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Some significant communications processing realized with adaptive Bragg gratings

Some significant communications processing realized with ada...

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SPIE conference on advanced signal processing algorithms, architectures, and implementations viII

作者： Upton, E Wickham, M Smith, M Hart, J Flynn, R Havstad, S Nachman, P TRW Co Inc Redondo Beach CA 90278 USA

ISBN: (纸本)0819429163

Higher data capacity demands and lower interference requirements in the wireless communications arena are exploiting higher carrier frequencies and wider modulation bandwidths. Circuitry which can perform intermediate frequency processing over these more demanding ranges is needed to provide complex signal processing without commercial penalties. Photonics technologies utilizing Bragg Grating signal processing (BGSP) can bridge the gap between the very high frequency RF millimeter wave integrated circuit (MIMIC) domains at the antenna interface and the CMOS digital signal processors at the base band frequency interface. The desirable benefits of multiple tap adaptive finite impulse response (FIR) and infinite impulse response (IIR) filters and equalizers are well known;however, they are usually the province of digital signal processing and force the sample rates prior to these processors to a higher overall system power consumption level. BGSP provides these functions with discrete taps and digital controls but at the bandwidths usually reserved for RF circuitry because the actual processing occurs at optical frequencies and at wave lengths which are compatible with integrated circuit technologies. The high performance benefits of photonic processing can be realized if the stability control of the Bragg grating is derived from the same metric which induces in photonics its sensitivity to drift. We will present a orthogonally coded tap modulation technique which stabilizes the transfer function of the signal processor and enables significant adaptive IF signal processing to be obtained with very low size, weight, and power. Our demonstration of a photonic proof-of-concept architecture is a reconfigurable multiple tap FIR filter that is dynamically controlled to perform low pass, high pass, band pass, and band stop filters operating over bandwidths of 3 GHz.

关键词： signal processing Fiber Bragg gratings Filtering (signal processing) Control systems Linear filtering Digital signal processing Optical filters Servomechanisms Modulators Electronic filtering

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Color characterization for image indexing and machine vision

Color characterization for image indexing and machine vision

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conference on advanced signal processing algorithms, architectures, and implementations X

作者： Lin, SM Nair, D Natl Instruments Austin TX 78759 USA

ISBN: (纸本)0819437611

Color representation and comparison based on the histogram has proved to be very efficient for image indexing in content-based image retrieval and machine vision applications. However, the issues of color constancy and accurate color similarity measures remain unsolved. This paper presents a new algorithm for intensity-insensitive color characterization for image retrieval and machine vision applications. The color characterization algorithm divides the HSI (hue, saturation and intensity) color space into a given number of bins in such a way that the color characterization represents all the colors in the hue/saturation plane as well as black, white and gray colors. The color distribution in these bins of the HSI space is represented in the form of a one-dimensional vector called Color Spectrum Vector (CSV). The color information that is stored in the CSV is insensitive to changes in the luminance. A weighted version of CSV called WCSV is introduced to take the similarity of the neighboring bins into account A Fuzzy Color Spectrum Vector (FCSV) color representation vector that takes into account the human uncertainty in color classification process is also introduced here. The accuracy and speed of the algorithm is demonstrated in this paper through a series of experiments on image indexing and machine vision applications.

关键词： color comparison pattern matching clustering algorithm fuzzy set theory image indexing machine vision image processing

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On the regularized solution of block banded block Toeplitz systems

On the regularized solution of block banded block Toeplitz s...

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conference on advanced signal processing algorithms, architectures, and implementations X

作者： Bini, DA Farusi, A Fiorentino, G Meini, B Dipartimento Matemat I-56127 Pisa Italy

ISBN: (纸本)0819437611

Given two n. x n Toeplitz matrices T-1 and T-2, and a vector b epsilon R-n2, consider the linear system Ax = b - eta, where eta epsilon R-n2 is an unknown vector representing the noise and A = T-1 x T-2. Recovering approximations of a, given A and b, is encountered in image restoration problems. We propose a method for the approximation of the solution a: that has good regularization properties. The algorithm is based on a modified version of Newton's iteration for matrix inversion and relies on the concept of approximate displacement rank. We provide a formal description of the regularization properties of Newton's iteration in terms of filters and determine bounds to the number of iterations that guarantee regularization. The method is extended to deal with more general systems where A = Sigma (h)(i=1) T-1((i)) x T-2((i)). The cost of computing regularized inverses is O(n log n) operations (ops), the cost of solving the system Ax = b is O(n(2) log n) ops. Numerical experiments which show the effectiveness of our algorithm are presented.

关键词： Band Toeplitz systems Newton's iteration regularization displacement rank image processing

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New signal adaptive approach for the reduction of interference terms in quadratic time-frequency distributions

New signal adaptive approach for the reduction of interferen...

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SPIE conference on advanced signal processing algorithms, architectures, and implementations viII

作者： Nickel, RM Williams, WJ Univ Michigan Dept Elect Engn & Comp Sci Ann Arbor MI 48109 USA

ISBN: (纸本)0819429163

One of the key problems in high resolution (quadratic), time-varying spectral analysis is the suppression of interference terms which can obscure the true location of auto components in the resulting time-frequency distribution (TFD). Commonly used reduced interference distributions (RIDs) tackle the problem with a properly chosen two-dimensional low pass filter (kernel). A recently published(8) novel approach uses alternative means to achieve the desired goal. The idea of the new method is to obtain an estimate of the cross terms from a given prior distribution based on the magnitude and location of its negative components. The estimate is constructed via an iterative projection method that guarantees that the resulting distribution is positive and satisfies the marginals. Even though the marginals are usually a desirable property of TFDs in general, they can impose an undesirably strong constraint on positive TFDs in particular (Examples are given in this paper). For these cases it is thus beneficial to relax the marginals-constraint. In this paper we present a new method that does not require the incorporation of this constraint and thus leads to positive TFDs with reduced interference terms but without the restrictions due to the marginals.

关键词： positive time-frequency distributions cross-terms successive projections penalty functions

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Element accessed array radar processing

Element accessed array radar processing

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SPIE conference on advanced signal processing algorithms, architectures, and implementations viII

作者： Wills, GD Rees, HD Skidmore, ID Def Evaluat & Res Agcy Malvern WR14 3PS Worcs England

ISBN: (纸本)0819429163

Digitising the signals from a phased array at the element level is becoming more practicable through advances in technology. Adaptive beamforming schemes such as a sample matrix inversion involve inverting a covariance matrix of dimension equal to the number of spatial channels. Since with an element digitised system there can be several thousand elements it becomes impractical to invert such large matrices. This paper presents a weight computation scheme for a fully digitised system. The clutter returns are firstly range gated and then the adaptive process is broken down into a smaller cascade of stages, or domains. Attenuation of the clutter is then achieved using a spatial Wiener filter in each domain. From a simulation of an airborne radar at a high pulse repetition frequency, examples are given of adapted beampatterns and clutter output doppler spectra within each domain. By choosing alias free domains, a clutter-free output is obtained for the sidelobe clutter scenario investigated. A comparison is then given with the performance of a 32 sub-array system. The sub-array system does not have the necessary control of the small scale order of the array weights to spatially null the clutter.

关键词： element level digitisation radar clutter subarrays

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Regularization of ill-posed problems using (symmetric) Cauchy-like preconditioners

Regularization of ill-posed problems using (symmetric) Cauch...

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SPIE conference on advanced signal processing algorithms, architectures, and implementations viII

作者： Kilmer, ME Northeastern Univ Dept Elect & Comp Engn Boston MA 02115 USA

ISBN: (纸本)0819429163

In certain applications, the discretization of two-dimensional integral equations can lead to systems involving matrices with block Toeplitz-Toeplitz block structure. Iterative Krylov subspace methods are sometimes employed to find regularized solutions to the related 2D discrete ill-posed problems;however, preconditioners which filter noise are needed to speed convergence to regularized solutions. We describe a preconditioning technique based on the Toeplitz-type structure of the matrix which generalizes the approaches in [1] and [2] to take advantage of symmetry and real arithmetic operations. We use fast sine transforms to transform the original system to a system whose matrix has partially reconstructible Cauchy-like blocks. The preconditioner is a block diagonal, rank m approximation to this matrix, with Cauchy-like blocks each augmented by an identity of appropriate dimension. We note that the initialization cost is in general less than that for the similar 2D preconditioner in [2] which does not take advantage of symmetry. Several examples are given which illustrate the success of our preconditioned methods.

关键词： regularization Toeplitz Cauchy-like minimal residual image processing deblurring

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Array processing based on the polarization diversity : the example of a ionospheric radio link

Array processing based on the polarization diversity : the e...

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conference on advanced signal processing algorithms, architectures, and implementations XIV

作者： Ehrel, YM Perrine, C Lemur, D Bourdillon, A Ctr Rech Ecoles Coetquidan F-56381 Guer France Univ Rennes 1 Inst Elect & Telecommun F-35042 Rennes France

ISBN: (纸本)0819454974

The aim of this paper is to present a high data rate transmission system through the ionospheric channel in the HF band (3-30 MHz). The applications expected in this study are image transmission and real-time videoconferencing. Very high rates are required compared to the standard modems (4.8 kbits/s). Therefore. an array processing performs in the multi channel receiving system. Its main originality stands in the development of a compact device of collocated antennas. the spatial responses of which are different one from each other. Besides. synchronization (Zero Crossing Detector) and spatio temporal equalization (L.M.S. algorithm) as well resort to classical and well-tested techniques involving training sequences. An experimental radio link has been under test with a range of 800 km. The corresponding results underlines the improvement of the bit transfer rate which reaches 20 kbits/s (QAM 16 in a 6 kHz bandwidth) or 30 kbits/s (QAM 16 in a 9 kHz bandwidth). Several transmitted images are presented and appear quite consistent with the original..

关键词： ionosphere collocated antennas spatio temporal equalization image compression

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Non-linear signal processing using index calculus DBNS arithmetic

Non-linear signal processing using index calculus DBNS arith...

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conference on advanced signal processing algorithms, architectures, and implementations X

作者： Muscedere, R Jullien, GA Dimitrov, VS Miller, WC Univ Windsor VLSI Res Grp Windsor ON N9B 3P4 Canada

ISBN: (纸本)0819437611

This paper discusses the use of a recently introduced index calculus Double-Base Number System (IDBNS) for representing and processing numbers for non-linear digital signal processing;the target application is a digital hearing aid processor. The IDBNS representation uses 2 orthogonal bases (2 and 3) to represent real numbers with arbitrary precision. By restricting the number of digits to one or two, it is possible to efficiently represent the real number using the indices of the bases rather than the distribution of the digits. In this paper we discuss the use of the two-digit form of this representation (2-IDBNS) to efficiently perform arithmetic associated with the non-linear processing required to correct the usual forms of hearing loss in a digital hearing aid. The non-linear processing takes the form of dynamic range compression as a function of frequency band. Currently developed digital hearing instrument processors require large dynamic range representations (20-24 bits) in order to accurately generate the dynamic range compression associated with typical hearing loss. We show that the natural non-linear representation afforded by the IDBNS provides both a more efficient signal representation and a more efficient technique for processing the dynamic range compression. We pay particular attention to a novel technique of converting from a linear binary input directly to the 2-IDBNS representation using an observation of partial cyclic repetition in the indices along with near unity approximants.

关键词： computer arithmetic non-linear number representation double-base number system hearing loss

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Generalized causal moving average (GCMA) smoothing filter for real-time applications

Generalized causal moving average (GCMA) smoothing filter fo...

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conference on advanced signal processing algorithms architectures and implementations XIII

作者： Rajagopalan, R Pratt & Whitney Control & Diagnost Syst Propul Syst Anal E Hartford CT 06108 USA

ISBN: (纸本)0819450782

Moving average filters are commonly used in industries for real-time processing of noisy data. Though they perform well in filtering out the noise, they introduce significant lag in the signal. The resulting peak value of the filtered signal at the operating point is likely to be lower due to averaging of higher and lower peak signals in the averaging interval. The generalized moving average smoothing filter by Golay-Savitzky preserves the higher moments and does not suffer from the limitations imposed by the conventional moving average filter. The smoothing strategy is derived from least squares fitting of a lower order polynomial to a number of consecutive points. Due to polynomial curve fitting as opposed to a line fitting in the case of conventional moving average filter, this filter preserves the higher frequency components of the signal and their line width. This paper presents a generalized causal moving average filter deduced using the concepts in Golay-Savitzky smoothing filter for real-time applications. Golay-Savitzky filter is non-causal, relies on the future data that is not available, hence not suitable for real-time applications. Further, the designed causal filter makes use of the filtered data as opposed to the original data in the case of Golay-Savitzky. This approach allows us to conduct frequency response studies to evaluate the quality and the applicability of the filter for various signals in the aircraft engines and other engineering applications. Frequency response studies cannot carried out using the Golay-Savitzky filter. This paper also investigates the performance of various polynomial orders in-reproducing the signal from a noisy data. Some of the performance measures used are bandwidth, overshoots, and lags introduced by the filter. The mathematical technique to extract the signal and deduce the coefficients in off-line is also presented.

关键词： Filtering (signal processing) Electronic filtering signal processing Smoothing Interference (communication) Control systems Digital signal processing Data processing Electromagnetic coupling Nonlinear filtering

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Bandlimits and coherent modulation filtering of non-stationary signals

Bandlimits and coherent modulation filtering of non-stationa...

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conference on advanced signal processing algorithms, architectures, and Implementantions XviII

作者： Clark, Pascal Atlas, Les Univ Washington Dept Elect Engn Seattle WA 98195 USA

ISBN: (纸本)9780819472946

Modulation filtering is a technique for filtering slowly-varying envelopes of frequency subbands of a non-stationary signal, ideally without affecting the signal's phase and fine-structure. Coherent modulation filtering is a potentially more effective subtype of such techniques where subband envelopes are determined through demodulation of the subband signal with a coherently detected subband carrier. In this paper we propose a coherent modulation filtering technique based on detecting the instantaneous frequency of a subband from its time-frequency representation. We show that coherent modulation filtering imposes a new bandlimiting constraint on the modulation product plus the ability to recover arbitrarily chosen envelopes and carriers from their modulation product. We show that a carrier estimate based on the time-varying spectral center-of-gravity satisfies the bandlimiting condition as well as Loughlin's previously derived bandlimiting constraint on the instantaneous frequency of carrier. These bandwidth constraints lead to effective and distortion-free modulation filters, offering new approaches for potential signal modification. The spectral center-of-gravity does not satisfy the condition on arbitrary recovery, however, which somewhat limits the flexibility of coherent modulation filtering. Demonstrations are provided with speech signals.

关键词： modulation time-frequency analysis time-varying filters bandwidth speech processing

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