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Showing papers on "Prototype filter published in 1989"


Journal ArticleDOI
TL;DR: A technique is developed for the design of analysis filters in an M-channel maximally decimated, perfect reconstruction, finite-impulse-response quadrature mirror filter (FIR QMF) bank that has a lossless polyphase-component matrix E(z).
Abstract: A technique is developed for the design of analysis filters in an M-channel maximally decimated, perfect reconstruction, finite-impulse-response quadrature mirror filter (FIR QMF) bank that has a lossless polyphase-component matrix E(z). The aim is to optimize the parameters characterizing E(z) until the sum of the stopband energies of the analysis filters is minimized. There are four novel elements in the procedure reported here. The first is a technique for efficient initialization of one of the M analysis filters, as a spectral factor of an Mth band filter. The factorization itself is done in an efficient manner using the eigenfilters approach, without the need for root-finding techniques. The second element is the initialization of the internal parameters which characterize E(z), based on the above spectral factor. The third element is a modified characterization, mostly free from rotation angles, of the FIR E(z). The fourth is the incorporation of symmetry among the analysis filters, so as to minimize the number of unknown parameters being optimized. The resulting design procedure always gives better filter responses than earlier ones (for a given filter length) and converges much faster. >

175 citations


Journal ArticleDOI
TL;DR: An analytical treatment of the two-beam coupling devices is given in a Laplace transform framework in the undepleted pump approximation assuming plane wave inputs to allow a unified treatment ofThe current status of optical novelty filters and related devices is reviewed.
Abstract: A novelty filter detects what is new in a scene and may be likened to a temporal high-pass filter. The current status of optical novelty filters and related devices, based upon four-wave mixing and two-beam coupling in photorefractive media, is reviewed. A detector that shows only what is not new, a monotony filter, may be likened to a temporal low-pass filter. Demonstrations of high- and low-pass and bandpass temporal image filters are then discussed. An analytical treatment of the two-beam coupling devices is given in a Laplace transform framework in the undepleted pump approximation assuming plane wave inputs. This allows a unified treatment of the various filter characteristics. >

138 citations


Journal ArticleDOI
K. Chen1
TL;DR: It is shown that the function of a stack filter can be realized in k-step recursive use of one binary processing circuit, and the time-area complexity of the proposed filter is O(k) as compared with O(2/sup k/) for stack filters.
Abstract: It is shown that the function of a stack filter can be realized in k-step recursive use of one binary processing circuit. The time-area complexity of the proposed filter is O(k) as compared with O(2/sup k/) for stack filters. The proposed digital realizations are simple and modular in structure, and suitable for VLSI implementation. Analog/digital (A/D) hybrid realizations have the advantage that there is no need for an A/D converter array when the original signals come from an integrated sensor array. An experimental digital rank-order filter with a window size of three and arbitrary number of input bits is designed and implemented in a 3- mu m double-metal polysilicon gate CMOS process. The chip has been fabricated and measurement results are correct with a clock frequency of up to 110 MHz. >

123 citations


Journal ArticleDOI
TL;DR: A design technique for variable filters with coefficients that are directly computable from the specified spectral parameters is proposed, which expresses the frequency specifications by using a curve-fitting technique.
Abstract: In some applications the frequency characteristics of a filter may be required to change during the course of signal processing. This requirement can be satisfied by filters with coefficients that are directly computable from the specified spectral parameters. Such filters are referred to as variable filters. A design technique for variable filters is proposed. The filter coefficients are expressed as analytical functions of the frequency specifications by using a curve-fitting technique. Several examples are presented to illustrate the method. >

119 citations


Journal ArticleDOI
TL;DR: The Ll- Filters are introduced to generate the order statistic filters (L-filters) and the nonrecursive linear, or finite-duration impulse-response (FIR), filters.
Abstract: The Ll-filters are introduced to generate the order statistic filters (L-filters) and the nonrecursive linear, or finite-duration impulse-response (FIR), filters. Such estimators are particularly effective filtering signals that do not necessarily follow Gaussian distributions. They can be designed to restore one-dimensional or multidimensional signals corrupted by noise of impulsive type. Such filters are appealing since they are suitable for being made robust against the presence of spurious outliers in the data. >

101 citations


Journal ArticleDOI
TL;DR: A number of previous theories characterizing the well-known median and ranked-order filters are extended to a broader class of filters and input signals.
Abstract: Necessary and/or sufficient conditions on both the filter coefficients and the signal process are derived in order that nonrecursive order statistic (OS) and linear filtering are equivalent operations. The results indicate that an understanding of OS filters hinges on a better understanding of the properties of signals containing logically monotonic components. The results extend a number of previous theories characterizing the well-known median and ranked-order filters to a broader class of filters and input signals. >

92 citations


Proceedings ArticleDOI
08 May 1989
TL;DR: In this paper, a class of window functions is introduced for designing FIR filters, which are obtained from the rectangular window by using a simple frequency transformation, which contains an adjustable parameter with which the mainlobe width and, correspondingly, the minimum stopband attenuation of the resulting filter can be controlled.
Abstract: A class of window functions is introduced for designing FIR filters. These window functions are obtained from the rectangular window by using a simple frequency transformation. The frequency transformation contains an adjustable parameter with which the mainlobe width and, correspondingly, the minimum stopband attenuation of the resulting filter can be controlled. The transition bandwidth of the filter can then be controlled by the filter order. Like the well-known Kaiser window, the proposed windows are close approximations to the discrete prolate functions which minimize the sidelobe energy. The FIR filters obtained by using the new window are slightly better than those obtained by using the Kaiser window. The main advantages of the proposed window compared to the Kaiser window are that the new window possesses analytic expressions in both the time and frequency domains and no power series expansions are required in evaluating the window function. Furthermore, it provides a better approximation to the discrete prolate functions. >

88 citations


Journal ArticleDOI
TL;DR: This study presents a performance evaluation of non-linear filters derived from the robust point estimation theory by classification of various approaches to nonlinear filtering into three types of estimators according to the process of the filter.
Abstract: Nonlinear filters are used in many applications, including speech and image processing, owing to their ability to suppress noise and preserve signal features such as edges. This study presents a performance evaluation of non-linear filters derived from the robust point estimation theory. The first part of the work is a classification of various approaches to nonlinear filtering into three types of estimators according to the process of the filter. The second part is a computer implementation and evaluation of all of the filters discussed. Finally, a summary of experimental results is presented.

83 citations


Journal ArticleDOI
TL;DR: In this paper, a technique in which the filter modulation is included in the synthesis of a synthetic-discriminant-function matched spatial filter (SDF MSF) is presented.
Abstract: A technique in which the filter modulation is included in the synthesis of a synthetic-discriminant-function matched spatial filter (SDF MSF) is presented. In the filter synthesis, a system of simultaneous nonlinear equations is solved with an iteration procedure. A computer simulation of the new method using thresholded images of the Space Shuttle over a range of aspect angles was performed for phase-only filters (POFs) and binary-phase-only filters (BPOFs). The filters constructed are capable of obtaining the specified peak-correlation response to within 1% with a high signal-to-clutter-ratio for the one-class problem, the two-class problem, and the multilevel problem. In contrast, conventional projection SDF POFs and BPOFs are unable to produce the desired peak-correlation response.

78 citations


Proceedings ArticleDOI
13 Jun 1989
TL;DR: In this paper, a capacitive compensation technique is described for the design of microstrip parallel coupled filters with improved passband symmetry and very low spurious response up to 2.5 times the center frequency.
Abstract: A capacitive compensation technique is described for the design of microstrip parallel coupled filters with improved passband symmetry and very low spurious response up to 2.5 times the center frequency. The compensated structure does not require any extra computer-aided design tools and is compatible with monolithic microwave integrated circuit technology. The technique is useful for the design of filters on alumina as well as GaAs substrates. Two design examples are considered. >

77 citations


Journal ArticleDOI
TL;DR: In this article, a general class of median-type filters which contain non-recursive (finite impulse response) and recursive (infinite impulse reaction) substructures is introduced.
Abstract: The authors introduce a general class of median-type filters which contain nonrecursive (finite impulse response) and/or recursive (infinite impulse response) substructures. The overall structure of the linear median hybrid (LMH) filters may be either nonrecursive or recursive. Altogether eight basic filter structures are introduced, and four of them are analyzed in more detail. General properties of the root signals, i.e. signals invariant to filtering, are discussed. Expressions characterizing the noise attenuating properties of some of the filters are derived, and a statistical analysis of the filters is performed. The LMH filters are simple structures enabling both analog and digital implementations and having properties not attainable with linear or median filters. By designing the linear substructures appropriately it is possible to tailor the root signals of the LMH filters to meet the time-domain specifications of a particular applications. >

Journal ArticleDOI
TL;DR: In this article, the dc-side and ac-side active filters were compared with the passive filters, both on the dc side and the ac side of the HVDC terminal, and the results showed that the DC side active filter can filter the three dominant harmonics at the 12th, the 24th and the 36th frequencies.
Abstract: The active filter concepts are compared with the passive filters, both on the dc-side and the ac-side of the HVDC terminal. The Dickinson terminal of the CU HVDC transmission line is used as the basis for this study. Based on this study, the following conclusions are reached for the dc-side and the ac-side active filters: DC-Side Active Filters 1. The active filter is designed for filtering of the three dominant harmonics at the 12th, the 24th and the 36th harmonic frequencies. Computer simulations indicate that these three dominant harmonics can be attenuated to less than 1 % at the transmission line terminal, compared to the harmonic voltages produced by the converter. It should be noted that any other harmonic frequencies which may be troublesome can be included in the injection current to filter these harmonics. Higher order harmonics have small amplitudes; hence, the required total injection current and the total kVAs of the components in the active filter are not changed significantly by adding higher order harmonics. 2. The effective cost of the dc-side active filter is estimated to be 186 k$, which also includes the present worth of the additional 17.3 kW losses in the active filter, as compared to the passive filter. The active filter costs is lower by only 27 k$, as compared to the passivefilter cost of 213 k$. Given the uncertainty in the cost estimates, both filters are comparable in terms of cost.

Patent
Andre Tore Mikael1
07 Apr 1989
TL;DR: An adaptive digital filter including a non-recursive part and a recursive part, which can be updated in a simple and reliable manner, is presented in this article, where a linear combination is formed with adaptive weighting factors (W0-W3) from the output signals of the recursive filters.
Abstract: An adaptive digital filter including a non-recursive part and a recursive part, and which can be updated in a simple and reliable manner. The recursive part of the filter has a plurality of separate, permanently set recursive filters (13-16) with different impulse responses, and a linear combination is formed with adaptive weighting factors (W0-W3) from the output signals of the recursive filters (13-16). The filter is updated by a single (e(n)) being utilized for updating the non-recursive part (11) of the filter and the adaptive weighting factors (W0-W3) in the recursive part of the filter.

Proceedings ArticleDOI
01 Jan 1989
TL;DR: By applying different partitioning schemes to the FDAF structure it is shown that it is possible to decouple the length of the transformation and the length N of the adaptivejlter.
Abstract: Convergence characteristics of adaptive filters are influenad b y the statistical properties (correlation) of the input signal. In order t o remove this dependency, decordation can be applied which can be performed in fmquency domain by simple power normalisation of each sepamte frequency bin. This leads to the Frequency Domain Adaptive Filter (FDAF), in which the length of the transformation between timeand frequency-domain is choosen to be equal to the length N of the adaptive filter. However the relevant length Of this transformation is determined by the stotieticol properties of the input signal and needs not to be equal to N . By applying different partitioning schemes to the FDAFstructure it is shown that it is possible to decouple the length of the transformation and the length N of the adaptivejlter. -

Journal ArticleDOI
TL;DR: An experimental 10.7-MHz switched-capacitor bandpass filter was demonstrated that exhibits a 400-kHz bandwidth with a 42-MHz sampling rate in this article, where basic design issues of such highfrequency filters were also addressed with emphasis on dynamic range and power constraints.
Abstract: An experimental 10.7-MHz switched-capacitor bandpass filter is demonstrated that exhibits a 400-kHz bandwidth with a 42-MHz sampling rate. Basic design issues of such high-frequency filters are also addressed with emphasis on dynamic range and power constraints. A theoretical square relation between power and center frequency agrees well with experimental results. The sixth-order differential bandpass filter chip occupies 2 mm/sup 2/ using a 2.25- mu m gate double-poly CMOS technology. >

Journal ArticleDOI
TL;DR: It is shown that although a conjecture concerning sufficient conditions to guarantee unimodality of error surfaces for adaptive pole-zero infinite-impulse-response (IIR) filters is valid for first- and second-order filters, it is not true in general without an additional constraint introduced by T. Soderstrom.
Abstract: It is shown that although a conjecture proposed by S. Stearns (ibid., vol.ASSP-29, p.763-6, June 1981) concerning sufficient conditions to guarantee unimodality of error surfaces for adaptive pole-zero infinite-impulse-response (IIR) filters is valid for first- and second-order filters, it is not true in general without an additional constraint introduced by T. Soderstrom (Autom., vol.11, p.93-9, Jan. 1982). Proofs of the conjecture for a first- and second-order adaptive filters are given. Counterexamples are presented to illustrate the failure of this conjecture for higher order filters when the condition of Soderstrom is not satisfied for both matching order and overparameterizing cases. >

Journal ArticleDOI
TL;DR: A 3rd order Butterworth low-pass filter has been implemented in a standard 2.4um CMOS process and achieves a dynamic range of 63dB.
Abstract: A continuous-time third-order Butterworth low-pass filter with a nominal cutoff frequency of 945 Hz implemented in a standard 2.4- mu m CMOS process is discussed. The filter is built from fully balanced linearized transconductors and capacitors. The power consumption of the filter, including bias and automatic-tuning circuits, is 12.6 mu W from a single 3-V power supply. It achieves a dynamic range of 63 dB with a total harmonic distortion of less than 0.5%. The total chip area is 1 mm/sup 2/. >

Journal ArticleDOI
TL;DR: In this article, a spectral transformation technique is proposed for the design of 3D recursive digital filters with planar and beam frequency responses, based on circuit-theoretic concepts to generate stable 3D continuous domain networks from a corresponding 1D prototype network.
Abstract: A spectral transformation technique is proposed for the design of three-dimensional (3-D) recursive digital filters having planar and beam frequency responses. Circuit-theoretic concepts are used to generate stable 3-D continuous domain networks from a corresponding one-dimensional (1-D) prototype network. Implementation schemes are developed in the discrete domain that convert the 3-D analog network into a 3-D wave digital filter (WDF). The proposed 3-D discrete beam and planar filters offer significant advantages over direct-form realizations including the low-sensitivity properties of the WDF and simplicity of the structure. >

Journal ArticleDOI
J.J. Nielsen1
TL;DR: It is shown that the expected disturbance filter frequency response due to quantization can be controlled and the effect is concentrated in tolerant frequency bands and significantly reduced in critical ones.
Abstract: A generalization of the statistical approach of D.S.K. Chan and L.R. Rabiner (IEEE Trans. Audio Electroacoust., vol.AU-21; p.354-66, Aug. 1973) for predicting the effects of quantized coefficients in direct-form finite-impulse response (FIR) filters is presented. The analysis is used on more sophisticated quantizing procedures involving linear feedback of rounding errors. It is shown that the expected disturbance filter frequency response due to quantization can be controlled. The effect is concentrated in tolerant frequency bands and significantly reduced in critical ones. The price paid is an increase of overall expected disturbance. The method is demonstrated in a practical filter design. A low-pass filter for factor 2 decimation that has greater margins for quantization effects in the passband than in the stopband is specified. >

Journal ArticleDOI
26 Jun 1989
TL;DR: A novel technique for building active filters using switched-capacitor (SC) circuits is proposed and it is shown that by using a single filter a wide range of harmonics can be controlled.
Abstract: A novel technique for building active filters using switched-capacitor (SC) circuits is proposed. The principle of operation, methods of control, and analysis and design of a typical SC filter are presented. To assess its effectiveness, the technique is used to control input current harmonics in a phase-controlled converter. It is shown that by using a single filter a wide range of harmonics can be controlled. >

Journal ArticleDOI
TL;DR: In this article, a variety of quasi-planar low-pass, bandpass, and bandstop filters suitable for millimeter-wave applications is reviewed and a modified finline filter is presented to improve the performance and manufacturing of filters in waveguides below cutoff.
Abstract: A variety of quasi-planar low-pass, bandpass, and bandstop filters suitable for millimeter-wave applications is reviewed. The emphasis is on ladder-shaped E-plane bandpass filters to highlight their advantages as well as limitations in terms of design, performance, and manufacturing. To extend their range of application it is suggested that E-plane filters be cascaded for better passband separation. A modified finline filter is presented to improve the performance and manufacturing of filters in waveguides below cutoff. It is shown that plated through-holes can simplify filter housing fabrication and that surface-metallized composite housings are a lightweight and low-cost alternative to metal housings. >

Journal ArticleDOI
G.D. Boyd1, F. Heismann1
TL;DR: In this paper, an acoustooptic filter in LiNbO/sub 3/ without the Doppler shift, which is undesirable as an intracavity element, is presented.
Abstract: A proposal is presented for an acoustooptic filter in LiNbO/sub 3/ without the Doppler shift, which is normally present in acoustooptic filters and which is undesirable as an intracavity element. This tunable filter has a significantly wider tuning range than previously demonstrated electrooptic filters. A structure incorporating an interdigital acoustic transducer, polarization filter, and single-mode optical waveguide is shown. The design presented works by cascading two acoustooptic filters with an intermediate polarizer, where the Doppler shift experienced in the first filter is precisely compensated for in the second filter. Hence, the output light of the two cascaded acoustooptic filters is not shifted in frequency, thus making the device suitable for applications inside a laser cavity. The filter with intermediate polarizer can be integrated with a single-mode waveguide and requires only a single interdigital acoustic transducer. Crystal symmetry and acoustic power considerations are treated in detail. >

Journal ArticleDOI
TL;DR: A practical method for obtaining minimum specifications for the gain and phase of the integrator circuits of a sixth-order Chebyshev band-pass filter from the filter specifications is developed.
Abstract: The design of integrated, high-frequency, continuous-time filters has made considerable progress in the past few years. As the signal frequencies increase the design of the integrator circuits used in most of these filters becomes more critical. To give direction to the circuit design, minimum specifications for the gain and phase of the integrator circuits would be helpful. A practical method for obtaining these integrator specifications from the filter specifications is developed. The method is applied to a sixth-order Chebyshev band-pass filter, and the result is verified by computer simulation. >

Journal ArticleDOI
TL;DR: In this paper, an alternative technique for 2-D variable digital filters is proposed, where the filter coefficients are expressed as analytic functions with their independent variables being the frequency specifications of interest.

Journal ArticleDOI
TL;DR: In this paper, the authors present principles for triple-mode filter designs based on a sidewall-tuned triplemode cavity and two iris geometries, which allow optimal filter realizations with a minimum number of triple- and dual-mode cavities.
Abstract: The authors present principles for triple-mode filter designs based on a sidewall-tuned triple-mode cavity and two iris geometries. It is shown that these principles allow optimal filter realizations with a minimum number of triple- and dual-mode cavities. Verification of these principles is obtained by realizing several filter types. Experimental results are provided for elliptic-function filters of fifth, sixth (both in in-line configuration), and seventh order. >

Journal ArticleDOI
TL;DR: A class of nonlinear filters that are based on the rank estimates (R-estimates) of location parameters in statistical theory is introduced and it is shown how moving-window rank filters ( R-filters) can be defined starting from rank estimates of location.
Abstract: A class of nonlinear filters that are based on the rank estimates (R-estimates) of location parameters in statistical theory is introduced. It is shown how moving-window rank filters (R-filters) can be defined starting from rank estimates of location. These filters utilize the relative ranks of the observations in each window to produce an output value. The idea of rank Winsorization is extended to that of averaging only observations which lie within small temporal neighborhoods. This leads to a definition of the class of generalized Wilcoxon (GW) filters, which are parameterized by three parameters, namely the degrees of temporal and rank Winsorization and the degree of averaging. The GW filters can be defined to have desirable characteristics of edge preservation, detail retention, and impulse rejection, in addition to the property of Gaussian noise smoothing. Performance characteristics of these filters are considered through analysis and simulation. The filters show that all three well-known classes of robust location estimates, the L-, M-, and R-estimates, can be applied to nonlinear smoothing of signals. >

Journal ArticleDOI
TL;DR: In this article, two switched-capacitor (SC) cells for simple design of bandpass active filters from reference lowpass filters are presented The pseudo-Npath approach used is based on the z to -z/sup N/ transformation An elliptic sixth-order symmetric bandpass filter with center frequency of 200 kHz and 02 dB in-band ripple within a bandwidth of 2 kHz has been designed.
Abstract: Two switched-capacitor (SC) cells for simple design of bandpass active filters from reference lowpass filters are presented The pseudo-N-path approach used is based on the z to -z/sup N/ transformation An elliptic sixth-order symmetric bandpass filter with center frequency of 200 kHz and 02 dB in-band ripple within a bandwidth of 2 kHz has been designed The basic lowpass differential topology has been derived using flowgraph techniques from a C03 to 10 21 degrees passive prototype Presently such a filter is under integration in a 3- mu m double poly CMOS technology >

Journal ArticleDOI
TL;DR: In this article, it was shown that the structure of the optimal receiving filter corresponds to a bank of matched filters, each followed by a linear transversal filter, which minimizes the mean-square error between the sampled input to the decision device and the corresponding transmitted symbols.
Abstract: The filter minimizes the mean-square error between the sampled input to the decision device and the corresponding transmitted symbols. It is shown that the structure of this optimal receiving filter corresponds to a bank of matched filters, each followed by a linear transversal filter. Applications of this theory cover nonlinear channels with memory, linearly or nonlinearly modulated signals, and disturbances other than Gaussian noise (e.g. interference from other data sources). >

Journal ArticleDOI
TL;DR: A purely analytical technique for the design of recursive digital filters with equiripple magnitude behavior is presented and it is suggested that this sort of algorithm can be used advantageously in adaptive filtering because of the simplicity of the analytical solutions.
Abstract: A purely analytical technique for the design of recursive digital filters with equiripple magnitude behavior is presented. The design is accomplished in a transformed variable denoted by w and defined as w=(z+1/z)/2. The method can be attributed to the class of rational Chebyshev approximations. It uses design steps corresponding to those for elliptical filters in the analog domain. Using Jacobian elliptic functions, a conformal mapping of the w-plane is introduced which describes the solutions of equiripple symmetric specifications. These are not halfband specifications, since the attenuation and the bandripple in the passband are chosen independently of those in the stopband. Examples are given to demonstrate the efficiency of the approach. It is suggested that this sort of algorithm can be used advantageously in adaptive filtering because of the simplicity of the analytical solutions. The algorithms are well behaved, since the infinite series expansion of elliptic functions and the consecutive manipulation with abridged forms of these series have been avoided. >

Proceedings ArticleDOI
08 May 1989
TL;DR: In this article, a new class of morphological filters is proposed for smoothing an image contaminated with noise, which includes the combination of linear and nonlinear operations in the design of the new filter.
Abstract: A new class of morphological filters is proposed for smoothing an image contaminated with noise. A multiple model that includes the combination of linear and nonlinear operations is used in the design of the new filter. The performance of the averaging version of this new filter is similar to that of the alpha-trimmed mean filter. The structure-preserving properties of this new filter depend on the values assigned to the coefficients in the filter. The idempotent property is obtained when a closing-min and opening-max version of the filter is used. The root structure of the output signal is also investigated. >