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Showing papers on "Filter design published in 2009"


Journal ArticleDOI
TL;DR: In this article, the authors proposed a new signal processing analysis of the bilateral filter, which complements the recent studies that analyzed it as a PDE or as a robust statistical estimator.
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 fast versions have been proposed. Unfortunately, little is known about the accuracy of such accelerations. 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 statistical estimator. 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. Importantly, this signal-processing perspective allows us to develop a novel bilateral filtering acceleration using downsampling in space and intensity. This affords a principled expression of accuracy in terms of bandwidth and sampling. The bilateral filter can be expressed as linear convolutions in this augmented space followed by two simple nonlinearities. This allows us to derive criteria for downsampling the key operations and achieving important acceleration of the bilateral filter. We show that, for the same running time, our method is more accurate than previous acceleration techniques. Typically, we are able to process a 2 megapixel image using our acceleration technique in less than a second, and have the result be visually similar to the exact computation that takes several tens of minutes. The acceleration is most effective with large spatial kernels. Furthermore, this approach extends naturally to color images and cross bilateral filtering.

789 citations


Journal ArticleDOI
TL;DR: This paper introduces a new iterative design procedure of L and LCL filters for low-voltage active-front-end PWM two-level voltage source converters using the analytical expression of the converter harmonic voltages by Bessel functions.
Abstract: This paper introduces a new iterative design procedure of L and LCL filters for low-voltage active-front-end PWM two-level voltage source converters. The analytical expression of the converter harmonic voltages by Bessel functions is applied to design the filter parameters for defined maximum grid current harmonics. Different filter designs are derived for various resonance frequencies and inductance split factors of LCL filter. The minimum amount of stored energy of passive components is used to select a final filter design. A voltage-oriented control scheme, including active damping, is applied in the (400 V and 50 kVA) experimental setup. Both simulation and experimental investigations are presented to verify the accuracy of the filter design procedure. Finally, the steady state and transient performance of the active front end with different LCL filters are depicted.

409 citations


Journal ArticleDOI
TL;DR: The aim is to develop a robust fault detection approach to the T-S fuzzy systems with Brownian motion by using a general observer-based fault detection filter as a residual generator, and attention is focused on the design of both the fuzzy-rule-independent and the fuzzy -rule-dependent fault detection filters guaranteeing a prescribed noise attenuation level in an Hinfin sense.
Abstract: The paper deals with the robust fault detection problem for Takagi-Sugeno (T-S) fuzzy Ito stochastic systems. Our aim is to develop a robust fault detection approach to the T-S fuzzy systems with Brownian motion. By using a general observer-based fault detection filter as a residual generator, the robust fault detection is formulated as a filtering problem. Attention is focused on the design of both the fuzzy-rule-independent and the fuzzy-rule-dependent fault detection filters guaranteeing a prescribed noise attenuation level in an Hinfin sense. Sufficient conditions are proposed to guarantee the mean-square asymptotic stability with an Hinfin performance for the fault detection system. The corresponding solvability conditions for the desired fuzzy-rule-independent and fuzzy-rule-dependent fault detection filters are also established. Finally, a numerical example is provided to illustrate the effectiveness of the proposed theory.

315 citations


Proceedings ArticleDOI
14 Jun 2009
TL;DR: A relationship between weighted sum-rate and weighted MMSE in the MIMO-BC is established and two low complexity algorithms for finding a local weighted Sum-rate optimum based on alternating optimization are proposed.
Abstract: This paper studies linear transmit filter design for Weighted Sum-Rate (WSR) maximization in the Multiple Input Multiple Output Broadcast Channel (MIMO-BC). The problem of finding the optimal transmit filter is non-convex and intractable to solve using low complexity methods. Motivated by recent results highlighting the relationship between mutual information and Minimum Mean Square Error (MMSE), this paper establishes a relationship between weighted sum-rate and weighted MMSE in the MIMO-BC. The relationship is used to propose a low complexity algorithm for finding a local weighted sum-rate optimum based on alternating optimization. Numerical results studying sum-rate show that the proposed algorithm achieves high performance with few iterations.

282 citations


Journal ArticleDOI
TL;DR: In this paper, a wideband and compact filter design for common-mode noise suppression in high-speed differential signals is proposed based on two U-shaped and one H-shaped coupled patterned ground structure.
Abstract: A wideband and compact filter design for common-mode noise suppression in high-speed differential signals is proposed based on two U-shaped and one H-shaped coupled patterned ground structure. An equivalent model of three coupled LC resonators to predict the common-mode suppression characteristics is also developed with good agreement to the full-wave simulation and measurement result. A test sample is designed and fabricated on a standard printed circuit board (PCB). It is found the common-mode noise can be reduced by the filter over 15 dB from 3.6 to 9.1 GHz and over 75% of amplitude in the time domain. More important, the differential signal integrity, in terms of insertion loss and group delay in the frequency domain and eye diagrams in the time domain, is not degraded within the wide bandwidth. The fractional bandwidth of the stopband can reach 87% and the filter size is about 0.44 lambdag times 0.44 lambdag. The radiation caused by the common-mode current on the attached I/O cables is also efficiently suppressed by 10 dB on average within the designed stopband. To our best knowledge, it is the first low-cost common-mode filter designed for the gigahertz high-speed signals with the largest fractional bandwidth and most compact size on a PCB.

218 citations


Journal ArticleDOI
TL;DR: A new filtering method to remove Rician noise from magnetic resonance images is presented that relies on a robust estimation of the standard deviation of the noise and combines local linear minimum mean square error filters and partial differential equations for MRI, as the speckle reducing anisotropic diffusion did for ultrasound images.
Abstract: A new filtering method to remove Rician noise from magnetic resonance images is presented. This filter relies on a robust estimation of the standard deviation of the noise and combines local linear minimum mean square error filters and partial differential equations for MRI, as the speckle reducing anisotropic diffusion did for ultrasound images. The parameters of the filter are automatically chosen from the estimated noise. This property improves the convergence rate of the diffusion while preserving contours, leading to more robust and intuitive filtering. The partial derivative equation of the filter is extended to a new matrix diffusion filter which allows a coherent diffusion based on the local structure of the image and on the corresponding oriented local standard deviations. This new filter combines volumetric, planar, and linear components of the local image structure. The numerical scheme is explained and visual and quantitative results on simulated and real data sets are presented. In the experiments, the new filter leads to the best results.

210 citations


Journal ArticleDOI
TL;DR: This adaptive filter procedure proved a reliable and efficient tool to remove ECG artefact from surface EMGs with mixed and varied patterns of transient, short and long lasting dystonic contractions.

190 citations


Journal ArticleDOI
TL;DR: This paper discusses a design of stable filters with H"~ disturbance attenuation of Takagi-Sugeno fuzzy systems with immeasurable premise variables with a solution in terms of linear matrix inequalities (LMIs).

167 citations


Journal ArticleDOI
TL;DR: The OTA-C filter can be adopted to eliminate the out-of-band interference of the electrocardiogram (ECG) whose signal bandwidth is located within 250 Hz.
Abstract: This study presents a systematic design of the fully differential operational transconductance amplifier-C (OTA-C) filter for a heart activities detection apparatus. Since the linearity and noise of the filter is dependent on the building cell, a precise behavioral model for the real OTA circuit is created. To reduce the influence of coefficient sensitivity and maintain an undistorted biosignal, a fifth-order ladder-type lowpass Butterworth is employed. Based on this topology, a chip fabricated in a 0.18- mum CMOS process is simulated and measured to validate the system estimation. Since the battery life and the integration with the low-voltage digital processor are the most critical requirement for the portable diagnosis device, the OTA-based circuit is operated in the subthreshold region to save power under the supply voltage of 1V. Measurement results show that this low-voltage and low-power filter possesses the HD3 of -48.9 dB, dynamic range (DR) of 50 dB, and power consumption of 453 nW. Therefore, the OTA-C filter can be adopted to eliminate the out-of-band interference of the electrocardiogram (ECG) whose signal bandwidth is located within 250 Hz.

128 citations


Journal ArticleDOI
TL;DR: In this paper, the authors investigate the filter topology selection issue and present their research results on the effectiveness and costs of various filter topologies for harmonic mitigation, they show that the association of three single-tuned filters is a very appropriate solution for most typical harmonic problems.
Abstract: Passive filters have been a very effective solution for power system harmonic mitigation. These filters have several topologies that give different frequency response characteristics. The current industry practice is to combine filters of different topologies to achieve a certain harmonic filtering goal. However, there is a lack of information on how to select different filter topologies. This decision is based on the experience of present filter designers. The goal of this paper is to investigate the filter topology selection issue. It presents our research results on the effectiveness and costs of various filter topologies for harmonic mitigation. The research results show that the association of three single-tuned filters is a very appropriate solution for most typical harmonic problems.

127 citations


Proceedings Article
01 Aug 2009
TL;DR: Numerical results are provided to characterize different optimization criteria in terms of frequency selectivity of resulting prototype filters and total interference level of the filter bank structure.
Abstract: This paper concentrates on an efficient prototype filter design in the context of filter bank based multicarrier (FBMC) transmission. An advantage of the chosen method, frequency sampling technique, is that near perfect reconstruction prototype filters can be expressed using a closed-form representation with only a few adjustable parameters. The performance of various designs are analyzed using the offset-QAM based FBMC system. Numerical results are provided to characterize different optimization criteria in terms of frequency selectivity of resulting prototype filters and total interference level of the filter bank structure. Furthermore, it is shown what kind of performance trade-offs can be obtained by adjusting those free parameters. In this sense, the presented results offer useful information to a system designer.

Journal ArticleDOI
TL;DR: FanFan filter as discussed by the authors is a 2D double filter consisting of a low-pass along the degree n and a high-pass on the order m whose contour projection onto the (n, m) plane is fan-shaped.
Abstract: [1] Spatial low-pass filtering is necessary for processing the GRACE time-variable gravity (TVG) data which are otherwise plagued with short-wavelength noises. Here we devise a new non-isotropic filter, called the fan filter: In terms of the spherical harmonic spectrum, the fan filter is simply a 2-D double filter consisting of a low-pass along the degree n (the same as the conventional isotropic filter) simultaneously with a low-pass along the order m, whose contour projection onto the (n, m) plane is fan-shaped. It is deterministic and independent of a priori or external information, its implementation is straightforward, and the result is objective. Most importantly, we show that this simple filter performs well among its counterparts under similar conditions, in particular against the N-S striping noises prevalent in the GRACE TVG solutions. We demonstrate this with Gaussian weights at filter length and hence spatial resolution as fine as 300 km. We also deduce the fan filter's nominal amplitude-reduction factor as a function of the filter length for TVG signals that follow the Kaula rule.

Journal ArticleDOI
TL;DR: In this article, the performance of six widely used filter methods and their parameter values to derive regionally averaged water mass variations from GRACE data was evaluated. And the best results were provided by a decorrelation method that uses GRACE orbits for the filter design.
Abstract: SUMMARY Approximately seven years of time-variable gravity data from the satellite mission Gravity Recovery and Climate Experiment (GRACE) are available to quantify present-day mass variations on and near the Earth's surface. Mass variations caused by the continental water cycle are the dominant signal component after subtracting contributions from atmosphere and oceans. This makes hydrology a primary area of application of GRACE data. To derive water storage variations at the scale of large river basins, appropriate filter techniques have to be applied to GRACE gravity fields given in a global spherical harmonic representation. A desirable filter technique minimises both GRACE data error and signal leakage across the border of the region of interest. This study evaluates the performance of six widely used filter methods (isotropic filters, anisotropic filters and decorrelation methods) and their parameter values to derive regionally averaged water mass variations from GRACE data. To this end, filtered time series from GRACE for 22 of the world's largest river basins were compared to continental water mass variations from a multimodel mean of three global hydrological models (WGHM, GLDAS and LaD). Filter-induced biases for seasonal amplitudes and phases of water storage variations, as well as satellite and leakage error budgets, were quantified for each river basin and explained in terms of storage variations in and around the basin. The optimum filter types and filter parameters were identified for each basin. The best results were provided by a decorrelation method that uses GRACE orbits for the filter design. Our ranking between all filter types and parameters depended on the geographical location, shape and signal characteristics of the specific river basin. Based on a multicriterial evaluation of satellite and leakage error, as well as an error assessment of the hydrological data, the filter selection and parameter optimisation results were shown to be reliable for 17 river basins. The results serve as a guideline for the optimal filtering of GRACE global spherical harmonic coefficients for hydrological applications.

Journal ArticleDOI
TL;DR: In this paper, a delay-dependent H"~ performance analysis result is established for error systems without ignoring any terms in the derivative of Lyapunov functional by considering the relationship between the time-varying delay and its upper bound.

Patent
08 Apr 2009
TL;DR: In this paper, an adaptive filter having filter coefficients is used to generate an anti-noise signal to drive a speaker to produce sound waves to destructively interfere with an undesired sound in a quiet zone.
Abstract: An active noise control system generates an anti-noise signal to drive a speaker to produce sound waves to destructively interfere with an undesired sound in a quiet zone. The anti-noise signal is generated with an adaptive filter having filter coefficients. The coefficients of the adaptive filter may be adjusted based on a first filter adjustment from a first listening region, and a second filter adjustment from a second listening region. A first weighting factor may be applied to the first filter adjustment, and a second weighting factor may be applied to the second filter adjustment. The first and second weighting factors may dictate the location and size of the quiet zone as being outside or partially within at least one of the first listening region and the second listening region.

Journal ArticleDOI
TL;DR: A 0.13-mu m CMOS fourth-order notch filter for the rejection of the 5-6 GHz interference in UWB front-ends is reported and an algorithm for the automatic tuning and calibration of the filter is discussed and demonstrated.
Abstract: A 0.13-mu m CMOS fourth-order notch filter for the rejection of the 5-6 GHz interference in UWB front-ends is reported. The filter is integrated into an analog front-end for Mode #1 UWB. A thorough analysis based on a simplified model of the filter is carried out. An algorithm for the automatic tuning and calibration of the filter is also discussed and demonstrated. Two versions of the circuit are designed and fabricated: the first comprises a low-noise amplifier and the filter, and the second expands it to a complete front-end. In the latter version the filter was also redesigned. The filter provides more than 35 dB of attenuation and has a tuning range of 900 MHz, adding less than 30% power consumption to the LNA. The out-of-band IIP3 (higher than -13.2 dBm with the filter off) takes a 9-dB advantage from the filter and the compression of the gain due to the out-of-band blocker is reduced by at least 6 dB in the complete front-end. The conversion gain of the front-end is 25 dB per channel, its average noise figure is lower than 6.2 dB, and its in-band 1-dB compression point is higher than - 30 dBm at a power consumption of 32 mW.

Journal ArticleDOI
TL;DR: In this article, a multilayer non-uniform periodical structure is deployed in the filter design to improve the selectivity and upper stopband performance of the filter, and short-circuited stub resonators are then integrated to obtain multiple notch bands.
Abstract: A novel ultra-wideband (UWB) bandpass filter that is capable of integrating multiple notch bands is proposed in this paper. A multilayer nonuniform periodical structure, which can generate multiple transmission zeros, is deployed in the filter design to improve the selectivity and upper stopband performance of the filter. Short-circuited stub resonators are then integrated to obtain multiple notch bands. The compact footprint of the filter is achieved. UWB filters of this type without a notch and with single-, double-, and triple-notch bands are designed to meet the Federal Communications Commission-defined UWB indoor limit. The designed filters are verified by experiments and fabricated by using multilayer liquid-crystal polymer lamination technology. Both full-wave simulated and measured results are presented and good agreement between them is observed. The proposed filter has good performance and is attractive for UWB communication and radar systems.

Journal ArticleDOI
TL;DR: A new method for reducing random, spike-like noise in seismic data based on a 1D stationary median filter — the 1D time-varying median filter (TVMF), which strikes a balance between eliminating random noise and protecting useful information.
Abstract: Random noise in seismic data affects the signal-to-noise ratio, obscures details, and complicates identification of useful information. We have developed a new method for reducing random, spike-like noise in seismic data. The method is based on a 1D stationary median filter (MF) — the 1D time-varying median filter (TVMF). We design a threshold value that controls the filter window according to characteristics of signal and random, spike-like noise. In view of the relationship between seismic data and the threshold value, we chose median filters with different time-varying filter windows to eliminate random, spike-like noise. When comparing our method with other common methods, e.g., the band-pass filter and stationary MF, we found that the TVMF strikes a balance between eliminating random noise and protecting useful information. We tested the feasibility of our method in reducing seismic random, spike-like noise, on a synthetic dataset. Results of applying the method to seismic land data from Texas demons...

Patent
26 Nov 2009
TL;DR: In this paper, a method of providing frequency dependent signal attenuation is proposed, in which an RF input signal is split into a first signal portion and a second signal portion, and a component output signal portion is combined with the filtered signal portion to provide an RF output signal having frequency dependent attenuation.
Abstract: A method of providing frequency dependent signal attenuation. An RF input signal is split into a first signal portion and a second signal portion. Discrete time filtering, a negative group delay and bandstop filtering are applied to the first signal portion to provide a filtered signal portion. The second signal portion is applied to a component, and a component output signal portion is received from the component. The component output signal portion is combined with the filtered signal portion to provide an RF output signal having frequency dependent attenuation.

Proceedings ArticleDOI
17 May 2009
TL;DR: In this paper, a reasonable design method was brought out in order to achieve optimal effect under the precondition of saving inductance magnetic core of LCL filter, at the same time guaranteeing the resonance frequency of lCL filter was not too small lest restrict current controller resign.
Abstract: Since LCL filter has smaller inductance value comparing to L type filter with the same performance in harmonic suppression. it is gradually used in high-power and low-frequency current-source-controlled grid-connected converters. However design of LCL filter's parameter not only relates switch frequency ripple attenuation, but also impacts on performance of grid-connected current controller. This paper firstly introduced a harmonic model of LCL filter in grid-connected operation, then researched the variable relationship among LCL filter's parameter and resonance frequency and high-frequency ripple attenuation. Based on above analysis a reasonable design method was brought out in order to achieve optimal effect under the precondition of saving inductance magnetic core of LCL filter, at the same time guaranteeing the resonance frequency of LCL filter was not too small lest restrict current controller resign. Finally this design method was verified by the experimental results.

Journal ArticleDOI
TL;DR: In this article, a new type of compact ultra-wideband (UWB) bandpass filter (BPF) based on interference techniques is presented. And the proposed filter is formed by a short-ended coupled line coupler connected in parallel to a transmission line.
Abstract: This letter presents a new type of compact ultra-wideband (UWB) bandpass filter (BPF) based on interference techniques. The structure is formed by a short-ended coupled line coupler connected in parallel to a transmission line. The transmission and reflection zeros of the filter can be controlled through the analytical equations and rules given. To validate the suggested filter scheme, the design, fabrication and measurement of a microstrip UWB BPF is presented.

Journal ArticleDOI
TL;DR: A structure for the compensation of frequency-response mismatch errors in M-channel time-interleaved analog-to-digital converters (ADCs) is introduced, which makes use of a number of fixed digital filters, approximating differentiators of different orders, and a few variable multipliers that correspond to parameters in polynomial models of the channel frequency responses.
Abstract: This paper introduces a structure for the compensation of frequency-response mismatch errors in M-channel time-interleaved analog-to-digital converters (ADCs). It makes use of a number of fixed digital filters, approximating differentiators of different orders, and a few variable multipliers that correspond to parameters in polynomial models of the channel frequency responses. Whenever the channel frequency responses change, which occurs from time to time in a practical time-interleaved ADC, it suffices to alter the values of these variable multipliers. In this way, expensive on-line filter design is avoided. The paper includes several design examples that illustrate the properties and capabilities of the proposed structure.

Journal ArticleDOI
TL;DR: The basic idea of the study is to formulate the robust fault detection filter design as a H"~ model-matching problem and a solution of the optimal problem is presented via a linear matrix inequality (LMI) formulation.
Abstract: In this paper, the robust fault detection filter design problem for linear time invariant (LTI) systems with unknown inputs and modeling uncertainties is studied. The basic idea of our study is to formulate the robust fault detection filter design as a H"~ model-matching problem. A solution of the optimal problem is then presented via a linear matrix inequality (LMI) formulation. The main results include the formulation of robust fault detection filter design problems, the derivation of a sufficient condition for the existence of a robust fault detection filter and construction of a robust fault detection filter based on the iterative of LMI algorithm.

Journal ArticleDOI
TL;DR: The generalized sampling theorem is applied to directly estimate the ideal output of each sub-ADC using the outputs of all the sub- ADCs to present a multichannel filtering approach for TIADC mismatch compensation.
Abstract: Published methods that employ a filter bank for compensating the timing and bandwidth mismatches of an M-channel time-interleaved analog-to-digital converter (TIADC) were developed based on the fact that each sub-ADC channel is a downsampled version of the analog input. The output of each sub-ADC is filtered in such a way that, when all the filter outputs are summed, the aliasing components are minimized. If each channel of the filter bank has N coefficients, the optimization of the coefficients requires computing the inverse of an MN times MN matrix if the weighted least squares (WLS) technique is used as the optimization tool. In this paper, we present a multichannel filtering approach for TIADC mismatch compensation. We apply the generalized sampling theorem to directly estimate the ideal output of each sub-ADC using the outputs of all the sub-ADCs. If the WLS technique is used as the optimization tool, the dimension of the matrix to be inversed is N times N. For the same number of coefficients (and also the same spurious component performance given sufficient arithmetic precision), our technique is computationally less complex and more robust than the filter-bank approach. If mixed integer linear programming is used as the optimization tool to produce filters with coefficient values that are integer powers of two, our technique produces a saving in computing resources by a factor of approximately (100.2N(M- 1)/(M-1) in the TIADC filter design.

Journal ArticleDOI
TL;DR: In this article, an analytical approach to extracting the coupling matrix of a narrow band general Chebyshev bandpass filter with losses is presented. And the concept of phase loading is revealed in the community of computer aided tuning.
Abstract: This paper presents a novel analytical approach to extracting of the coupling matrix of a narrow band general Chebyshev bandpass filter with losses. The approach is very useful for computer aided tuning of a microwave bandpass filter. The concept of phase loading is revealed for the first time in the community of computer aided tuning. The analytical approach consists of three elements: 1) a theoretic formula that leads to a practical scheme for determining the phase loading; 2) a theoretic formula for de-embedding the section of unknown transmission lines at the two ports of a filter; and 3) a theory for determining the unloaded Q of a filter if the loss for each resonator is nearly the same. To make the approach easy to use, some practical techniques for reconstructing rational functions of Y-parameters from a set of filter response are also provided in the paper. The proposed diagnosis approach is applicable to a general coupled resonator filter with losses and therefore can be effectively used in computer aided tuning of high order filters with cross couplings.

Journal ArticleDOI
TL;DR: An extensive evaluation using simulation and experimental results from a DSP-based laboratory prototype is conducted in order to verify and validate the good performance achieved by the proposed FBS passive/hybrid power filter.
Abstract: This paper presents a new solution for filtering current harmonics in three-phase four-wire networks. The original four-branch star (FBS) filter topology presented in this paper is characterized by a particular layout of single-phase inductances and capacitors, without using any transformer or special electromagnetic device. Via this layout, a power filter, with two different and simultaneous resonance frequencies and sequences, is achieved-one frequency for positive-/negative-sequence and another one for zero-sequence components. This filter topology can work either as a passive filter, when only passive components are employed, or as a hybrid filter, when its behavior is improved by integrating a power converter into the filter structure. The paper analyzes the proposed topology, and derives fundamental concepts about the control of the resulting hybrid power filter. From this analysis, a specific implementation of a three-phase four-wire hybrid power filter is presented as an illustrative application of the filtering topology. An extensive evaluation using simulation and experimental results from a DSP-based laboratory prototype is conducted in order to verify and validate the good performance achieved by the proposed FBS passive/hybrid power filter.

Journal ArticleDOI
TL;DR: In this paper, a new method for planning single-tuned passive harmonic filters to control harmonic voltage distortion throughout a power system is presented, where the probabilistic characteristics of the harmonic source currents and network harmonic impedances in the filter planning are taken into account.
Abstract: This paper presents a new method for planning single-tuned passive harmonic filters to control harmonic voltage distortion throughout a power system. In the problem, the probabilistic characteristics of the harmonic source currents and network harmonic impedances in the filter planning are taken into account. The objective is to minimize the total filter installation cost, while the harmonic voltage limits and filter component constraints are satisfied with predetermined confidence levels. To obtain the optimal size of each filter component of the planning problem, the proposed procedure is first to find the candidate filter buses based on the sensitivity analysis. Next, the formulated probability-constrained problem is transformed into a deterministic nonlinear programming problem and is solved by a genetic-algorithm-based optimizer. The proposed solution procedure is tested with an actual distribution network and is verified by the conventional deterministic approach and by the Monte Carlo simulation. Numerical experiences show that the proposed method yields favorable results compared with the other two approaches.

Journal ArticleDOI
TL;DR: In this paper, a recursive implementation of the direct Park transformation was proposed to avoid the filtering stage and allow accurate SRF controllers to be designed, and the proposed implementation is not dependent on PCC conditions.
Abstract: Load harmonic currents and load unbalances reduce power quality (PQ) supplied by electrical networks. Shunt active power filters (SAPFs) are a well-known solution that can be employed to enhance electrical PQ by injecting a compensation current at the point of common coupling (PCC) of the SAPF, the load, and the electrical grid. Hence, SAPF controllers must determine the instantaneous values of the compensation reference current, including nondesirable components of the load current. A family of SAPF controllers, which evaluates the compensation reference current using synchronous rotating frames (SRFs), employs a structure based on Park transformations: direct transform, low- pass filtering (LPF), and inverse transform. The cutoff frequency and the filter order of the LPF stage must be designed properly in order to obtain an accurate reference current and a fast dynamic response of these SAPF controllers. This paper proposes a recursive implementation of the direct Park transformation that avoids the filtering stage and allows accurate SRF controllers to be designed. Moreover, the proposed implementation is not dependent on PCC conditions. The proposed implementation is evaluated using a three-phase, three-wire SAPF and compared with LPF-based controllers by simulation and experiment.

Journal ArticleDOI
TL;DR: A delay-dependent approach developed to derive the main results in terms of linear matrix inequalities (LMIs) and these results have less conservatism than those for the fuzzy-rule-independent filter design case.

Book ChapterDOI
01 Jan 2009
TL;DR: In this paper, a bandpass filter is defined as the connection of a high pass filter and a low pass filter in series, which is commonly done in wideband filter applications.
Abstract: Publisher Summary The simplest design of a bandpass filter is the connection of a high pass filter and a low pass filter in series, which is commonly done in wideband filter applications. Thus, a first order high pass filter and a first order low pass provide a second order bandpass, while a second order high pass filter and a second order low pass result in a fourth order bandpass response. In comparison to wideband filters, narrowband filters of higher order consist of cascaded second order bandpass filters that use the Sallen–Key or the multiple feedback topology. A filter with an even order number consists of second order stages only, while filters with an odd order number include an additional first order stage at the beginning. For low pass filter design, the higher the corner frequency of a partial filter, the higher is its Q. Therefore, to avoid the saturation of the individual stages, the filters need to be placed in the order of rising Q values. To design the first stage of a third order unity gain Bessel low pass filter, assuming the same values for fC and C1, requires a different value for R1. When operating at unity gain, the noninverting amplifier reduces to a voltage follower, thus inherently providing superior gain accuracy. High pass filters use the same two topologies as the low pass filters: Sallen–Key and multiple feedback. The only difference is that the positions of the resistors and the capacitors have changed. As with the low pass filters, higher order high pass filters are designed by cascading first order and second order filter stages. The filter coefficients are the same ones used for the low pass filter design.