Showing papers on "Prototype filter published in 1990"

A new approach to harmonic compensation in power systems-a combined system of shunt passive and series active filters

Abstract: A novel approach to compensating for harmonics in power systems is presented. It is a combined system of a shunt passive filter and a small rated series active filter. The compensation principle is described, and some filtering characteristics are discussed in detail. Excellent practicability and validity to compensate for harmonics in power systems are demonstrated experimentally. Although the source harmonic voltage was only 1%, the source harmonic current reached about 10% before the series active filter was started. After it was started, no harmonic current flowed into the shunt passive filter. In addition, no harmonic voltage appeared at the terminals of the shunt passive filter, because the source harmonic voltage was applied to the series active filter. The total loss of the series active filter was less than 40 W. It is concluded that the combined system is far superior in efficiency to conventional shunt active filters. >

A practical approach to harmonic compensation in power systems-series connection of passive and active filters

Abstract: The authors present a combined system with a passive filter and a small-rated active filter, both connected in series with each other. The passive filter removes load produced harmonics just as a conventional filter does. The active filter plays a role in improving the filtering characteristics of the passive filter. This results in a great reduction of the required rating of the active filter and in eliminating all the limitations faced by using only the passive filter, leading to a practical and economical system. The active filter has a much smaller rating than a conventional active filter. Experimental results obtained from a prototype model are shown to verify the theory developed. >

Multidelay block frequency domain adaptive filter

Abstract: A flexible multidelay block frequency domain (MDF) adaptive filter is presented. The distinctive feature of the MDF adaptive filter is to allow one to choose the size of an FFT tailored to the efficient use of the hardware, rather than the requirements of a specific application. The MDF adaptive filter also requires less memory and thus reduces the hardware requirements and cost. In performance, the MDF adaptive filter introduces smaller block delay and is faster, making it ideal for a time-varying application such as modeling an acoustic path in a teleconference room. This is achieved by using a smaller block size, updating the weight vectors more often, and reducing the total execution time of the adaptive process. The MDF adaptive filter compares favorably to other frequency-domain adaptive filters when its adaptation speed and misadjustment are tested in computer simulations. >

Microwave active filters based on coupled negative resistance method

Abstract: A novel coupled negative resistance method for building a microwave active bandpass filter is introduced. Based on this method, four microstrip line end-coupled filters were built. Two are fixed-frequency one-pole and two-pole filters, and two are tunable one-pole and two-pole filters. In order to broaden the bandwidth of the end-coupled filter, a modified end-coupled structure is proposed. Using the modified structure, an active filter with a bandwidth up to 7.5% was built. All of the filters show significant passband performance improvement. Specifically, the passband bandwidth was broadened by a factor of 5 to 20. >

Optimized fast Hankel transform filters

Abstract: In the linear digital filter theory for calculation of Hankel transforms it is possible to find explicit series expansions for the filter coefficients. A method is presented for optimizing the Hankel filters calculated in this way. For a certain desired accuracy of computation, the sampling density and filter length are minimized by choosing the parameters determining the filter characteristics according to the analytical properties of the input function. A new approach to the calculation of the filter coefficients has been developed for these optimized filters. The length of the filters may be further reduced by introducing a shift in the sampling scheme.

Minimum mean absolute error estimation over the class of generalized stack filters

Abstract: A class of sliding window operators called generalized stack filters is developed. This class of filters, which includes all rank order filters, stack filters, and digital morphological filters, is the set of all filters possessing the threshold decomposition architecture and a consistency property called the stacking property. Conditions under which these filters possess the weak superposition property known as threshold decomposition are determined. An algorithm is provided for determining a generalized stack filter which minimizes the mean absolute error (MAE) between the output of the filter and a desired input signal, given noisy observations of that signal. The algorithm is a linear program whose complexity depends on the window width of the filter and the number of threshold levels observed by each of the filters in the superposition architecture. The results show that choosing the generalized stack filter which minimizes the MAE is equivalent to massively parallel threshold-crossing decisions making when the decision are consistent with each other. >

Efficient reconstruction of band-limited sequences from nonuniformly decimated versions by use of polyphase filter banks

Abstract: An efficient polyphase structure for the reconstruction of a band-limited sequence from a nonuniformly decimated version is developed. Theoretically, the reconstruction involves the implementation of a bank of multilevel filters, and it is shown that how all these reconstruction filters can be obtained at the cost of one Mth band low-pass filter and a constant matrix multiplier. The resulting structure is therefore more general than previous schemes. In addition, the method offers a direct means of controlling the overall reconstruction distortion T(z) by appropriate design of a low-pass prototype filter P(z). Extension of these results to multiband band-limited signals and to the case of nonconsecutive nonuniform subsampling are also summarized, along with generalizations to the multidimensional case. Design examples are included to demonstrate the theory, and the complexity of the new method is seen to be much lower than earlier ones. >

Generation of OTA-C filter structures from active RC filter structures

Abstract: The nodal voltage simulation method has been employed previously to derive new active RC and MOSFET-C filters from well-known active RC filters. This technique is shown here to also be useful for obtaining new OTA-C filters from well-known active RC filters. Specifically, it is shown how OTA-C biquads can be directly generated from active RC filters while retaining their useful properties. >

Steerable filters for early vision, image analysis, and wavelet decomposition

Abstract: An efficient architecture is presented to synthesize filters of arbitrary orientations from linear combinations of basis filters, allowing one to adaptively 'steer' a filter to any orientation, and to determine analytically the filter output as a function of orientation. The authors show how to design and steer filters, and present examples of their use in several tasks: the analysis of orientation and phase, angularly adaptive filtering, edge detection, and shape-from-shading. It is also possible to build a self-similar steerable pyramid representation which may be considered to be a steerable wavelet transform. The same concepts can be generalized to the design of 3-D steerable filters, which should be useful in the analysis of image sequences and volumetric data. >

Design of recursive digital filters with variable characteristics

Abstract: A new method for changing the cut-off frequency of infinite impulse response (IIR) digital filters with a single parameter is derived. the method is based on the use of the Taylor series expansion of the lowpass-to-lowpass frequency transformation. the filter structure is a parallel connection of real or complex allpass sections. the tuning range is several octaves for narrowband filters. By taking also the complementary output, a power-complementary filter pair with tunable crossover frequency is readily obtained. Tunable centre frequency and bandwidth are obtained by incorporating a special lowpass-to-bandpass transformation into the structure. Design examples and a description of a signal processor implementation are included.

Linear phase coefficient composite filter banks for distortion-invariant optical pattern recognition

Abstract: A new class of composite filters called the linear phase coefficient composite filters (LPCCFs) is introduced. Unlike previous approaches to composite filter design, this method considers the training set selection and the filter design simultaneously. The LPCCF is obtained by summing the training images weighted by complex weights of unit magnitude and linear phase. This paper also presents a way of combining the outputs of a bank of LPCCFs. Computer simulations are also included to quantify the performance of this approach.

Adaptive linearization schemes for weakly nonlinear systems using adaptive linear and nonlinear FIR filters

Abstract: Three adaptive linearization schemes are proposed. In the first scheme, linearization is performed by canceling nonlinearity at the output of a physical system. In the second, a nonlinear postprocessor is employed to postdistort signals. In the third, a preprocessor is used. The schemes using a postprocessor and a preprocessor are designed for weakly nonlinear systems, whereas the scheme of linearization by cancellation at the output can be applied to problems with stronger nonlinearities. In all three methods, necessary estimates of linear and nonlinear operators are provided by adaptive linear and nonlinear filters. Typical simulation results for a physical system modeled by a Volterra series with a linear term, a quadratic term, and a cubic term are presented and are judged encouraging. >

Hybrid diffractive optical filter

Abstract: A hybrid filter having an inherently planar interference stack adhered to a nonplanar diffractive grating. The filter can have a planarization layer between the stack and grating. The hybrid filter has a spectral light transmittance value that is at least equal to or greater than the transmittance value of the least transmissive filter component. The filter also has a spectral bandwidth that is narrower than the spectral bandwidth of either filter component.

Convergence behavior and N-roots of stack filters

Abstract: The convergence behavior of two types of stack filters is investigated. Both types are shown to possess the convergence property and to exhibit nontrivial behavior. The first type of stack filter has the erosive property; it erodes any input signal to a root after a sufficient number of passes. The second type of stack filter has the dilative property; it dilates any input signal to a root after a sufficient number of passes. For each type of stack filter, an algorithm is presented which can determine a filter that has any specific signal or set of signals as roots. These two algorithms are efficient in that their execution time is a linear function of the length of the input signal, the width of the filter window, and the number of signals to be preserved. Since some stack filters have the phenomenon of oscillations when they filter some input signals successively, a partial ordering is defined over the set of stack filters which makes it possible to determine upper and lower bounds for these oscillations. >

Nonrecursive and recursive stack filters and their filtering behavior

Abstract: It is shown that stack filters that are based on symmetric threshold functions and preserve median-filter roots make all inputs converge to roots or to cycles of period two. This is an important result, since these filters have useful roots (the median-filter roots), and they are time symmetric, i.e. time reversal of an input sequence of such a filter is equivalent to time reversal of the output sequence of the filter. In order to construct stack filters without cycles, the recursive stack filter, which is an extension of the recursive median filter, is introduced. It is shown that a recursive stack filter has the same roots as the corresponding nonrecursive stack filter; also, given a nonrecursive filter from the class mentioned above, the corresponding recursive filter will make every input signal of finite converge to a root in a finite number of passes. >

Tracking of bandpass signals using center-frequency adaptive filters

Abstract: A class of constrained adaptive filters, called recursive center-frequency adaptive filters, is applied to the tracking of bandpass signals. For this application, these filters form a class of completely digital tracking filters which are shown to have several advantages over existing analog and semidigital tracking filters. A procedure of analyzing the tracking behavior of these filters using the ordinary differential equation approach is presented. The tracking performance of two examples of these filters is analyzed for step, ramp, and sinusoidal variations of the input signal center frequency, and these are shown to track the signals very effectively. The dependency of the performance on the parameters of the algorithm, transfer function of the filter, and the input signal is studied. >

The design of multiplierless digital data transmission filters with powers-of-two coefficients

Abstract: An efficient search technique is presented for the design of FIR (finite impulse response) digital transmit and receive matched filters whose coefficients are represented by sums and/or differences of powers-of-two. These filters are ideally suited for custom VLSI implementation since power-of-two multipliers are obtained for free in a dedicated hardware implementation. Thus only a few adders or subtracters are required for each tap of the filter, and therefore fairly high-order filters can be implemented on a single VLSI chip. Due to their very simple structure these multiplierless filters could potentially operate at very high sampling rates to accommodate baud rates in the microwave digital radio range. >

Filter system with controlled amplitude in stopband or passband

Abstract: A filter system which may be configured either as a stopband (notch) or passband filter having controlled gain and low ripple throughout its stopband or passband. The system includes a filter network which is operative for splitting an input signal spectrum into a pair of complimentary spectrum components and a coupler for proportionately combining these spectrum components in accordance with a prescribed gain factor. The filter network may either comprise a pair of complimentary band reject and band pass filters connected to a common input terminal or a diplexer including identical filters coupled between a splitter and a combiner.

Combination of active and passive filters

Abstract: A combination of passive electrical filters and active electrical filters in a series arrangement forming active passive band pass or high pass or low pass or band reject filters. The preferred embodiment includes an amplifier stage to include high input impedance. These filter combinations are free of internal electrical noise.

Design of linear phase FIR notch filters

Abstract: This paper investigates some approaches for designing one-dimensional linear phase finite-duration impulse-responses (FIR) notch filters, which are based on the modification of several established design techniques of linear phase FIR band-selective filters Based on extensive design examples and theoretical analysis, formulae have been developed for estimating the length of a linear phase FIR notch filter meeting the given specifications In addition, the design of two-dimensional linear phase FIR notch filters is briefly considered Illustrative examples are included

Some characteristics of error surfaces for insufficient order adaptive IIR filters

Abstract: The error surfaces associated with insufficient order adaptive filters with white input are considered. It is shown that some sufficient, but not necessary, conditions for multimodality of these surfaces can be identified. A convenient method of determining the stationary points of some insufficient order adaptive filters with colored inputs is presented. >

An octave-band MMIC active filter

Abstract: The design, fabrication, and measurement of a GaAs monolithic, active bandpass filter with a passband from 4 to 8 GHz is described. The circuit uses a set of cascaded lumped- and distributed-element LC circuits isolated by a feedback amplifier to realize an equivalent fourth-order filter response. The final circuit was fabricated on a 3-mm*2-mm GaAs substrate. >

Optimal, efficient, recursive edge detection filters

Abstract: The design of an optimal, efficient, infinite-impulse-response (IIR) edge detection filter is described. J. Canny (1986) approached the problem by formulating three criteria designed in any edge detection filter: good detection, good localization, and low spurious response. He maximized the product of the first two criteria while keeping the spurious response criterion constant. Using the variational approach, he derived a set of finite extent step edge detection filters corresponding to various values of the spurious response criterion, approximating the filters by the first derivative of a Gaussian. A more direct approach is described in this paper. The three criteria are formulated as appropriate for a filter of infinite impulse response, and the calculus of variations is used to optimize the composite criteria. Although the filter derived is also well approximated by first derivative of a Gaussian, a superior recursively implemented approximation is achieved directly. The approximating filter is separable into two linear filters operating in two orthogonal directions allowing for parallel edge detection processing. The implementation is very simple and computationally efficient. >

Frequency agile, dielectrically loaded resonator filter

Abstract: A frequency agile, dielectrically loaded bandpass filter is disclosed in which capacitive layers on the top surface of the filter are selectively switched to ground in order to affect a change in the center frequency of the passband response of the filiter. A PIN diode switching network, including a means for biasing the diode, is used to effectively place the capacitive layers in parallel with the quarter-wavelength transmission line resonators contained within the block of the filter.

Design of digital filters using simulated annealing

Abstract: A simulated annealing optimization algorithm is used in the design of finite impulse response (FIR) filters and fifth-order LDI all-pole digital filters. Using simulated annealing, sin(x)/x precompensating scheme based on both the FIR and LDI structures is presented. The design process begins with a random set of finite precision coefficients for the filter, making no attempt to obtain a good initial starting coefficient. The use of simulated annealing allows the use of nonclassical transfer functions in the design of digital filters, i.e. it allows for the design of arbitrary magnitude response filters. >

Design and DSP-chip implementation of a novel bilinear-LDI digital Jaumann filter

Abstract: The design and implementation, on a digital signal processing (DSP) chip, of a novel high-quality recursive digital filter is presented. The proposed digital filter is derived from an equally resistively terminated lossless Jaumann two-port network by using the bilinear-LDI (bilinear lossless discrete integrator) design technique. It has the important practical property that all the inductor-based states can be computed simultaneously and all the capacitor-based states can be computed simultaneously, thereby permitting a fast parallel-processing speed which is virtually independent of the order of the filter. This digital filter can be made minimal in the number of multipliers, requiring n multipliers for the realization of lowpass and bandpass filters, and n+1 multipliers for the realization of highpass and bandstop filters, where n is the order of the continuous-time prototype reference filter. It is shown that when implemented using modern DSP chips, such a filter exhibits very high-quality performance characteristics. >

Mechanically tunable MSW bandpass filter with combined magnetic units

Abstract: An MSW (magnetostatic wave) bandpass filter with combined magnetic units composed of main and submagnetic units has been developed. Its volume is reduced to one fifth that of conventional yttrium-iron-garnet (YIG)-sphere filters. The filter can be tuned mechanically and keep the same shape of transmission response at 2.5 approximately 2.75 GHz. The insertion loss can be kept less than 3 dB within 20-MHz bandwidth by adjusting the magnetic field distribution. >

A CMOS switched-current filter technique

Abstract: Basic design techniques and considerations for switched-current (SI) circuits are presented, and experimental results from integrated filters are given. By means of analogies to switched-capacitor circuits, a five-pole lowpass Chebyshev SI filter has been integrated in a 2- mu m N-well, double-metal CMOS technology. The average die area is about 200 mil/sup 2/ per SI pole. The current mirror gain factors were derived by means of signal flow-graph techniques starting with the RLC prototype. A doubly terminated five-pole Chebyshev filter was designed for a 0.1-dB ripple bandwidth of 5 kHz with a sampling frequency of 128 kHz. The measured response is shown. The noise floor is about 70 dB down with respect to the passband. A three-pole elliptic SI filter has also been integrated to illustrate the realization of transmission zeros with SI filter techniques. >

Stable interruptible filter for dual beam computed tomography

Abstract: A data acquisition filter for a dual beam CT machine uses two spectral filters for alternately receiving the signal from each detector depending on the state of the x-ray beam. When one filter is filtering the detector signal, the other filter is in a "hold" state where its output and internal values are frozen. Each filter effectively filters only the signal occurring during one beam state without being effected by the signal occurring during the other beam state or by the passage of time during the other beam state The output of the filter in the holding state is constant and may be sampled at any time during this period.

Automatic layout synthesis for FIR filters using a silicon compiler

Abstract: A silicon compiler for finite impulse response (FIR) filters is presented. The synthesis system takes as inputs only filter specifications and processing word lengths, and generates the FIR filter mask patterns in a few minutes. The system consists of two programs: an FIR filter design program to determine FIR filter coefficients at the minimal filter order to meet design objectives, and a module generator to generate mask patterns according to optimal parameters obtained by the filter design program. For describing layout structures correctly and easily, the module generator provides graphical layout description tools, and includes mechanisms to permit designing the structures before leaf-cells are completed. Layouts for several filters which have been successfully generated in a short time are described. A system-generated single-chip VLSI chrominance-luminance separator for NTSC composite TV signals using four FIR filters is shown. >