerci, Joseph R.; Feria, Erlan H. TITLE: Application of a least squares predictive-transform modeling methodology to space-time adaptive array processing. SOURCE: IEEE Transactions on Signal Processing v. 44 (July '96) p. 1825-33 bibl diags. STANDARD NO: 1053-587X DATE: 1996 PLACE: United States RECORD TYPE: art CONTENTS: feature article SUBJECT: Moving target indicators. Radar signal processing.

Record: 60 AUTHOR: Jin, Qu.; Luo, Zhi-Quan (Tom); Wong, Kon Max. TITLE: Optimum filter banks for signal decomposition and its application in adaptive echo cancellation. SOURCE: IEEE Transactions on Signal Processing v. 44 (July '96) p. 1669-80 bibl diags. STANDARD NO: 1053-587X DATE: 1996 PLACE: United States RECORD TYPE: art CONTENTS: feature article SUBJECT: Adaptive signal processing. Multidimensional filters - Design. Echo suppression.

Record: 61 AUTHOR: Yu, Shiann-Jeng.; Lee, Ju-Hong. TITLE: Adaptive array beamforming for cyclostationary signals. SOURCE: IEEE Transactions on Antennas and Propagation v. 44 (July '96) p. 943-53 bibl diags. STANDARD NO: 0018-926X DATE: 1996 PLACE: United States RECORD TYPE: art CONTENTS: feature article SUBJECT: Beamforming - Mathematical models. Adaptive signal processing. Adaptive algorithms.

Record: 62 AUTHOR: Candy, James V.; Sullivan, Edmund J. TITLE: Model-based identification: an adaptive approach to ocean- acoustic processing. SOURCE: IEEE Journal of Oceanic Engineering v. 21 (July '96) p. 273- 89 bibl diag. STANDARD NO: 0364-9059 DATE: 1996 PLACE: United States RECORD TYPE: art CONTENTS: feature article SUBJECT: Underwater sound. Acoustic signal processing. Parameter estimation.

Record: 65 AUTHOR: Bermudez, Jose Carlos M.; Bershad, Neil J. TITLE: A nonlinear analytical model for the quantized LMS algorithm-- the arbitrary step size case. SOURCE: IEEE Transactions on Signal Processing v. 44 (May '96) p. 1175- 83 bibl diags. STANDARD NO: 1053-587X DATE: 1996 PLACE: United States RECORD TYPE: art CONTENTS: feature article
ABSTRACT: An investigation of the quantization effects in the finite precision least mean squares (LMS) algorithm with arbitrary step sizes. The LMS algorithm is unquestionably one of the most popular algorithms for digital implementation of real- time high-speed adaptive filters. Deterministic nonlinear recursions are developed for the mean and second moment matrix of the weight vector about the Wiener weight for white Gaussian data models and small algorithm step sizes. These recursions are solved numerically and demonstrated to be in very close agreement with the Monte Carlo simulations during all phases of the adaptation process. A design example is used to illustrate how the theory can be employed to select the number of quantizer bits and the adaptation stepsize m to yield a desired transient behavior and cancellation performance. SUBJECT: Adaptive signal processing. Least square algorithms.

Record: 66 AUTHOR: Doroslovacki, Milos.; Fan, H. (Howard) TITLE: Wavelet-based linear system modeling and adaptive filtering. SOURCE: IEEE Transactions on Signal Processing v. 44 (May '96) p. 1156- 67 bibl diags. STANDARD NO: 1053-587X DATE: 1996 PLACE: United States RECORD TYPE: art CONTENTS: feature article
ABSTRACT: The authors report on wavelet-based linear system modeling and adaptive filtering. Adaptive filtering of signals attempts to generate, from an input signal, an output signal with desired properties, while learning to do it in the best possible way. Optimal coefficients and the corresponding minimum mean square error were found, and they were, in general, shown to be time varying. Least-mean-square adaptive filtering algorithms were developed for on-line filtering and system identification. Theoretically and by simulations, the advantages of using wavelet-based filtering were demonstrated: