Showing papers on "Filter design published in 2006"

A fast approximation of the bilateral filter using a signal processing approach

Abstract: The bilateral filter is a nonlinear filter that smoothes a signal while preserving strong edges. It has demonstrated great effectiveness for a variety of problems in computer vision and computer graphics, and a fast version has been proposed. Unfortunately, little is known about the accuracy of such acceleration. In this paper, we propose a new signal-processing analysis of the bilateral filter, which complements the recent studies that analyzed it as a PDE or as a robust statistics estimator. Importantly, this signal-processing perspective allows us to develop a novel bilateral filtering acceleration using a downsampling in space and intensity. This affords a principled expression of the accuracy in terms of bandwidth and sampling. The key to our analysis is to express the filter in a higher-dimensional space where the signal intensity is added to the original domain dimensions. The bilateral filter can then be expressed as simple linear convolutions in this augmented space followed by two simple nonlinearities. This allows us to derive simple criteria for downsampling the key operations and to achieve important acceleration of the bilateral filter. We show that, for the same running time, our method is significantly more accurate than previous acceleration techniques.

Design, Implementation, and Experimental Results of a Quaternion-Based Kalman Filter for Human Body Motion Tracking

Abstract: Real-time tracking of human body motion is an important technology in synthetic environments, robotics, and other human-computer interaction applications. This paper presents an extended Kalman filter designed for real-time estimation of the orientation of human limb segments. The filter processes data from small inertial/magnetic sensor modules containing triaxial angular rate sensors, accelerometers, and magnetometers. The filter represents rotation using quaternions rather than Euler angles or axis/angle pairs. Preprocessing of the acceleration and magnetometer measurements using the Quest algorithm produces a computed quaternion input for the filter. This preprocessing reduces the dimension of the state vector and makes the measurement equations linear. Real-time implementation and testing results of the quaternion-based Kalman filter are presented. Experimental results validate the filter design, and show the feasibility of using inertial/magnetic sensor modules for real-time human body motion tracking

Mode-Independent ${\cal H}_{\infty}$ Filters for Markovian Jump Linear Systems

Abstract: This note addresses the problem of Hinfin filtering for continuous-time linear systems with Markovian jumping parameters. The main contribution of the note is to provide a method for designing an asymptotically stable linear time-invariant Hinfin filter for systems where the jumping parameter is not accessible. The cases where the transition rate matrix of the Markov process is either exactly known, or unknown but belongs to a given polytope, are treated. The robust Hinfin filtering problem for systems with polytopic uncertain matrices is also considered and a filter design method based on a Lyapunov function that depends on the uncertain parameters is developed. The proposed filter designs are given in terms of linear matrix inequalities

A Dynamic Compressive Gammachirp Auditory Filterbank

Abstract: It is now common to use knowledge about human auditory processing in the development of audio signal processors. Until recently, however, such systems were limited by their linearity. The auditory filter system is known to be level-dependent as evidenced by psychophysical data on masking, compression, and two-tone suppression. However, there were no analysis/synthesis schemes with nonlinear filterbanks. This paper describe 18300060s such a scheme based on the compressive gammachirp (cGC) auditory filter. It was developed to extend the gammatone filter concept to accommodate the changes in psychophysical filter shape that are observed to occur with changes in stimulus level in simultaneous, tone-in-noise masking. In models of simultaneous noise masking, the temporal dynamics of the filtering can be ignored. Analysis/synthesis systems, however, are intended for use with speech sounds where the glottal cycle can be long with respect to auditory time constants, and so they require specification of the temporal dynamics of auditory filter. In this paper, we describe a fast-acting level control circuit for the cGC filter and show how psychophysical data involving two-tone suppression and compression can be used to estimate the parameter values for this dynamic version of the cGC filter (referred to as the "dcGC" filter). One important advantage of analysis/synthesis systems with a dcGC filterbank is that they can inherit previously refined signal processing algorithms developed with conventional short-time Fourier transforms (STFTs) and linear filterbanks

Influence of long-period filter cut-off on elastic spectral displacements

Abstract: The effect of the long-period filter cut-off, Tc, on elastic spectral displacements is investigated using a strong ground-motion database from Europe and the Middle East. The relation between the filter and oscillator responses is considered to observe the influence of Tc for both analogue and digital records, and the variations with site classification, magnitude, filter order and viscous damping. Robust statistics are derived using the re-processed European data to generalize the effects of the long-period filter cut-off on maximum oscillator deformation demands as a function of these seismological and structural features. Statistics with a 95% confidence interval are derived to suggest usable period ranges for spectral displacement computations as a function of Tc. The results indicate that the maximum period at which spectral displacements can be confidently calculated depend strongly on the site class, magnitude and filter order. The period range where reliable long-period information can be extracted from digital accelerograms is twice that of analogue records. Copyright © 2006 John Wiley & Sons, Ltd.

Differential Mode Input Filter Design for a Three-Phase Buck-Type PWM Rectifier Based on Modeling of the EMC Test Receiver

Abstract: For a three-phase buck-type pulsewidth modulation rectifier input stage of a high-power telecommunications power supply module, a differential-mode (DM) electromagnetic compatibility (EMC) filter is designed for compliance to CISPR 22 Class B in the frequency range of 150 kHz-30 MHz. The design is based on a harmonic analysis of the rectifier input current and a mathematical model of the measurement procedure including the line impedance stabilization network (LISN) and the test receiver. Guidelines for a successful filter design are given, and components for a 5-kW rectifier prototype are selected. Furthermore, formulas for the estimation of the quasi-peak detector output based on the LISN output voltage spectrum are provided. The damping of filter resonances is optimized for a given attenuation in order to facilitate a higher stability margin for system control. Furthermore, the dependence of the filter input and output impedances and the attenuation characteristic on the inner mains impedance are discussed. As experimentally verified by using a three-phase common-/Differential-Mode separator, this procedure allows accurate prediction of the converter DM conducted emission levels and therefore could be employed in the design process of the rectifier system to ensure compliance to relevant EMC standards

Optimal waveform design for UWB radios

Abstract: With transmit power spectra strictly limited by regulatory spectral masks, the emerging ultra-wideband (UWB) communication systems call for judicious pulse shape design in order to achieve optimal spectrum utilization, spectral mask compatibility, and coexistence with other wireless services. Meanwhile, orthogonal pulse sets are often desired in order to apply high-rate multidimensional modulation and (carrier-free) orthogonal frequency-division multiple access. Motivated by these considerations, we suggest a digital finite impulse response (FIR) filter approach to synthesizing UWB pulses and propose filter design techniques by which optimal waveforms that satisfy the spectral mask can be efficiently obtained. For single pulse design, we develop a convex formulation for the design of the FIR filter coefficients that maximize the spectrum utilization efficiency in terms of both the bandwidth and power allowed by the spectral mask. For orthogonal pulse design, a sequential strategy is derived to formulate the overall pulse design problem as a set of convex subproblems, which are then solved in a sequential manner to yield a set of mutually orthogonal pulses. Our design techniques not only provide waveforms with high spectrum utilization and guaranteed spectral mask compliance but also permit simple modifications that can accommodate several other system objectives.

Network-Based Robust $H_infty$ Filtering for Uncertain Linear Systems

Abstract: The problem of network-based robust Hinfin filtering for uncertain linear systems is investigated. Different from the design of the traditional filter, the effects of the network-induced delay and data dropout on the performance of a filtering-error system are considered. The derived criteria for Hinfin performance analysis of the filtering-error system and filter design are expressed as a set of linear matrix inequalities, which can be solved by using convex optimization method. Numerical examples show the effectiveness of the design method

An Improved H α Filter Design for Systems With Time-Varying Interval Delay

Abstract: This brief is concerned with Hinfin filter design for systems with time-varying interval delay (i.e., the time delay is varying in an interval). An appropriate type of Lyapunov functionals is proposed to investigate the delay-dependent Hinfin filter design problem. Improved delay-dependent results are presented by taking into account the interval range. Finally, a numerical example is given to demonstrate the effectiveness and the benefits of the proposed method

Robust /spl Hscr//sub /spl infin// filtering for discrete-time linear systems with uncertain time-varying parameters

Abstract: This paper deals with the problem of robust /spl Hscr//sub /spl infin// filtering for linear discrete-time state-space models with uncertain time-varying parameters. The parameters enter affinely into the state-space model matrices, and their admissible values and variations are assumed to belong to given intervals. A method is derived for designing a linear stationary asymptotically stable filter with a prescribed /spl Hscr//sub /spl infin// performance, in spite of large parameter uncertainty. The proposed method incorporates information on available bounds on both the admissible values and variation of the uncertain parameters and is based on a Lyapunov function with quadratic dependence on the parameters. The filter design is given in terms of linear matrix inequalities.

Fixed-order robust H/sub /spl infin// filter design for Markovian jump systems with uncertain switching probabilities

Abstract: This paper discusses the fixed-order robust H/sub /spl infin// filtering problem for a class of Markovian jump linear systems with uncertain switching probabilities. The uncertainties under consideration are assumed to be norm-bounded in the system matrices and to be elementwise bounded in the mode transition rate matrix, respectively. First, a criterion based on linear matrix inequalities is provided for testing the H/sub /spl infin// filtering level of a filter over all the admissible uncertainties. Then, a sufficient condition for the existence of the fixed-order robust H/sub /spl infin// filters is established in terms of the solvability of a set of linear matrix inequalities with equality constraints. To determine the filter, a globally convergent algorithm involving convex optimization is suggested. Finally, a numerical example is used to illustrate that the developed theory is more effective than the existing results.

Methods and apparatus for targeted sound detection

Abstract: Targeted sound detection methods and apparatus are disclosed. A microphone array has two or more microphones M0 . . . MM. Each microphone is coupled to a plurality of filters. The filters are configured to filter input signals corresponding to sounds detected by the microphones thereby generating a filtered output. One or more sets of filter parameters for the plurality of filters are pre-calibrated to determine one or more corresponding pre-calibrated listening zones. Each set of filter parameters is selected to detect portions of the input signals corresponding to sounds originating within a given listening zone and filter out sounds originating outside the given listening zone. A particular pre-calibrated listening zone is selected at a runtime by applying to the plurality of filters a set of filter coefficients corresponding to the particular pre-calibrated listening zone. As a result, the microphone array may detect sounds originating within the particular listening sector and filter out sounds originating outside the particular listening zone.

Analysis and design of high-performance asynchronous sigma-delta Modulators with a binary quantizer

Abstract: Asynchronous sigma-delta modulators (ASDMs) are closed-loop nonlinear systems that transform the information in the amplitude of their input signal into time information in the output signal, without suffering from quantization noise such as in synchronous sigma-delta modulators. This is an important advantage with many interesting applications. In contrast with their synchronous counterparts, ASDMs have been underexposed. Both conceptually and analytically, they are quite complex. This paper investigates in detail the analysis, design and circuit-implementation aspects of ASDMs with a binary quantizer. In the ASDM, the amplitude-time transformation is done using an inherent self-oscillation denoted as a limit cycle. The oscillation frequency is addressed as the main design parameter that determines the spectral properties of the ASDMs and the quality of the amplitude-time transformation. Analytical and graphical derivations of the limit cycle frequency are treated. The impact of the filter order and the properties of the nonlinear element are elaborated on. Circuit implementations and the tradeoffs in the design are presented for a first- and a second-order ASDM that target the VDSL front-end specifications. Prototypes are implemented in a digital 0.18-/spl mu/m 1.8-V CMOS technology. The measured SFDR is 75dB in a frequency band of 8MHz for the first-order ASDM, and 72dB in a band of 12MHz for the second-order ASDM. The dissipated power is 1.5 mW and 2.2 mW, respectively.

Active noise-reduction control apparatus and method

Abstract: Apparatus includes generator generating reference signal based on noise emitted from sound source, detector detecting level of reference signal and change in level, unit comparing change with threshold-value range and produce compared result, filter filtering reference signal, adaptive filter having variable filter coefficient, unit updating filter coefficient according to change of level of reference signal for obtaining an updated filter coefficient, unit stopping updating of filter coefficient in response to compared result when change falls outside threshold-value range, unit storing updated filter coefficient each time filter coefficient is updated, generator generating control signal using stored filter coefficient, unit generating control sound based on control signal, microphone detecting synthesis sound pressure of control sound and noise to produce an error signal, and unit setting stored filter coefficient to more accurate coefficient than stored filter coefficient based on error signal, and signal acquired by filtering control signal through filter.

Noniterative WLS design of allpass variable fractional-delay digital filters

Abstract: This paper presents a noniterative weighted-least-squares (WLS) method for designing allpass variable fractional-delay (VFD) digital filters. After expressing each coefficient of an allpass VFD filter as a polynomial of the VFD parameter p, we develop a noniterative technique for finding the optimal polynomial coefficients, and show that the allpass VFD filter design problem can be efficiently solved without using any iterative procedure while a closed-form solution can be easily obtained through solving a matrix equation. Compared with the existing iterative WLS method that solves a series of approximately linearized WLS minimization problems, the proposed noniterative one can yield much better design results with significantly reduced computational complexity. Moreover, the new WLS method does not involve any convergence issue.

Harmonic Voltage Reduction Using a Series Active Filter Under Different Load Conditions

Abstract: This paper proposes a series active filter using a simple control technique. The series active filter is applied as a controlled voltage source contrary to its common usage as variable impedance. It reduces the terminal harmonic voltages, supplying linear or even nonlinear loads with a good quality voltage waveform. The operation principle, control strategy, and theoretical analysis of the active filter are presented. These aspects were proven by the results of numerical simulations. Experimental results of the series active filter demonstrated its good performance under different load conditions

An LTCC balanced-to-unbalanced extracted-pole bandpass filter with complex load

Abstract: In this paper, the fundamental characteristics of a novel third-order RF balanced-to-unbalanced filter, namely, a balun filter, for integrated RF module applications are presented. This center-tapped transformer-based new device works concurrently as a balun, an extracted-pole bandpass filter, and a matching network. As coupled resonant tanks are employed to perform the balun type of operation, traditional coupled-resonator filter theory can thus be used to design and analyze such a new device. Moreover, an extracted-pole technique is used not only for creating a transmission zero, but also provides a capability to match the filter with a complex load. In addition to providing a simple design procedure for the device, its working mechanism is also revealed mathematically. Specifically, return-loss sensitivity with respect to each resonator admittance and complex load matching capability are discussed in details. This balun filter has been implemented in a multilayered low-temperature co-fired ceramic substrate, demonstrating its promising potentials in miniaturized RF front-end modules. Experimental measurements are also presented to validate the theory and computer simulations.

Method and apparatus for enhanced video coding

Abstract: Standard video compression techniques apply motion-compensated prediction combined with transform coding of the prediction error 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 efficiency obtained by motion compensation In order to consider aliasing, quantization and motion estimation errors, camera noise, etc, we analytically developed a twodimensional (2D) non-separable interpolation filter, which is independently calculated for each frame by minimizing the prediction error energy For every fractional-pel position to be interpolated, an individual set of 2D filter coefficients is determined Since transmitting filter coefficients as side information results in an additional bit rate, which is almost constant for different image resolutions and total bit rates, the loss in coding gain increases when total bit rates sink Therefore, we developed an algorithm, which regards the non-separable two-dimensional filter as a polyphase filter For each frame, predicting the interpolation filter impulse response through evaluation of the polyphase filter, we only have to encode the prediction error of the filter coefficients

Independent component analysis based filter design for defect detection in low-contrast textured images

Abstract: In this paper, we propose a convolution filtering scheme for detecting defects in low-contrast textured surface images and, especially, focus on the application for glass substrates in liquid crystal display (LCD) manufacturing. A defect embedded in a low-contrast surface image shows no distinct intensity from its surrounding region, and even worse, the sensed image may present uneven brightness on the surface. All these make the defect detection in low-contrast surface images extremely difficult. In this study, a constrained ICA (independent component analysis) model is proposed to design an optimal filter with the objective that the convolution filter will generate the most representative source intensity of the background surface without noise. The prior constraint incorporated in the ICA model confines the source values of all training image patches of a defect-free image within a small interval of control limits. In the inspection process, the same control parameter used in the constraint is also applied to set up the thresholds that make impulse responses of all pixels in faultless regions within the control limits, and those in defective regions outside the control limits. A stochastic evolutionary computation algorithm, particle swarm optimization (PSO), is applied to solve for the constrained ICA model. Experimental results have shown that the proposed method can effectively detect defects in textured LCD glass substrate images

Steerable frequency-invariant beamforming for arbitrary arrays

Abstract: Frequency-invariant beamforming aims to parameterize array filter coefficients such that the spectral and spatial response profiles of the array can be adjusted independently. Solutions to this problem have been presented for specific sensor configurations often requiring a larger number of sensors. However, in practical applications, the number and location of sensors are often restricted. This paper proposes to find an optimal linear basis transformation that decouples the frequency response from the spatial response. A least-squares optimal basis transform can be computed numerically for arbitrary sensor configurations, for which typically no exact analytical solutions are available. This transform can be further combined with a spherical harmonics basis resulting in readily steerable broadband beams. This solution to broadband beamforming effectively decouples the array geometry from the steering geometry. Furthermore, for frequency-invariant beams, this approach results in a significant reduction in the number of beam-design parameters. Here, the method is demonstrated for an optimal design of far-field response for an irregular linear array with as few as three sensors.

Attitude Estimation from Vector Observations Using a Genetic-Algorithm-Embedded Quaternion Particle Filter

Abstract: A novel algorithm is presented for the estimation of spacecraft attitude quaternion from vector observations in gyro-equipped spacecraft The new estimator is a particle filter that uses approximate numerical representation techniques for performing the otherwise exact time propagation and measurement update of potentially non-Gaussian probability density functions in inherently nonlinear systems The new method can be applied using various kinds of vector observations In this paper, the case of a low-Earth-orbit spacecraft, acquiring noisy geomagnetic field measurements via a three-axis magnetometer, is considered A genetic algorithm is used to estimate the gyro bias parameters, avoiding the need to augment the particle filter's state and rendering the estimator computationally efficient Contrary to conventional filters, which address the quaternion's unit norm constraint via special (mostly ad hoc) techniques, the new filter maintains this constraint naturally An extensive simulation study is used to compare the new filter to three extended Kalman filters and to the unscented Kalman filter in Gaussian and non-Gaussian scenarios The new algorithm is shown to be robust with respect to initial conditions and to possess a fast convergence rate An evaluation of the Cramer-Rao estimation error lower bound demonstrates the filter's asymptotic statistical efficiency and optimality

Two-dimensional iir filter design with modern search heuristics: a comparative study

Abstract: In the past few years, there has been a massive growth in the field of biologically inspired global search heuristics. Computational cost having been reduced almost dramatically, researchers from all corners are taking more interset in following the underlying principles of nature to solve nearly intractable search problems. In this paper, we attempt to solve one very important optimization problem arising in the field of two-dimensional IIR (infinite impulse response) filter design, with three naturally inspired global search algorithms. We have used a state-of-the-art real coded genetic algorithm (GA), one very recent and modified version of the particle swarm opimization (PSO) and finally an improved version of the differential evolution (DE) algorithm. The DE algorithm has been modified by us to prevent its premature convergence to some suboptimal region of the search space. The design task is formulated as a constrained minimization problem and solved by the three metaheuristics. Numerical results are presented over three difficult instances of the design problem. The study also compares the results with two recently published filter design methods. Our experiments reveal that the DE family of algorithms should receive primary attention in solving the constrained multidimensional filter design tasks.

Reconstruction of nonuniformly sampled bandlimited signals by means of time-varying discrete-time FIR filters

Abstract: This paper deals with reconstruction of nonuniformly sampled bandlimited continuous-time signals using time-varying discrete-time finite-length impulse response (FIR) filters. The main theme of the paper is to show how a slight oversampling should be utilized for designing the reconstruction filters in a proper manner. Based on a time-frequency function, it is shown that the reconstruction problem can be posed as one that resembles an ordinary filter design problem, both for deterministic signals and random processes. From this facts, an analytic least-square design technique is then derived. Furthermore, for an important special case, corresponding to periodic nonuniform sampling, it is shown that the reconstruction problem alternatively can be posed as a filter bank design problem, thus with requirements on a distortion transfer function and a number of aliasing transfer functions. This eases the design and offers alternative practical design methods as discussed in the paper. Several design examples are included that illustrate the benefits of the proposed design techniques over previously existing techniques.