Showing papers on "Filter design published in 2001"

Noise adaptive soft-switching median filter

Abstract: Existing state-of-the-art switching-based median filters are commonly found to be nonadaptive to noise density variations and prone to misclassifying pixel characteristics at high noise density interference. This reveals the critical need of having a sophisticated switching scheme and an adaptive weighted median filter. We propose a novel switching-based median filter with incorporation of fuzzy-set concept, called the noise adaptive soft-switching median (NASM) filter, to achieve much improved filtering performance in terms of effectiveness in removing impulse noise while preserving signal details and robustness in combating noise density variations. The proposed NASM filter consists of two stages. A soft-switching noise-detection scheme is developed to classify each pixel to be uncorrupted pixel, isolated impulse noise, nonisolated impulse noise or image object's edge pixel. "No filtering" (or identity filter), standard median (SM) filter or our developed fuzzy weighted median (FWM) filter will then be employed according to the respective characteristic type identified. Experimental results show that our NASM filter impressively outperforms other techniques by achieving fairly close performance to that of ideal-switching median filter across a wide range of noise densities, ranging from 10% to 70%.

Theory and Design of Microwave Filters

Abstract: This is a thorough, graduate-level text which provides a single source for filter design including basic circuit theory, network synthesis and the design of a variety of microwave filter structures. The aim is to present design theories followed by specific examples with numerical simulations of the designs, with pictures of real devices wherever possible. The book is aimed at designers, engineers and researchers working in microwave electronics who need to design or specify filters.

Stochastic Methods for Sequential Data Assimilation in Strongly Nonlinear Systems

Abstract: This paper considers several filtering methods of stochastic nature, based on Monte Carlo drawing, for the sequential data assimilation in nonlinear models. They include some known methods such as the particle filter and the ensemble Kalman filter and some others introduced by the author: the second-order ensemble Kalman filter and the singular extended interpolated filter. The aim is to study their behavior in the simple nonlinear chaotic Lorenz system, in the hope of getting some insight into more complex models. It is seen that these filters perform satisfactory, but the new filters introduced have the advantage of being less costly. This is achieved through the concept of second-order-exact drawing and the selective error correction, parallel to the tangent space of the attractor of the system (which is of low dimension). Also introduced is the use of a forgetting factor, which could enhance significantly the filter stability in this nonlinear context.

Specification and design of a prototype filter for filter bank based multicarrier transmission

Abstract: The specifications of filter banks for multicarrier transmission systems with a large number of subchannels are discussed, with application to xDSL and power line communication in mind. The near perfect reconstruction (PR) modulated approach is considered and the importance, for the system, of the prototype filter delay is stressed. An existing design technique known to be particularly relevant to the context is revisited from a frequency sampling perspective. The performance results in terms of subchannel noise floor and delay are given for several filter lengths and an experimental validation is provided. Finally, an improvement to the design technique is proposed, which brings a gain of 3.3 dB in subchannel interference power level.

Space variant median filters for the restoration of impulse noise corrupted images

Abstract: This brief proposes a generalized framework of median based switching schemes, called multi-state median (MSM) filter. By using a simple thresholding logic, the output of the MSM filter is adaptively switched among those of a group of center weighted median (CWM) filters that have different center weights. As a result, the MSM filter is equivalent to an adaptive CWM filter with a space varying center weight which is dependent on local signal statistics. The efficacy of the proposed filter has been evaluated by extensive simulations.

Filtering of laser altimetry data using a slope adaptive filter

Abstract: A point set obtained by laser altimetry represents points from not only the ground surface but also objects found on it. For civil works applications points representing the surface of non-ground objects have to be removed from the point set in a filtering process. This paper describes modifications made to an existing “slope based” filtering algorithm, and presents some results obtained from the use of the filter. The “slope based” filter operates on the assumption that terrain slopes do not rise above a certain threshold, and that features in the data that have slopes above this threshold do not belong to the natural terrain surface. However, this assumption limits the use of the filter to terrain with gentle slopes. To overcome this limitation, the filter was modified in manner that the threshold varies with respect to the slope of the terrain. The results of tests carried out using the modified filter confirm that the modification reduces the number of Type I errors (ground points in steep terrain are not filtered off). Further numerical comparison of the filter output with a reference data set for the same site (obtained photogrammetrically) show that the filter generates relatively minimal Type II errors. The output of the modified slope filter was also compared with the output from a filtering found in the commercial software package, “Terrascan”.

Robust H/spl infin/ filter design for uncertain linear systems with multiple time-varying state delays

Abstract: The problem of robust H/spl infin/ filtering for continuous-time uncertain linear systems with multiple time-varying delays in the state variables is investigated. The uncertain parameters are supposed to belong to a given convex bounded polyhedral domain. The aim is to design a stable linear filter assuring asymptotic stability and a prescribed H/spl infin/ performance level for the filtering error system, irrespective of the uncertainties and the time delays. Sufficient conditions for the existence of such a filter are established in terms of linear matrix inequalities, which can be efficiently solved by means of powerful convex programming tools with global convergence assured. An example illustrates the proposed methodology.

Amplifier measurement and modeling processes for use in generating predistortion parameters

Abstract: A wideband predistortion system compensates for a nonlinear amplifier's frequency and time dependent AM-AM and AM-PM distortion characteristics. The system comprises a data structure in which each element stores a set of compensation parameters (preferably including FIR filter coefficients) for predistorting the wideband input transmission signal. The parameter sets are preferably indexed within the data structure according to multiple signal characteristics, such as instantaneous amplitude and integrated signal envelope, each of which corresponds to a respective dimension of the data structure. To predistort the input transmission signal, an addressing circuit digitally generates a set of data structure indices from the input transmission signal, and the indexed set of compensation parameters is loaded into a compensation circuit which digitally predistorts the input transmission signal. This process of loading new compensation parameters into the compensation circuit is preferably repeated every sample instant, so that the predistortion function varies from sample-to-sample. The sets of compensation parameters are generated periodically and written to the data structure by an adaptive processing component that performs a non-real-time analysis of amplifier input and output signals. The adaptive processing component also implements various system identification processes for measuring the characteristics of the power amplifier and generating initial sets of filter coefficients. In an antenna array embodiment, a single adaptive processing component generates the compensation parameters sets for each of multiple amplification chains on a time-shared basis. In an embodiment in which the amplification chain includes multiple nonlinear amplifiers that can be individually controlled (e.g., turned ON and OFF) to conserve power, the data structure separately stores compensation parameter sets for each operating state of the amplification chain.

Efficient Implementation of Quaternion Fourier Transform, Convolution, and Correlation by 2-D

Abstract: The recently developed concepts of quaternion Fourier transform (QFT), quaternion convolution (QCV), and quaternion correlation, which are based on quaternion algebra, have been found to be useful for color image processing. However, the necessary computational algorithms and their complexity still need some attention. In this paper, we will develop efficient algorithms for QFT, QCV, and quaternion correlation. The conventional complex two-dimensional (2-D) Fourier transform (FT) is used to implement these quaternion operations very efficiently. By these algorithms, we only need two complex 2-D FTs to implement a QFT, six complex 2-D FTs to implement a one-side QCV or a quaternion correlation and 12 complex 2-D FTs to implement a two-side QCV, and the efficiency of these quaternion operations is much improved. Meanwhile, we also discuss two additional topics. The first one is about how to use QFT and QCV for quaternion linear time-invariant (QLTI) system analysis. This topic is important for quaternion filter design and color image processing. Besides, we also develop the spectrum-product QCV. It is the improvement of the conventional form of QCV. For any arbitrary input functions, it always corresponds to the product operation in the frequency domain. It will be very useful for quaternion filter design.

Filtering of multichannel SAR images

Abstract: An explicit form of the linear multichannel synthetic aperture radar (SAR) intensity filter, which preserves radiometry while optimally reducing speckle is derived, together with a compact expression for the theoretical gain in equivalent numbers of looks (ENLs). The filter can be applied to mixed data types, which is demonstrated using a combination of ERS and JERS satellite data, and confirms the filter performance predicted by the theory. Tests indicate that a simplified form of the filter, which neglects correlation between images, gives an ENL only slightly less than optimal, while being much easier to implement. Exact analysis of the effect of estimating filter weights shows that the linear increase in ENL with the number of images predicted for the ideal filter does not occur. In practice, the ENL is affected by the window size used to estimate the weights and saturates as the number of images increases. An efficient recursive form of the filter is described, which is most naturally applied to multitemporal data for the practically important case where the current image is uncorrelated with previous images in a data sequence.

Fast impulsive noise removal

Abstract: A generic n-dimensional filter with the primary purpose of eliminating impulsive-like noise is presented. This recursive nonlinear filter is composed of two conditional rules, which are applied independently, in any order, one after the other. It identifies noisy items by inspection of their surrounding neighborhood, and afterwards it replaces their values with the most "conservative" ones out of their neighbors' values. In this way, no new values are introduced and the histogram distribution range is conserved. This n-dimensional filter can be decomposed recursively to a lower dimensional space, each time generating two sets of n(n-1)-dimensional filters. This study, which focuses on the case of two-dimensional signals (gray scale images), explores one possible implementation of this new filter and orients the evaluation of its performance toward the median filter, as this filter is the basis of many more sophisticated filters for impulsive noise reduction. Tests were carried out using both real and artificial images. We found this new filter to be much faster than the median filter while performing comparably in terms of both image information conservation and noise reduction, which suggests that it could replace the median filter for the preliminary processing included in state-of-the-art noise removal filters. This new filter should either eliminate or attenuate most noisy pixels in synthetic and natural images not excessively contaminated. It has a slight smoothing effect on nonnoisy image regions. In addition, it is scalable, easily implemented, and adaptable to specific applications.

Dual-mode stepped-impedance ring resonator for bandpass filter applications

Abstract: It is well known that two orthogonal resonant modes exist within a one-wavelength ring resonator. In this paper, we focus on a ring resonator possessing an impedance step as a form of perturbation. A convenient analyzing method for obtaining the resonance characteristics of this resonator structure is presented. Furthermore, generation of attenuation poles obtained by the dual-mode ring resonator is discussed. In addition, a filter design method based on this resonator is explained, followed by experimental results, which prove the validity of the proposed design method.

Adaptive filtering algorithms with selective partial updates

Abstract: In some applications of adaptive filtering such as active noise reduction, and network and acoustic echo cancellation, the adaptive filter may be required to have a large number of coefficients in order to model the unknown physical medium with sufficient accuracy. The computational complexity of adaptation algorithms is proportional to the number of filter coefficients. This implies that, for long adaptive filters, the adaptation task can become prohibitively expensive, ruling out cost-effective implementation on digital signal processors. The purpose of partial coefficient updates is to reduce the computational complexity of an adaptive filter by adapting a block of the filter coefficients rather than the entire filter at every iteration. In this paper, we develop a selective-partial-update normalized least-mean-square (NLMS) algorithm, and analyze its stability using the traditional independence assumptions and error-energy bounds. Selective partial updating is also extended to the affine projection (AP) algorithm by introducing multiple constraints. The new algorithms appear to have good convergence performance as attested to by computer simulations with real speech signals.

Discretization-free design of variable fractional-delay FIR digital filters

Abstract: This paper presents a closed-form solution for obtaining the optimal coefficients of variable finite impulse-response (FIR) filters with continuously adjustable fractional-delay (FD) response. The design is formulated as a weighted-least-squares (WLS) approximation problem without discretizing (sampling) the frequency /spl omega/ and the variable FD p in the filter design process, and the objective error function of variable frequency response can be derived by numerical integration, thus the variable FD filter coefficients can be obtained in a closed-form. Compared to the existing WLS method, since the discretization-free method does not need parameter discretizations in deriving the objective error function, the closed-form solution is optimal in the sense that the filter design accuracy is not affected by the sampling grid densities, and higher design accuracy can be guaranteed. Furthermore, since the discretization-free method does not need to sum up all the squared errors at a great number of discrete points when evaluating the objective error function, the computational complexity can be reduced considerably.

A new H/sub /spl infin// filter design for linear time delay systems

Abstract: A new delay-dependent H/sub /spl infin// filtering design is proposed for linear, continuous, time-invariant systems with time delay. The obtained filter is of the Luenberger observer type. The design guarantees that the H/sub /spl infin//-norm of the system, relating the exogenous signals to the estimation error, is less than a prescribed level. The filter is based on the application of a newly derived version of the bounded real lemma for time-delayed systems. This novel approach is compared, via an example, with another solution that appears in the literature.

Microphone array apparatus

Abstract: A microphone array apparatus includes a microphone array including microphones, one of the microphones being a reference microphone, filters receiving output signals of the microphones, and a filter coefficient calculator which receives the output signals of the microphones, a noise and a residual signal obtained by subtracting filtered output signals of the microphones other than the reference microphone from a filtered output signal of the reference microphone and which obtain filter coefficients of the filters in accordance with an evaluation function based on the residual signal.

Two-planar vena cava filter with self-centering capabilities

Abstract: A self-centering filter is useful for trapping blood clots, reducing their size and arresting their further migration from the vena cava into pulmonary circulation. The two-planar filter design is formed from a conical array of filter wires wherein two sets of filter wires are included, each set differing in length from the other. The filter wires diverge from a common apex at one end and extend radially outwardly therefrom to an attachement end attachable to the wall of a blood vessel such that the attachement ends of the first set of wires are anchored at a location spaced apart from the location of anchoring of the attachement ends of the second set, effecting a two-planar filter design. This establishes a single filtering element having two planes of contact with the vein wall which also provides for centering of the filtering element. At least one of the sets of wires contains barbs, e.g. hooks, for anchoring the filter to the inner venal wall.

Interference management for CDMA systems through power control, multiuser detection, and beamforming

Abstract: Among the ambitious challenges to be met by the third-generation systems is to provide high-capacity flexible services. Code-division multiple access (CDMA) emerges as a promising candidate to meet these challenges. It is well known that CDMA systems are interference-limited, and interference management is needed to maximally utilize the potential gains of this access scheme. Several methods of controlling and/or suppressing the interference through power control, multiuser detection (temporal filtering), and receiver beamforming (spatial filtering) have been proposed to increase the capacity of CDMA systems up to date. We investigate the capacity increase that is possible by combining power control with intelligent temporal and spatial receiver filter design. The signal-to-interference ratio maximizing joint temporal-spatial receiver filters in unconstrained and constrained filter spaces are derived. Two-step iterative power control algorithms that converge to the optimum powers and the joint temporal and spatial receiver filters in the corresponding filter domains are given. A power control algorithm with a less complex filter update procedure is also given. We observe that significant savings in total transmit power are possible if filtering in both domains is utilized compared with conventional power control and joint optimal power control and filtering in only one domain.

New approaches to robust minimum variance filter design

Abstract: This paper is concerned with the design of robust filters that ensure minimum filtering error variance bounds for discrete-time systems with parametric uncertainty residing in a polytope. Two efficient methods for robust Kalman filter design are introduced. The first utilizes a recently introduced relaxation of the quadratic stability requirement of the stationary filter design. The second applies the new method of recursively solving a semidefinite program (SDP) subject to linear matrix inequalities (LMIs) constraints to obtain a robust finite horizon time-varying filter. The proposed design techniques are compared with other existing methods. It is shown, via two examples, that the results obtained by the new methods outperform all of the other designs.

A passive series, active shunt filter for high power applications

Abstract: This paper presents a hybrid series passive/shunt active power filter system for high power nonlinear loads. This work is motivated by the fact that the ability of a converter to perform effectively as an active filter is limited by the power and the frequency distribution of the distortion for which it must compensate. This system is comprised of a three-phase shunt active filter and series AC line smoothing reactance installed in front of the target load. The proposed system significantly reduces the required shunt active filter bandwidth. The space-vector pulse width modulation (PWM) controller is based on a dead-beat control model. It is implemented digitally using a single 16-bit microcontroller. This controller requires only the supply current to be monitored, an approach different from conventional methods. The paper provides background on the operation of the filter, the details of the power circuit, the details of the control design, representative waveforms, and spectral performance for a filter which supports a 15 kVA phase controlled rectifier load. Experimental data indicate that the active filter typically consumes 2% or less of the average load power, suggesting that a parallel filter is an efficient compensation approach. The spectral performance shows that the active filter brings the system into compliance with IEEE519-1992 up to the 33rd harmonic for an AC line smoothing reactance of 0.13 p.u.

Speech recognition using microphone antenna array

Abstract: A system and method of audio processing provides enhanced speech recognition. Audio input is received at a plurality of microphones. The multi-channel audio signal from the microphones may be processed by a beamforming network to generate a single-channel enhanced audio signal, on which voice activity is detected. Audio signals from the microphones are additionally processed by an adaptable noise cancellation filter having variable filter coefficients to generate a noise-suppressed audio signal. The variable filter coefficients are updated during periods of voice inactivity. A speech recognition engine may apply a speech recognition algorithm to the noise-suppressed audio signal and generate an appropriate output. The operation of the speech recognition engine and the adaptable noise cancellation filter may advantageously be controlled based on voice activity detected in the single-channel enhanced audio signal from the beamforming network.

Automated filter tuning using generalized low-pass prototype networks and gradient-based parameter extraction

Abstract: A novel technique for automated filter tuning is introduced. The filter to be tuned is represented by a generalized filter low-pass prototype model rather than a specialized equivalent network. The prototype model is based on the minimum number of characteristic filter parameters to represent the filter transfer function correctly. The parameter values are found from a gradient-based parameter-extraction process using measured S-parameters. Automated filter tuning is performed as a two-step procedure. First, the parameter sensitivities with respect to the tuning elements are determined by a series of S-parameter measurements. Second, the parameter values of the filter are compared to the values of the ideal filter prototype found from a filter synthesis, thus yielding the optimal screw positions. This novel tuning technique has been tested successfully with direct coupled three-resonator and cross-coupled four- and six-resonator filters.

A systematic algorithm for the design of multiplierless FIR filters

Abstract: A systematic algorithm is proposed for designing multiplierless finite-impulse response (FIR) filters This algorithm minimizes the number of adders required to implement the overall filter to meet the given amplitude criteria The optimization is performed in two basic steps First, a linear programming algorithm is used for determining a parameter space of the infinite-precision coefficients including the feasible space where the filter meets the given amplitude specifications The second step involves finding the filter parameters in this space such that the resulting filter meets the given criteria with the simplest coefficient representation forms The efficiency of the proposed algorithm is illustrated by means of several examples taken from the literature

Design considerations on the line-side filter used in the dynamic voltage restorer

Abstract: The effects of harmonic voltages generated by the voltage source inverter within a dynamic voltage restorer can be mitigated through the use of a line-side filter. However, injudicious design of the harmonic-filtering system can degrade the power-quality-enhancement capability of the restorer. Inappropriate filter design may also result in the need for the inverter current rating to be increased significantly, The purpose of the investigation is therefore to carry out a systematic analysis of the filter design by considering first the selection of the filter-capacitor on the basis of meeting a prespecified harmonic-performance criterion. The effect of the capacitor on the inverter rating is then evaluated and the method of determining the filter capacity is presented. An example, which involves electromagnetic transient studies, is included to demonstrate the efficacy of the proposed design method.

Efficient FIR filter for high-speed communication

Abstract: A method for reducing a propagation delay of a digital filter. The digital filter has an input path and an output path and includes a set of delay elements and a number of taps. The taps couples the input path to the output path. Each of the taps includes a coefficient, a multiplier and an adder. Each of the delay elements is disposed between two adjacent taps. The delay elements are placed in both the input path and the output path of the digital filter, such that the digital filter has fewer delay elements in the input path than a direct-form digital filter having the same number of taps in a direct-form structure and has fewer delay elements in the output path than a transposed-form digital filter having the same number of taps in a transposed-form structure, and such that the digital filter has same transfer function as the direct-form digital filter and the transposed-form digital filter.

A Simple Method to Design Wideband Electronically Tunable Combline Filters

Abstract: A new systematic approach for designing wide- band tunable combline filters is presented. New results on tunable combline filter theory are proposed and explicit de- sign formulas, to obtain the filter design parameters from specifications, are included. These design parameters are: center frequency, resonator electrical length, instantaneous bandwidth and tuning capacitance. The proposed design technique is used to construct an X-band wideband mi- crostrip tunable filter from 8.0 GHz to 12.0 GHz with com- mercial GaAs FETs as tuning elements. Parasitic effects and simulation problems are also discussed.

A 2.7-V CMOS dual-mode baseband filter for PDC and WCDMA

Abstract: A fifth-order analog CMOS RC-opamp baseband filter for a dual-mode cellular phone receiver was designed with maximum component sharing in the two modes, The filter meets the bandwidth specifications of both the PDC and WCDMA standards, which represent the two extremes with respect of the channel bandwidth. The total area of 4.8 mm/sup 2/ was minimized by reducing the filter order from five to three in the PDC mode, Also, the operational amplifiers with adjustable GBW were used to minimize PDC-mode power consumption. The capacitance matrices were made only partially overlapping to reduce the resistance spread, The largest resistors were implemented with T networks and the smallest capacitors with series connections to extend the range of feasible passive component values. The measured integrated input referred noise is 17 /spl mu/V and 47 /spl mu/V in the PDC and WCDMA modes, respectively. The IIP3 is +35 dBV in the WCDMA mode, and the circuit consumes 6.8 mW and 25.4 mW in the PDC and WCDMA modes, respectively. The supply voltage is 2.7 V.

Method and apparatus for synthetic widening of the bandwidth of voice signals

Abstract: The invention provides a method and an apparatus for synthetic widening of the bandwidth of voice signals. This is done by providing a narrowband voice signal at a predetermined sampling rate; carrying out analysis filtering on the sampled voice signal using filter coefficients, which are estimated from the sampled voice signal, for envelope widening; carrying out residual signal widening on the analysis-filtered voice signal; and carrying out synthesis filtering on the residual-signal-widened voice signal in order to produce a broader band voice signal. The analysis filtering is carried out using identical filter coefficients to those used for the synthesis filtering.

Improved Kalman filter design for three-dimensional radar tracking

Abstract: The problem of three-dimensional (3D) radar tracking is considered. The usual tracking filter design relying on first-order (or linear) approximations leads to poor convergence and erratic filter behavior in highly nonlinear situations. Simple filter algorithms that can overcome these ill effects are developed for two different types of 3D radar measurement. For each type of radar measurement, an accurate expression for the measurement covariance is obtained by evaluating inherent nonlinearities of radar measurements via coordinate transformation. Then algebraic manipulations and reasonable approximations are employed to yield a simple filter formulation based on the expression. The resulting filter equations are similar to the extended Kalman filter (EKF) and provide some useful insights into the behavior of linearized Kalman filters designed with radar measurements. Finally, simulation results show that the proposed approach is very effective in accounting for the measurement nonlinearities.