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Showing papers on "Filter design published in 2005"


Journal ArticleDOI
TL;DR: The transmit filters are based on similar optimizations as the respective receive filters with an additional constraint for the transmit power and has similar convergence properties as the receive Wiener filter, i.e., it converges to the matched filter and the zero-forcing filter for low and high signal-to-noise ratio, respectively.
Abstract: We examine and compare the different types of linear transmit processing for multiple input, multiple output systems, where we assume that the receive filter is independent of the transmit filter contrary to the joint optimization of transmit and receive filters. We can identify three filter types similar to receive processing: the transmit matched filter, the transmit zero-forcing filter, and the transmit Wiener filter. We show that the transmit filters are based on similar optimizations as the respective receive filters with an additional constraint for the transmit power. Moreover, the transmit Wiener filter has similar convergence properties as the receive Wiener filter, i.e., it converges to the matched filter and the zero-forcing filter for low and high signal-to-noise ratio, respectively. We give closed-form solutions for all transmit filters and present the fundamental result that their mean-square errors are equal to the errors of the respective receive filters, if the information symbols and the additive noise are uncorrelated. However, our simulations reveal that the bit-error ratio results of the transmit filters differ from the results for the respective receive filters.

792 citations


Journal ArticleDOI
TL;DR: The derivation of the details for the marginalized particle filter for a general nonlinear state-space model is derived and it is demonstrated that the complete high-dimensional system can be based on a particle filter using marginalization for all but three states.
Abstract: The particle filter offers a general numerical tool to approximate the posterior density function for the state in nonlinear and non-Gaussian filtering problems. While the particle filter is fairly easy to implement and tune, its main drawback is that it is quite computer intensive, with the computational complexity increasing quickly with the state dimension. One remedy to this problem is to marginalize out the states appearing linearly in the dynamics. The result is that one Kalman filter is associated with each particle. The main contribution in this paper is the derivation of the details for the marginalized particle filter for a general nonlinear state-space model. Several important special cases occurring in typical signal processing applications will also be discussed. The marginalized particle filter is applied to an integrated navigation system for aircraft. It is demonstrated that the complete high-dimensional system can be based on a particle filter using marginalization for all but three states. Excellent performance on real flight data is reported.

649 citations


Patent
27 Jan 2005
TL;DR: In this article, a method and system for programming the digital filter compensation coefficients of a digitally controlled switched mode power supply within a distributed power system is provided, which includes a plurality of point-of-load (POL) regulators each comprising at least one power switch adapted to convey power to a load and a digital controller adapted to control operation of the power switch responsive to a feedback measurement.
Abstract: A method and system is provided for programming the digital filter compensation coefficients of a digitally controlled switched mode power supply within a distributed power system. The distributed power system comprises a plurality of point-of-load (POL) regulators each comprising at least one power switch adapted to convey power to a load and a digital controller adapted to control operation of the power switch responsive to a feedback measurement. The digital controller further comprises a digital filter having a transfer function defined by plural filter coefficients. A serial data bus operatively connects each of the plurality of POL regulators. A system controller is connected to the serial data bus and is adapted to communicate digital data to the plurality of POL regulators via the serial data bus. The digital data includes programming data for programming the plural filter coefficients. The system controller further comprises a user interface adapted to receive the programming data therefrom.

525 citations


Journal ArticleDOI
TL;DR: In this paper, operational transconductance amplifier (OTA) and filter design for analog circuits with very low supply voltages, down to 0.5 V, are presented. But they do not consider the effect of low-voltage analog circuits on the performance.
Abstract: We present design techniques that make possible the operation of analog circuits with very low supply voltages, down to 0.5 V. We use operational transconductance amplifier (OTA) and filter design as a vehicle to introduce these techniques. Two OTAs, one with body inputs and the other with gate inputs, are designed. Biasing strategies to maintain common-mode voltages and attain maximum signal swing over process, voltage, and temperature are proposed. Prototype chips were fabricated in a 0.18-/spl mu/m CMOS process using standard 0.5-V V/sub T/ devices. The body-input OTA has a measured 52-dB DC gain, a 2.5-MHz gain-bandwidth, and consumes 110 /spl mu/W. The gate-input OTA has a measured 62-dB DC gain (with automatic gain-enhancement), a 10-MHz gain-bandwidth, and consumes 75 /spl mu/W. Design techniques for active-RC filters are also presented. Weak-inversion MOS varactors are proposed and modeled. These are used along with 0.5-V gate-input OTAs to design a fully integrated, 135-kHz fifth-order elliptic low-pass filter. The prototype chip in a 0.18-/spl mu/m CMOS process with V/sub T/ of 0.5-V also includes an on-chip phase-locked loop for tuning. The 1-mm/sup 2/ chip has a measured dynamic range of 57 dB and draws 2.2 mA from the 0.5-V supply.

471 citations


Journal ArticleDOI
TL;DR: This paper revisits the problem of mixed H/sub 2//H/sub /spl infin// filtering for polytopic discrete-time systems and makes full use of the parameter-dependent stability idea, which results in a much less conservative filter design method.
Abstract: This paper revisits the problem of mixed H/sub 2//H/sub /spl infin// filtering for polytopic discrete-time systems. Differing from previous results in the quadratic framework, the filter design makes full use of the parameter-dependent stability idea: Not only is the filter dependent of the parameters (which are assumed to reside in a polytope and be measurable online), but in addition, the Lyapunov matrices are different for the entire polytope domain, as well as for different channels with respect to the mixed performances. These ideas are realized by introducing additional slack variables to the well-established performance conditions and by employing new bounding techniques, which results in a much less conservative filter design method. A numerical example is presented to illustrate the effectiveness and advantage of the developed filter design method.

322 citations


Journal ArticleDOI
TL;DR: The Savitzky-Golay smoothing and differentiation filter is extended for even number data to validate the feasibility of such an approach and some corresponding properties are discussed.

287 citations


Journal ArticleDOI
TL;DR: From performance simulations on a wireless dispersive fading channel, it is observed that the IBDFE outperforms existing DFEs and exhibits a reduction of the computational complexity when compared against existing schemes, both in signal processing and in filter design.
Abstract: Error-propagation phenomena and computational complexity of the filters' design are important drawbacks of existing decision-feedback equalizers (DFE) for dispersive channels. In this paper, we propose a new iterative block DFE (IBDFE) which operates iteratively on blocks of the received signal. Indeed, a suitable data-transmission format must be used to allow an efficient implementation of the equalizer in the frequency domain, by means of the discrete Fourier transform. Two design methods are considered. In the first method, hard detected data are used as input of the feedback, and filters are designed according to the correlation between detected and transmitted data. In the second method, the feedback signal is directly designed from soft detection of the equalized signal at the previous iteration. Estimators of the parameters involved in the IBDFE design are also derived. From performance simulations on a wireless dispersive fading channel, we observed that the IBDFE outperforms existing DFEs. Moreover, the IBDFE exhibits a reduction of the computational complexity when compared against existing schemes, both in signal processing and in filter design.

240 citations


Journal ArticleDOI
TL;DR: The objective is to build a set of filters that are capable of responding stronger to features present in vehicles than to nonvehicles, therefore improving class discrimination and unifies filter design with filter selection by integrating genetic algorithms (GAs) with an incremental clustering approach.
Abstract: Robust and reliable vehicle detection from images acquired by a moving vehicle is an important problem with numerous applications including driver assistance systems and self-guided vehicles. Our focus in this paper is on improving the performance of on-road vehicle detection by employing a set of Gabor filters specifically optimized for the task of vehicle detection. This is essentially a kind of feature selection, a critical issue when designing any pattern classification system. Specifically, we propose a systematic and general evolutionary Gabor filter optimization (EGFO) approach for optimizing the parameters of a set of Gabor filters in the context of vehicle detection. The objective is to build a set of filters that are capable of responding stronger to features present in vehicles than to nonvehicles, therefore improving class discrimination. The EGFO approach unifies filter design with filter selection by integrating genetic algorithms (GAs) with an incremental clustering approach. Filter design is performed using GAs, a global optimization approach that encodes the Gabor filter parameters in a chromosome and uses genetic operators to optimize them. Filter selection is performed by grouping filters having similar characteristics in the parameter space using an incremental clustering approach. This step eliminates redundant filters, yielding a more compact optimized set of filters. The resulting filters have been evaluated using an application-oriented fitness criterion based on support vector machines. We have tested the proposed framework on real data collected in Dearborn, MI, in summer and fall 2001, using Ford's proprietary low-light camera.

235 citations



Journal ArticleDOI
17 Jan 2005
TL;DR: In this article, a new combline filter structure with a continuous tunability for both the center frequency and bandwidth is presented, which is achieved by placing variable coupling reducers between the filter resonators.
Abstract: A new combline filter structure with a continuous tunability for both the center frequency and bandwidth is presented in this paper. The passband-width tunability is achieved by placing variable coupling reducers between the filter resonators. The coupling reducers, operating as bandwidth control subnetworks, are designed as detuned resonators made up of a line segment ending in a variable capacitor. The proposed filter structure is experimentally validated with the design, construction in suspended stripline technology, and characterization of a low-cost filter prototype for terrestrial digital video broadcasting receivers operating in the UHF band (470-862 MHz). Other relevant factors, such as the intermodulation distortion produced by the varactors used to control the bandwidth electronically or the power-handling performance of the constructed filter, are also discussed. The reconfigurable filter module described in this paper is very suitable for the design of flexible multifunction receiver subsystems simultaneously supporting signals with a different bandwidth.

225 citations


Journal ArticleDOI
TL;DR: Differential evolution (DE) algorithm is a new heuristic approach mainly having three advantages; finding the true global minimum of a multimodal search space regardless of the initial parameter values, fast convergence, and using a few control parameters.
Abstract: Any digital signal processing algorithm or processor can be reasonably described as a digital filter. The main advantage of an infinite impulse response (IIR) filter is that it can provide a much better performance than the finite impulse response (FIR) filter having the same number of coefficients. However, they might have a multimodal error surface. Differential evolution (DE) algorithm is a new heuristic approach mainly having three advantages; finding the true global minimum of a multimodal search space regardless of the initial parameter values, fast convergence, and using a few control parameters. In this work, DE algorithm has been applied to the design of digital IIR filters and its performance has been compared to that of a genetic algorithm.

Book
05 Oct 2005
TL;DR: The z Transform Method Revisited, a model from Other Models, and a solution using MATLAB Functions to Solve Difference Equations Using the Classical Method are presented.
Abstract: Preface. 1. Introduction. 1.1 Introduction. 1.2 Application of DSP. 1.3 Discrete-Time Signals. 1.4 History of Filter Design. 1.5 Analog and Digital Signal Processing. 1.6 Summary. Problems. References. 2. Time-Domain Analysis and z Transform. 2.1 A Linear, Time-Invariant System. 2.2 z Transform Theory. 2.3 Using z Transform to Solve Difference Equations. 2.4 Solving Difference Equations Using the Classical Method. 2.5 z Transform Method Revisited. 2.6 Convolution Revisited. 2.7 A Model from Other Models. 2.8 Stability. 2.9 Solution Using MATLAB Functions. 2.10 Summary. Problems. References. 3. Frequency-Domain Analysis. 3.1 Introduction. 3.2 Theory of Sampling. 3.3 DTFT and IDTFT. 3.4 DTFT of Unit Step Sequence. 3.5 Use of MATLAB to Compute DTFT. 3.6 DTFS and DFT. 3.7 Fast Fourier Transform. 3.8 Use of MATLAB to Compute DFT and IDFT. 3.9 Summary/ Problems. References. 4. Infinite Impulse Response Filters. 4.1 Introduction. 4.2 Magnitude Approximation of Analog Filters. 4.3 Analog Frequency Transformations. 4.4 Digital Filters. 4.5 Impulse-Invariant Transformation. 4.6 Bilinear Transformation. 4.7 Digital Spectral Transformation. 4.8 Allpass Filters. 4.9 IIR Filter Design Using MATLAB. 4.10 Yule-Walker Approximation. 4.11 Summary. Problems. References. 5. Finite Impulse Response Filters. 5.1 Introduction. 5.2 Linear Phase Fir Filters. 5.3 Fourier Series Method Modified by Windows. 5.4 Design of Windowed FIR Filter Using MATLAB. 5.5 Equiripple Linear Phase FIR Filters. 5.6 Design of Equiripple FIR Filters Using MATLAB. 5.7 Frequency Sampling Method. 5.8 Summary. Problems. References. 6. Filter Realizations. 6.1 Introduction. 6.2 FIR Filter Realizations. 6.3 IIR Filter Realizations. 6.4 Allpass Filters in Parallel. 6.5 Realization of FIR and IIR Filters Using MATLAB. 6.6 Summary. Problems. References. 7. Quantized Filter Analysis. 7.1 Introduction. 7.2 Filter Design-Analysis Tool. 7.3 Quantized Filter Analysis. 7.4 Binary Numbers and Arithmetic. 7.5 Quantization Analysis of IIR Filters. 7.6 Quantization Analyis of FIR Filters. 7.7 Summary. Problems. References. 8. Hardware Design Using DSP Chips. 8.1 Introduction. 8.2 Simulink and Real-Time Workshop. 8.3 Design Preliminaries. 8.4 Code Generation. 8.5 Code Composer Studio. 8.6 Simulator and Emulator. 8.7 Conclusion. References. 9. MATLAB Primer. 9.1 Introduction. 9.2 Signal Processing Toolbox. References. Index.

Journal ArticleDOI
M Zhong1, H Ye1, Peng Shi1, G Wang1
08 Jul 2005
TL;DR: In this article, the robust fault detection problem for a class of discrete-time linear Markovian jump systems with an unknown input is formulated as an H∞-filtering problem, in which the filter matrices are dependent on the system mode.
Abstract: The paper deals with the robust fault detection problem for a class of discrete-time linear Markovian jump systems with an unknown input. By using a general observer-based fault detection filter as residual generator, the robust fault detection filter design is formulated as an H∞-filtering problem, in which the filter matrices are dependent on the system mode, i.e. the residual generator is a Markovian jump linear system as well. The main objective is to make the error between residual and fault (or, more generally, weighted fault) as small as possible. A sufficient condition to solve this problem is established in terms of the feasibility of certain linear matrix inequalities (LMI), which can be solved with the aid of Matlab LMI Toolbox. A numerical example is given to illustrate the effectiveness of the proposed techniques.

Journal ArticleDOI
TL;DR: This paper presents a new fault detection and diagnosis (FDD) algorithm for general stochastic systems that uses the measured output probability density functions and the input of the system to construct a stable filter-based residual generator such that the fault can be detected and diagnosed.
Abstract: This paper presents a new fault detection and diagnosis (FDD) algorithm for general stochastic systems Different from the classical FDD design, the distribution of system output is supposed to be measured rather than the output signal itself The task of such an FDD algorithm design is to use the measured output probability density functions (PDFs) and the input of the system to construct a stable filter-based residual generator such that the fault can be detected and diagnosed For this purpose, square root B-spline expansions are applied to model the output PDFs and the concerned problem is transformed into a nonlinear FDD algorithm design subjected to a nonlinear weight dynamical system A linear matrix inequality based solution is presented such that the estimation error system is stable and the fault can be detected through a threshold Moreover, an adaptive fault diagnosis method is also provided to estimate the size of the fault Simulations are provided to show the efficiency of the proposed approach

Journal ArticleDOI
07 Nov 2005
TL;DR: In this article, the authors present new ideas for the design and implementation of microwave filters with single and dual stopbands, which can be realized with waveguide, coaxial, dielectric resonators, or in a planar technology.
Abstract: This paper presents new ideas for the design and implementation of microwave filters with single and dual stopbands. They can be realized with waveguide, coaxial, dielectric resonators, or in a planar technology. The new methods represent an advance over present methods in that the resonators are direct coupled, thus avoiding the need for transmission line phase lengths between resonator stubs that tend to degrade performance due to their dispersion and are difficult to adjust during tuning. Three bandstop (BS) configurations are presented. The first will accommodate even or odd characteristics and also asymmetric responses, although some negative or diagonal cross-couplings will be needed. The second resembles the cul-de-sac configuration for bandpass filters and needs no diagonal or negative couplings even for asymmetric characteristics. The third is an application of the cul-de-sac synthesis technique to dual-band bandstop (DBBS) filters. All these BS designs are very similar to regular bandpass filters in their design and realization. The design of a DBBS filter is presented and compared with an equivalent bandpass filter to demonstrate its advantages. Finally, the simulated and measured results of a fourth-degree BS filter design in the novel cul-de-sac configuration are presented.

Journal ArticleDOI
TL;DR: It is shown that the new filter outperforms the classical-order statistics filtering techniques and its performance is similar to FSVF, outperforming it in some cases.
Abstract: In this paper, the problem of impulsive noise reduction in multichannel images is addressed. A new filter is proposed on the basis of a recently introduced family of computationally attractive filters with a good detail-preserving ability (FSVF). FSVF is based on privileging the central pixel in each filtering window in order to replace it only when it is really noisy and preserve the original undistorted image structures. The new filter is based on a novel fuzzy metric and it is created by combining the mentioned scheme and the fuzzy metric. The use of the fuzzy metric makes the filter computationally simpler and it allows to adjust the privilege of the central pixel giving the filter an adaptive nature. Moreover, it is shown that the new filter outperforms the classical-order statistics filtering techniques and its performance is similar to FSVF, outperforming it in some cases.

Journal ArticleDOI
12 Jun 2005
TL;DR: The theory and practical implementation of a continuous-time LMS adaptive filter of the TX leakage in CDMA receivers are described, which achieved the maximum TXRR of 28 dB, which was limited by the reference signal coupling.
Abstract: The theory and practical implementation of a continuous-time LMS adaptive filter of the TX leakage in CDMA receivers are described. The filter works by injecting a matched out-of-phase copy of the TX leakage into the LNA output. It requires a reference signal coupled from the TX chain, whose I and Q components are appropriately scaled to generate the matched copy. The scale factors are the results of the correlation between the filter output signal and the I/Q components of the reference signal. The filter was designed as part of a 0.25-/spl mu/m CMOS cellular-band receiver. The effect of the DC offsets in the correlators on the TX leakage rejection ratio (TXRR) was minimized by using the sign-data variant of the LMS algorithm and by increasing the gain of the correlating multipliers. The loop stability margin was improved by swapping the I and Q reference inputs of the scaling multipliers. Without a significant group delay of the TX leakage relative to the reference signal, the filter achieved the maximum TXRR of 28 dB, which was limited by the reference signal coupling. The group delay introduced by the SAW duplexer reduced the minimum TXRR to 10.8 dB. The filter degraded the LNA noise factor and gain by 1.3 dB and 1.7 dB, respectively.

Journal ArticleDOI
TL;DR: In this paper, an all-optical bandpass microwave filter is implemented using an electrooptic phase modulator combined with a dispersive device to eliminate the baseband resonance of a typical low-pass filter.
Abstract: Theoretical analysis and experimental implementation of an all-optical bandpass microwave filter are presented. Bandpass filtering is implemented using an electrooptic phase modulator combined with a dispersive device to eliminate the baseband resonance of a typical low-pass filter. In addition to bandpass operation, the proposed filter also provides an improved mainlobe-to-sidelobe ratio (MSR) and a reduced mainlobe bandwidth compared with those of the conventional microwave filters with windowing. A four-tap bandpass microwave filter with a 3-dB mainlobe bandwidth of 2.65 GHz and an MSR of 30 dB is demonstrated. The filter performances, including the reconfigurability, tunability, and the dynamic range, are also discussed.

Proceedings ArticleDOI
31 Jul 2005
TL;DR: A new, single-pass nonlinear filter for edge-preserving smoothing and visual detail removal for N dimensional signals in computer graphics, image processing and computer vision applications built from two modified forms of Tomasi and Manduchi's bilateral filter.
Abstract: We present a new, single-pass nonlinear filter for edge-preserving smoothing and visual detail removal for N dimensional signals in computer graphics, image processing and computer vision applications. Built from two modified forms of Tomasi and Manduchi's bilateral filter, the new "trilateral" filter smoothes signals towards a sharply-bounded, piecewise-linear approximation. Unlike bilateral filters or anisotropic diffusion methods that smooth towards piecewise constant solutions, the trilateral filter provides stronger noise reduction and better outlier rejection in high-gradient regions, and it mimics the edge-limited smoothing behavior of shock-forming PDEs by region finding with a fast min-max stack. Yet the trilateral filter requires only one user-set parameter, filters an input signal in a single pass, and does not use an iterative solver as required by most PDE methods. Like the bilateral filter, the trilateral filter easily extends to N-dimensional signals, yet it also offers better performance for many visual applications including appearance-preserving contrast reduction problems for digital photography and denoising polygonal meshes.

Proceedings ArticleDOI
14 Nov 2005
TL;DR: This paper shows how zeroes can be imposed in the filters so that the iterated structure produces regular basis functions, and proposes a proposed design framework that yields filters that can be implemented efficiently through a lifting factorization.
Abstract: In this paper we study the nonsubsampled contourlet transform. We address the corresponding filter design problem using the Mc-Clellan transformation. We show how zeroes can be imposed in the filters so that the iterated structure produces regular basis functions. The proposed design framework yields filters that can be implemented efficiently through a lifting factorization. We apply the constructed transform in image noise removal where the results obtained are comparable to the state-of-the art, being superior in some cases.

Journal ArticleDOI
TL;DR: In this paper, a general simple technique for reducing any canonical n+2 coupling matrix to the more useful modular (i.e., cascaded N-tuplets) form is presented.
Abstract: In this paper, a general simple technique for reducing any canonical n+2 coupling matrix to the more useful modular (i.e., cascaded N-tuplets) form is presented. This is accomplished by performing a suitable sequence of matrix rotations whose angles are not derived through optimization, but are analytically computed from the starting coupling matrix elements and the transmission zeros frequencies. The proposed technique allows the association of a transmission zero to a specific block (i.e., a triplet or a quadruplet) increasing the degree of freedom in the filter design process.

Journal ArticleDOI
TL;DR: The work that led to what is now known as the Parks-McClellan algorithm is described, i.e., the Remez exchange algorithm with optimal Chebyshev approximation for FIR filter design.
Abstract: This article describes the work that led to what is now known as the Parks-McClellan algorithm. Within the bigger picture of filter design methods, this paper recount events that had an impact on the inspiration to develop the Parks-McClellan algorithm, i.e., the Remez exchange algorithm with optimal Chebyshev approximation for FIR filter design.

Proceedings ArticleDOI
18 Mar 2005
TL;DR: FPGA implementation results confirm that the proposed DA architecture can implement a 1024-tap FIR filter with significantly smaller area usage than the original LUT-based DA and the Lut-less DA-OBC.
Abstract: The paper presents a new memory-efficient distributed arithmetic (DA) architecture for high-order FIR filters. The proposed architecture is based on a memory reduction technique for DA look-up-tables (LUTs); it requires fewer transistors for high-order filters than original LUT-based DA, DA-offset binary coding (DA-OBC), and the LUT-less DA-OBC. Recursive iteration of the memory reduction technique significantly increases the maximum number of filter order implementable on an FPGA platform by not only saving transistor counts, but also balancing hardware usage between logic element (LE) and memory. FPGA implementation results confirm that the proposed DA architecture can implement a 1024-tap FIR filter with significantly smaller area usage (<50%) than the original LUT-based DA and the LUT-less DA-OBC.

Journal ArticleDOI
TL;DR: An extended family of cardinal splines is introduced-the generalized E-splines-to generalize the concept for all convolution operators with rational transfer functions, and shows how the formalism can be used to obtain exact, discrete implementations of analog filters.
Abstract: By interpreting the Green-function reproduction property of exponential splines in signal processing terms, we uncover a fundamental relation that connects the impulse responses of allpole analog filters to their discrete counterparts. The link is that the latter are the B-spline coefficients of the former (which happen to be exponential splines). Motivated by this observation, we introduce an extended family of cardinal splines-the generalized E-splines-to generalize the concept for all convolution operators with rational transfer functions. We construct the corresponding compactly supported B-spline basis functions, which are characterized by their poles and zeros, thereby establishing an interesting connection with analog filter design techniques. We investigate the properties of these new B-splines and present the corresponding signal processing calculus, which allows us to perform continuous-time operations, such as convolution, differential operators, and modulation, by simple application of the discrete version of these operators in the B-spline domain. In particular, we show how the formalism can be used to obtain exact, discrete implementations of analog filters. Finally, we apply our results to the design of hybrid signal processing systems that rely on digital filtering to compensate for the nonideal characteristics of real-world analog-to-digital (A-to-D) and D-to-A conversion systems.

Proceedings Article
30 Jul 2005
TL;DR: This work presents a self adaptive version of the particle filter that uses statistical methods to adapt the number of particles and the propagation function at each iteration and shows the advantages of the self adaptive filter by applying it to a synthetic example and to the visual tracking of targets in a real video sequence.
Abstract: The particle filter has emerged as a useful tool for problems requiring dynamic state estimation. The efficiency and accuracy of the filter depend mostly on the number of particles used in the estimation and on the propagation function used to reallocate these particles at each iteration. Both features are specified beforehand and are kept fixed in the regular implementation of the filter. In practice this may be highly inappropriate since it ignores errors in the models and the varying dynamics of the processes. This work presents a self adaptive version of the particle filter that uses statistical methods to adapt the number of particles and the propagation function at each iteration. Furthermore, our method presents similar computational load than the standard particle filter. We show the advantages of the self adaptive filter by applying it to a synthetic example and to the visual tracking of targets in a real video sequence.

Journal ArticleDOI
TL;DR: In this article, the authors generalized the iterated extended Kalman filter to solve a nonlinear smoothing problem for the current and past sample intervals using iterative numerical techniques, which is useful when nonlinearities might significantly degrade the accuracy or convergence reliability of other filters.
Abstract: The principle of the iterated extended Kalman filter has been generalized to create a new filter that has superior performance when the estimation problem contains severe nonlinearities. The new filter is useful when nonlinearities might significantly degrade the accuracy or convergence reliability of other filters. The new filter solves a nonlinear smoothing problem for the current and past sample intervals using iterative numerical techniques. This approach retains the nonlinearities of a fixed number of stages that precede the stage of interest, and it processes information from earlier stages in an approximate manner. The algorithm has been simulation tested on a difficult spacecraft attitude estimation problem that includes sensing of fewer than three axes and significant dynamic model uncertainty. The filter compensates for this uncertainty via simultaneous estimation of moment of inertia parameters. The new filter exhibits markedly better convergence reliability and accuracy than an extended Kalman filter and an unscented Kalman filter for estimation problems that start with large initial attitude or attitude rate errors.

Proceedings ArticleDOI
14 Nov 2005
TL;DR: A two-dimensional (2D) non-separable interpolation filter, which is calculated for each frame independently by minimising the prediction error energy, is developed in the context of prediction with fractional-pel motion vector resolution.
Abstract: In the context of prediction with fractional-pel motion vector resolution it was shown, that aliasing components contained in an image signal are limiting the prediction accuracy obtained by motion compensation. In order to consider aliasing, quantisation and motion estimation errors, camera noise, etc., we analytically developed a two-dimensional (2D) non-separable interpolation filter, which is calculated for each frame independently by minimising the prediction error energy. For every fractional-pel position to be interpolated, an individual set of 2D filter coefficients is determined. As a result, a coding gain of up to 1,2 dB for HDTV-sequences and up to 0,5 dB for CIF-sequences compared to the standard H.264/AVC is obtained.

Proceedings ArticleDOI
16 Jun 2005
TL;DR: In this article, the authors proposed a novel method for LCL type filter design, which makes the task very convenient and can be easily done by solving the equations by step-by-step design procedure, which is verified on the experimental set-up.
Abstract: LCL type filter becomes more and more attractive as utility interface for grid-connected voltage source rectifier (VSR). Compared to L type filter, LCL type filter can render better switching harmonics attenuation using lower inductance, which makes it suitable for higher power applications. However, LCL filter design is complex and needs to consider many constraints, such as current ripple through inductors, total impedance of the filter, switching harmonic attenuation, resonance phenomenon and reactive power absorbed by filter capacitors, etc. Try-error method is inconvenient and time-consuming. This paper proposes a novel method for LCL type filter design, which makes the task very convenient. At first, the total inductance should be determined according to current ripple requirement. With filter capacitor insertion, total inductance is split into two parts. A set of equations is obtained to represent the relationship between the impedances at switching frequency with consideration of switching harmonic attenuation and reactive power constrains. The other constraints are considered as the limitation for solvability condition for equations. So the overall design can be easily done by solving the equations. Step-by-step design procedure is described as a design example, which is verified on the experimental set-up

Journal ArticleDOI
TL;DR: The proposed sharpness dependent filter design based on the fairing of surface normal is superior to other approaches for smoothing a polygon mesh, as well as for preserving its sharp features.

Journal ArticleDOI
Rainer Storn1
TL;DR: In this paper, an alternative method for nonstandard filter design has been described, recasting the filter design problem as a minimization problem and solving the minimization via the DE minimizer.
Abstract: An alternative method for nonstandard filter design has been described. This method recasts the filter design problem as a minimization problem and solves the minimization via the DE minimizer, for which public domain software has been made available previously. The advantages of this method are its simplicity as well as the capability to design unconventional filter types. A great asset of this approach is that it can be applied with minimal knowledge of digital filter design theory.