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H_∞ filtering of network-based systems with communication constraints

IET signal processing 2010年第1期4卷 69-77页

作者： Song, H. Zhang, W. -A. Yu, L. Zhejiang Univ Technol Dept Automat Zhejiang Prov United Key Lab Embedded Syst Hangzhou 310032 Zhejiang Peoples R China

This work studies the H-infinity filtering problem for a class of discrete-time network-based systems with communication constraints. The considered plant has multiple outputs and only one of them can be sent to the filter at each transmission instant because of the communication constraints. A new compensation scheme is proposed, upon which the filtering error system is modelled as a switched system by considering mode-dependent filters. Sufficient conditions are derived for the filtering error system to be exponentially stable and ensure a prescribed H-infinity disturbance attenuation level bound by using the average dwell-time method. Based on the derived conditions, a design procedure for the H-infinity filter is also presented. An illustrative example is given to show the effectiveness of the proposed method.

关键词： filtering methods in signal processing compensation filtering error system discrete time network based systems average dwell-time method H∞ optimisation H∞ disturbance attenuation level discrete time filters switched system Optimisation techniques filtering theory signal processing theory communication constraints H∞ filtering mode dependent filters

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Least square identification of alias components of linear periodically time-varying systems and optimal training signal design

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IET signal processing 2010年第2期4卷 149-157页

作者： Yin, W. Mehr, A. Saadat Univ Saskatchewan Dept Elect & Comp Engn Saskatoon SK S7N 5A9 Canada

A least-squares (LS) method for identifying alias components of discrete linear periodically time-varying (LPTV) systems is proposed. The authors apply a periodic input signal to a finite impulse response (FIR)-LPTV system and measure the noise-contaminated output. The output of this LPTV system has the same period as the input when the period of the input signal is a multiple of the period of the LPTV system. The authors show that the input and the output can be related by using the discrete Fourier transform. In the frequency domain, an LS method can be used to identify the alias components. A lower bound on the mean square error (MSE) of the estimated alias components is given for FIR -LPTV systems. The optimal training signal achieving this lower MSE bound is designed subsequently. The algorithm is extended to the identification of infinite impulse response (IIR)-LPTV systems as well. Simulation results show the accuracy of the estimation and the efficiency of the optimal training signal design.

关键词： discrete Fourier transforms discrete systems infinite impulse response-LPTV systems least square identification filtering methods in signal processing Integral transforms in numerical analysis alias components least squares approximations IIR filters mean square error methods optimal training signal design signal processing theory time-varying systems finite impulse response-LPTV system Interpolation and function approximation (numerical analysis) discrete Fourier transform linear systems FIR filters discrete linear periodically time-varying systems mean square error

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Frequency-domain adaptive detectors for single-carrier frequency-domain equalisation in multiuser direct-sequence ultra-wideband systems based on structured channel estimation and direct adaptation

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IET COMMUNICATIONS 2010年第13期4卷 1636-1650页

作者： Li, S. de Lamare, R. C. Univ York Dept Elect Commun Res Grp York YO10 5DD N Yorkshire England

Here, the authors propose two adaptive detection schemes based on single-carrier frequency-domain equalisation (SC-FDE) for multiuser direct-sequence ultra-wideband systems, which are termed structured channel estimation (SCE) and direct adaptation (DA). Both schemes use the minimum mean square error (MMSE) linear detection strategy and employ a cyclic prefix. In the SCE scheme, adaptive channel estimation is performed in the frequency domain and the despreading is implemented in the time domain after the FDE. In this scheme, the MMSE detection requires the knowledge of the number of users and the noise variance. For this purpose, simple algorithms are proposed for estimating these parameters. In the DA scheme, the interference suppression task is fulfilled with only one adaptive filter in the frequency domain and a new signal expression is adopted to simplify the design of such a filter. Least mean squares, recursive least squares and conjugate gradient adaptive algorithms are then developed for both schemes. A complexity analysis compares the computational complexity of the proposed algorithms and schemes, and simulation results for the downlink illustrate their performance.

关键词： Electromagnetic compatibility and interference adaptive filters adaptive channel estimation least mean squares methods minimum mean square error linear detection strategy least mean squares algorithm ultra wideband communication frequency-domain adaptive detectors structured channel estimation multiuser direct-sequence ultra-wideband systems channel estimation direct adaptation adaptive filter gradient methods filtering methods in signal processing equalisers interference suppression task Interpolation and function approximation (numerical analysis) recursive least squares algorithm single-carrier frequency-domain equalisation computational complexity Radio links and equipment cyclic prefix conjugate gradient adaptive algorithms Communication channel equalisation and identification spread spectrum communication interference suppression

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Shift-invariant B-spline wavelet transform for multi-scale analysis of neuroelectric signals

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IET signal processing 2010年第6期4卷 603-609页

作者： Olkkonen, H. Olkkonen, J. T. Univ Kuopio Dept Phys FIN-70211 Kuopio Finland VTT Tech Res Ctr Finland Espoo 02044 Finland

In wavelet theory the two-scale dilation equation has a central role. B-splines serve as good candidates for wavelet analysis, since they obey the two-scale dilation equation. This work describes the B-spline wavelet transform, which is based on the polyphase decomposition of the two-scale dilation equation. We construct a linear quadrature mirror filter (QMF) B-spline wavelet filter bank, which can be effectively implemented by the polyphase filters. The interpolating property of the two-scale dilation equation is applied for constructing the shift-invariant complex QMF B-spline wavelets. The validity of the B-spline wavelet transform is warranted in multi-scale analysis of neuroelectric signal waveforms.

关键词： wavelet transforms shift-invariant wavelets polyphase filters Bioelectric signals wavelet filter bank channel bank filters wavelet transform neuroelectric signals multiscale analysis filtering methods in signal processing two-scale dilation equation medical signal processing polyphase decomposition splines (mathematics) Integral transforms Interpolation and function approximation (numerical analysis) quadrature mirror filters B-spline linear quadrature mirror filter

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Kalman filtering for continuous-time systems with time-varying delay

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IET CONTROL THEORY AND APPLICATIONS 2010年第4期4卷 590-600页

作者： Wang, W. Zhang, H. Xie, L. Harbin Inst Technol Shenzhen Grad Sch Shenzhen Peoples R China Shandong Univ Sch Control Sci & Engn Jinan Peoples R China Nanyang Technol Univ Sch Elect & Elect Engn Singapore Singapore

The optimal filtering problem for continuous-time linear systems with time-varying delay is considered. The purpose is to find a Riccati equation-based solution to the linear minimum mean square error filter. The time-varying delayed observation is firstly rewritten as an equivalent observation with multiple constant delays by defining a binary variable to model the arrival process of the observation. Then, by constructing certain multiple channel observations sequence that contains the same amount of information as the original one and applying the re-organised innovation analysis approach, the proposed problem is transformed to performing standard time-varying Kalman filtering.

关键词： signal processing theory least mean squares methods Kalman filters Riccati equation time-varying systems Time-varying control systems re-organised innovation analysis approach linear systems time-varying filters linear minimum mean square error filter delays continuous time filters time-varying Kalman filtering Mathematical analysis time-varying delay filtering methods in signal processing binary variable continuous time linear systems multiple channel observations sequence Riccati equations Distributed parameter control systems optimal filtering problem

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Constrained adaptive filtering algorithms based on conjugate gradient techniques for beamforming

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IET signal processing 2010年第6期4卷 686-697页

作者： Wang, L. de Lamare, R. C. Univ York Dept Elect Commun Res Grp York YO10 5DD N Yorkshire England

This article proposes constrained adaptive algorithms based on the conjugate gradient (CG) method for adaptive beamforming. The proposed algorithms are derived for the implementation of the beamformer according to the minimum variance and constant modulus criteria subject to a constraint on the array response. A CG-based weight vector strategy is created for enforcing the constraint and computing the weight expressions. The devised algorithms avoid the covariance matrix inversion and exhibit fast convergence with low complexity. A complexity analysis compares the proposed algorithms with the existing ones. The convergence properties of the CCM criterion are studied, conditions for convexity are established and a convergence analysis for the proposed algorithms is derived. Simulation results are conducted for both stationary and non-stationary scenarios, showing the convergence and tracking ability of the proposed algorithms.

关键词： array signal processing CCM criterion adaptive beamforming covariance matrix inversion CG-based weight vector strategy complexity analysis Interpolation and function approximation (numerical analysis) filtering methods in signal processing computational complexity adaptive filters tracking ability convergence analysis conjugate gradient methods constrained adaptive filtering algorithms signal processing theory conjugate gradient techniques

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Three-channel variable filter-bank for digital hearing aids

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IET signal processing 2010年第2期4卷 181-196页

作者： Deng, T. -B. Toho Univ Fac Sci Dept Informat Sci Chiba 2748510 Japan

This study proposes a three-channel (3-channel) variable filter-bank (VFB) that consists of variable lowpass, variable bandpass and variable highpass digital filters. The three variable digital filters are obtained from a normalised analog prototype Chebyshev type-I lowpass filter using analog frequency transformations along with a modified bilinear transformation. Since both the magnitudes (gains) and band edge frequencies of the three variable digital filters are independently adjustable, the 3-channel VFB is considerably flexible, and can be successfully applied for compensating various hearing loss patterns in digital hearing aids. Various audiograms have been used to verify that high-accuracy fittings can be achieved with low order variable filters. Moreover, the authors reveal and theoretically prove the numerator coefficient-symmetries of the variable lowpass, variable bandpass and variable highpass filters, and show that each variable filter requires only one multiplication for its numerator filtering operations, so the total number of multiplications can be significantly reduced. More specifically, only 11 multiplications and 14 additions are required in the whole 3-channel VFB. Therefore the 3-channel VFB has extremely simple structure and high tuning flexibility for hearing aids.

关键词： Prosthetics and orthotics three channel variable filter bank Chebyshev filters high-pass filters Digital filters channel bank filters variable highpass digital filters hearing aids variable lowpass digital filters filtering methods in signal processing Chebyshev type-I lowpass filter modified bilinear transformation variable bandpass digital filters numerator filtering operations high tuning flexibility digital hearing aids analog frequency transformations digital filters low-pass filters audiograms

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Adaptive sensor selection in wireless sensor networks for target tracking

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IET signal processing 2010年第5期4卷 530-536页

作者： Zoghi, M. Kahaei, M. H. Iran Univ Sci & Technol Sch Elect Engn Tehran Iran

Sensor selection in wireless sensor networks (WSN) is considered for target tracking. A decentralised estimation scenario is considered based on the extended information filter. A cost function using the geometrical dilution of precision (GDOP) measure is derived for active sensor selection. Accordingly, a new adaptive sensor selection (ASS) algorithm is proposed in which the number of active sensors is adaptively determined based on the absolute local innovations vector. Simulation results show that the tracking accuracy of the ASS is comparable to that of the other algorithms, but with less energy consumption.

关键词： adaptive sensor selection absolute local innovations vector extended information filter filtering methods in signal processing Wireless sensor networks geometrical dilution of precision decentralised estimation sensor placement target tracking Kalman filters wireless sensor networks active sensors

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Fusion of panchromatic and multispectral images using temporal Fourier transform

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IET IMAGE processing 2010年第4期4卷 255-260页

作者： Mahyari, A. G. Yazdi, M. Shiraz Univ Fac Elect & Comp Engn Shiraz Iran

Image fusion techniques can be used to enhance the resolution of multispectral (MS) image, which is helpful for categorisation, recognition as well as other decision making processes. In this paper, a new class of image fusion algorithms is proposed that decomposes images into similar and non-similar information and fuses the corresponding non-similar one. It is based on temporal Fourier analysis. The details of panchromatic (PAN) image are fused with details of MS image by eliminating their similar information which is done by high-pass filtering of their temporal Fourier transform. The cut-off frequency of this filtering is obtained adaptively based on the input MS and PAN images. It results in minimum spectral and spatial distortion. The experimental results demonstrate the superior performance of the proposed method

关键词： image resolution PAN images cut-off frequency multispectral image resolution filtering methods in signal processing spectral analysis high pass filtering Computer vision and image processing techniques image recognition panchromatic images image fusion temporal Fourier transform Optical, image and video signal processing Integral transforms image classification decision making processes filtering theory Fourier transforms image fusion algorithm MS images

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Switchable filtering in Vivaldi antenna

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ELECTRONICS LETTERS 2010年第7期46卷 477-U21页

作者： Hamid, M. R. Hall, P. S. Gardner, P. Ghanem, F. Univ Birmingham Birmingham B15 2TT W Midlands England Univ Teknol Malaysia Fak Kejuruteraan Elekt Skudai Malaysia

Presented is a new frequency switchable Vivaldi antenna that has a capability to operate in a wideband mode (1 - 3 GHz) and reconfigure to six different subbands of operations. The reconfiguration is realised by coupling and changing the effective electrical length of ring slots inserted in the structure by means of pin diode switches. To examine antenna performances, simulated and measured results are presented. Good impedance matches and radiation patterns have been achieved. The proposed antenna is suitable for wideband and multimode radio applications.

关键词： Junction and barrier diodes filtering theory switchable filtering filtering methods in signal processing antenna radiation patterns slot antennas radiation patterns impedance matches Single antennas switches frequency switchable Vivaldi antenna pin diode switches Relays and switches ring slots electrical length p-i-n diodes

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