Showing papers on "Prototype filter published in 1993"

LUM filters: a class of rank-order-based filters for smoothing and sharpening

Abstract: A new class of rank-order-based filters, called lower-upper-middle (LUM) filters, is introduced. The output of these filters is determined by comparing a lower- and an upper-order statistic to the middle sample in the filter window. These filters can be designed for smoothing and sharpening, or outlier rejection. The level of smoothing done by the filter can range from no smoothing to that of the median filter. This flexibility allows the LUM filter to be designed to best balance the tradeoffs between noise smoothing and signal detail preservation. LUM filters for enhancing edge gradients can be designed to be insensitive to low levels of additive noise and to remove impulsive noise. Furthermore, LUM filters do not cause overshoot or undershoot. Some statistical and deterministic properties of the LUM filters are developed, and a number of experimental results are presented to illustrate the performance. These experiments include applications to 1D signals and to images. >

Input filter design for power factor correction circuits

Abstract: The issues involved in the design of power factor correction circuit input filters are significantly different than those involved in the design of input filters for DC-DC power converters. So far, there exist no guidelines for high-frequency AC power converter input filter design. This paper addresses these issues and proposes the use of a high order, actively damped filter to achieve the required EMI attenuation and power factor requirements. The new filter topology typically provides 50% filter size reduction over the standard filter designs, and simultaneously minimizes the filter-converter interaction. >

The gamma-filter-a new class of adaptive IIR filters with restricted feedback

Abstract: Generalized feedforward filters, a class of adaptive filters that combines attractive properties of finite impulse response (FIR) filters with some of the power of infinite impulse response (IIR) filters, are described. A particular case, the gamma filter, generalizes Widrow's adaptive transversal filter (adaline) to an infinite impulse response filter. Yet, the stability condition for the gamma filter is trivial, and LMS adaptation is of the same computational complexity as the conventional transversal filter structure. Preliminary results indicate that the gamma filter is more efficient than the adaptive transversal filter. The authors extend the Wiener-Kopf equation to the gamma filter and develop some analysis tools. >

Narrowband interference cancellation through the use of digital recursive notch filters

Abstract: A digital notch filter apparatus for removing narrowband interference signals from a wideband communication signal. The apparatus includes a recursive digital passband interference acquisition filter, an acquisition mode center frequency control circuit, a recursive digital passband interference tracking filter, a tracking mode center frequency control circuit and an interference detection circuit. The tracking filter has a narrower bandwidth than the acquisition filter. Both the acquisition and tracking filters are gang tuned so that the center frequencies of their passbands are adjusted in unison. During interference acquisition mode, the acquisition mode center frequency control circuit is used is to adjust the center frequency of the filters over the bandwidth covered by the interference canceler. When the interference detection circuit detects an interference signal within the passband of the tracking filter, the tracking center frequency control circuit is used to adjust the center frequency of the acquisition and tracking filters to insure that the filters remain locked onto the narrowband interference signal. Upon detection of the interference signal, the portion of the communication signal which includes the narrowband interference signal that is passed through the narrow passband tracking filter is subtracted from the communication signal to remove the narrowband interference signal. When no interference signal is detected, the communication signal is permitted to pass to the output of the notch filter circuit without attenuation.

Active varactor tunable bandpass filter

Abstract: A microstrip active varactor tunable bandpass filter is presented. The filter is based upon the negative resistance method using a microstrip combline network. A 1.8-GHz, two-pole device with a bandwidth of 2% was constructed. 0.6 octave tuning with significant improvement in passband performance was achieved compared with its passive counterpart. >

A new class of nonlinear filters-neural filters

Abstract: A class of nonlinear filters based on threshold decomposition and neural networks is defined. It is shown that these neural filters include all filters defined either by continuous functions, such as linear finite impulse response (FIR) filters, or by Boolean functions, such as generalized stack filters. Adaptive least-mean-absolute-error and adaptive least-mean-square-error algorithms are derived for determining optimal neural filters. As special cases, adaptive generalized stack and adaptive generalized weighted order statistic filtering algorithms under both error criteria are derived. Experimental results in 1D and 2D signal processing are presented to compare the performances of the adaptive neural filters and other widely used filters. >

Recursively implementating the Gaussian and its derivatives

Abstract: Gaussian filtering is one of the most successfully operation in computer vision in order to reduce noise, calculating the gradient intensity change or performing Laplacian or the second directional derivative of an image. However, it is well known that in a multi-resolution context, where the need for large filters is required, this technique suffers from the fact it is a computationally expensive since the number of operations per point in convolving an image with a Gaussian filter is directly proportional to the width of the operator. We propose in this paper a technique in order to use Gaussian filtering with a reduced and fixed number of operations per output independently of the size of the filter. The key of our approach is to approximate in a mean square sense the prototype Gaussian filters with an exponentially based filter family depending on the same scale factor than the Gaussian filters (i.e. s) and then to implement in an exact and recursive way the approximate filters. An important point of the design presented in this paper is that dealing with Gaussian filters having different scale factor (i.e. s) will not require a new design algorithm as. The coefficients looked for in the recursive realization are determined function of the scale factor of each considered prototype filter, namely the Gaussian filter, its first and second derivative. Some experimental results will be shown to illustrate the efficiency of the approximation process and some applications to edge detection problems and multi-resolution techniques will be considered and discussed.

The design of arbitrary FIR digital filters using the eigenfilter method

Abstract: The design of finite impulse response (FIR) digital filters for approximating an arbitrary function (in both magnitude and phase) in the least-square sense is studied. The design method is based on the computation of an eigenvector of an appropriate real, symmetric and positive-definite matrix. The design of the complex-coefficient filter is shown to be an extension of the design of the real-coefficient filter. Several design examples, including the constant-group-delay filters and digital phase all-pass filters, are presented. Comparisons to existing methods are made. >

Multidimensional multirate filters and filter banks derived from one-dimensional filters

Abstract: A method by which every multidimensional (M-D) filter with an arbitrary parallelepiped-shaped passband support can be designed and implemented efficiently is presented. It is shown that all such filters can be designed starting from an appropriate one-dimensional prototype filter and performing a simple transformation. With D denoting the number of dimensions, the complexity of design and implementation of the M-D filter are reduced from O(N/sup D/) to O(N). Using the polyphase technique, an implementation with complexity of only 2N is obtained in the two-dimensional. Even though the filters designed are in general nonseparable, they have separable polyphase components. One special application of this method is in M-D multirate signal processing, where filters with parallelepiped-shaped passbands are used in decimation, interpolation, and filter banks. Some generalizations and other applications of this approach, including M-D uniform discrete Fourier transform (DFT) quadrature mirror filter banks that achieve perfect reconstruction, are studied. Several design example are given. >

A polynomial-time algorithm for designing digital filters with power-of-two coefficients

Abstract: An algorithm is presented for designing digital filters with coefficients expressible as sums of signed power-of-two (SPT) terms. For each filter gain, the time complexity of the algorithm is a second-order polynomial in the filter order and is a first-order polynomial in the filter wordlength. Unlike conventional methods where each coefficient is allocated a fixed number of SPT terms, the author's method allows the number of SPT terms for each coefficient to vary subject to the number of SPT terms for the entire filter. This provides the possibility of finding a better filter without increasing the number of adders, which determines the realization cost for a given filter length. Application of the algorithm to finite impulse response (FIR) filter designs shows that it achieves up to 8.9 dB improvement over simulated annealing and mixed integer linear programing on the normalized peak ripples of example filters. >

Design and characterization of optimal FIR filters with arbitrary phase

Abstract: An algorithm for designing a Chebyshev optimal FIR filter that approximates an arbitrary complex-valued frequency response is presented. This algorithm computes the optimal filter by solving the dual to the filter design problem. It is guaranteed to converge theoretically and requires O(N/sup 2/) computations per iteration for a filter of length N. For the first time, properties of the optimal filter are derived, and the case where the desired filter has arbitrary constant group delay is studied in detail. >

Reconfigurable programmable digital filter architecture useful in communication receiver

Abstract: A VLSI integrated circuit, which comprises a single IIR input and global section and identically-structured cascadable filter sections, each of which filter sections includes a pair of time-multiplexed, real-coefficient, input-weighted FIR filter units and additional delay means, can be selectively programmed to operate in any one of a number of different filter configurations that can define real FIR or IIR filters, complex FIR or IIR filters, or filters which are various combinations thereof. One or more of such integrated circuits are useful for implementing a digital deghosting and/or equalization filter.

Multiplierless 2-band perfect reconstruction quadrature mirror filter (PR-QMF) banks

Abstract: The Multiplierless Quadrature Mirror Filter concept is used in the design of analysis and synthesis filter banks to be used for the sub-band coding of various types of signals. The individual filters in the analysis and synthesis filter banks are designed to be near linear in phase, non-symmetrical in time, and to have equal bandwidth frequency responses. These multiplierless filters are relatively easy to implement in hardware and allow for the sub-band coding of signals with minimal computational complexity so as to result perfect signal reconstruction. Furthermore, these filters are particularly well suited for configuration in hierarchical sub-band structures.

A Spectral Factorization Approach to Pseudo-QMF Design

Abstract: A novel approach to the design of M-channel pseudo-quadrature mirror filter (QMF) banks is presented. In this approach, the prototype filter is obtained as a spectral factor of a 2M/sup th/ band filter. This completely eliminates the need for optimization, whereas in conventional pseudo-QMF designs the main computational effort is in optimization of the prototype. The aliasing cancellation constraint is derived such that all the significant aliasing terms are canceled. The overall transfer function of the analysis/synthesis system has an approximately flat magnitude response in the frequency region epsilon >

Dual-mode dielectric resonator filters with improved spurious performance

Abstract: Novel configurations for dual-mode dielectric resonators operating in the HEH/sub 11/ mode or the HEE/sub 11/ mode are presented. The use of the proposed configurations in the design of dual-mode dielectric resonator filters leads to a design with a remarkably improved spurious performance. Techniques to control the frequency separation between the resonance frequencies of the two dual-modes HEH/sub 11/ and HEE/sub 11/ are discussed. Experimental results are presented to verify the validity of the proposed configurations. The proposed configurations promise to be useful in the design of dual-mode dielectric resonator filters for satellite multiplexers having stringent rejection requirements. >

Optimal digital filters for long‐latency components of the event‐related brain potential

Abstract: A fundamentally important problem for cognitive psychophysiologists is selection of the appropriate off-line digital filter to extract signal from noise in the event-related brain potential (ERP) recorded at the scalp. Investigators in the field typically use a type of finite impulse response (FIR) filter known as moving average or boxcar filter to achieve this end. However, this type of filter can produce significant amplitude diminution and distortion of the shape of the ERP waveform. Thus, there is a need to identify more appropriate filters. In this paper, we compare the performance of another type of FIR filter that, unlike the boxcar filler, is designed with an optimizing algorithm that reduces signal distortion and maximizes signal extraction (referred to here as an optimal FIR filter). We applied several different filters of both types to ERP data containing the P300 component. This comparison revealed that boxcar filters reduced the contribution of high-frequency noise to the ERP but in so doing produced a substantial attenuation of P300 amplitude and, in some cases, substantial distortions of the shape of the waveform, resulting in significant errors in latency estimation. In contrast, the optimal FIR filters preserved P300 amplitude, morphology, and latency and also eliminated high-frequency noise more effectively than did the boxcar filters. The implications of these results for data acquisition and analysis are discussed.

A stable truncated series approximation of the reduction-to-the-pole operator

Abstract: We combine a stabilized reduction‐to‐the‐pole and an upward continuation filter to produce meaningful reduced‐to‐the‐pole fields at low magnetic latitudes. The stabilizing procedure is based on the development, in Taylor’s series, of the theoretical expression for the reduction‐to‐the‐pole filter in the wavenumber domain. The filter instability is caused by the huge filter amplitudes along the magnetization azimuth, which are expressed by the infinite sum of terms close to unity. The stabilizing procedure reduces to simply truncating the infinite series. The upward continuation filter attenuates the high wavenumber component of the noise and allows us to design a stabilized filter closer to the theoretical one. Besides, quantitative interpretations of source depth based on the filtered field are more reliable when using upward continuation as compared with arbitrary low‐pass filters. The proposed filter was applied to synthetic data of a single prism uniformly magnetized along a supposedly known direction...

Combined decimation/interpolation filter for ADC and DAC

Abstract: A COMBINED DECIMATION/INTERPOLATION FILTER FOR ADC AND DAC (analog-to-digital converter and digital-to-analog converter) provides a single filter which may be used both as a decimation filter and as an interpolation filter. It is simple and inexpensive. It gains simplicity and inexpensivenes by carefully selecting the tap weights of the filter, cascading multiple filters, time domain multiplexing the multiple filters into a single filter, and using adders instead of multipliers to provide the tap weights.

Exact reconstruction analysis/synthesis filter banks with time-varying filters

Abstract: The authors examine some of the analysis/synthesis issues associated with FIR (finite impulse response) time-varying filter banks where the filter bank coefficients are allowed to change in response to the input signal. Several issues are identified as being important for realizing performance gains from time-varying filter banks in image coding applications. These issues relate to the behavior of the filters as transition from one set of filter banks to another occurs. Lattice structure formulations for the time-varying filter bank problems are introduced and discussed in terms of their properties and transition characteristics. >

Compensation Characteristics of Shunt Active and Series Active Power Filters

Abstract: Active power filters have been used in practice to suppress the harmonic interference in power systems. To compensate for harmonic currents of loads active power filters are usually connected to power systems in parallel with the loads. These filters which are called shunt active filters here are very effective to loads that can be considered as current sources, such a thyristor rectifiers with large DC reactances. Many papers have covered the shunt active filters applied to these currentsource loads. No paper, however, has discussed characteristics of the shunt active filters when they are applied to voltage-source loads.On the other hand, since more and more diode rectifiers with capacitive DC filters are used recently, harmonics generated by which become an issue.The diode rectifier with capacitive DC filters behaves like a voltage source rather than a current source. When a shunt active filter is applied to such a diode rectifier, the current injected from the shunt active filter may flow into the diode rectifier. As a result, harmonics of the source current cannot be reduced effectively, and harmonic current flowing into the diode rectifier increases largely.This paper reveals the above-mentioned problem of shunt active filters analytically and experimentally. Then a series active filter is proposed to suppress harmonic current of the diode rectifiers. The features, operating conditions, and considerations of shunt active filters and series active filters are described analytically and demonstrated experimentally. Taking a diode rectifier with capacitive DC filter as a typical voltage-source load, compensation characteristics of shunt active filters and series active filters are discussed by experiment and simulation. The validity of the series active filters is illustrated experimentally.

A general class of current mode filters

Abstract: A new class of nonlinear analog filters with linear input-output response is proposed. These filters are an extension of the recently introduced 'log filter' class. By virtue of the small node voltage excursions, these filters share the current mode characteristic of log filters. It is shown that the equations for the filter are obtained by using hyperbolic functions for mappings on the state space in linear systems. The resulting exponential state space filters are then given circuit implementations. >

Very high sample rate digital filters using the δ operator

Abstract: The paper reports upon the theoretical and practical issues concerned with achieving digital filters in which the sample rate is very much higher than the dominant frequency, illustrated by means of a general-purpose second-order digital filter section which can readily be configured for lowpass, highpass, bandpass or bandreject operation over a range of frequencies. The δ operator approach is used throughout, although a brief comparison is included to show what the implications would be of using the z operator. The paper shows how full performance can be achieved even with four orders of magnitude between the filter frequency and the sampling frequency, including practical results from implementation using a DSP device.

Partial spectrum reconstruction using digital filter banks

Abstract: The problem of reconstructing a part of the spectrum is reduced to designing the filter bank to satisfy a set of conditions. For the case considered here, these conditions cannot be satisfied simultaneously, so perfect reconstruction is not possible. The necessary and sufficient conditions on the filters so that the resulting filter bank cancels most alias components are found. Such filter banks are called partial alias cancellation filter banks. The product of the polyphase transfer matrices of these filter banks must be a block pseudocirculant matrix. An algorithm design procedure is discussed, and examples are given to demonstrate the theory. >

Input filter design for multiple-module DC power systems

Abstract: When multiple DC-DC converters are operated from a DC bus with a nonzero source impedance, the possibility exists for undesirable interactions between each individual regulator and the input impedance of the other regulators on the bus. Consequently, criteria for input filter design in the presence of a significant source impedance are developed, which, when used in conjunction with already-known input filter criteria, permit the input filter to be designed so that the entire system operates reliably. Proper filter design decouples the negative regulator impedances from the bus, leaving only the passive input filter impedances to affect the other converters. These filter impedances appear in parallel with the source impedance, and reduce the overall source impedance. Hence the use of multiple modules on the same bus improves the performance of the individual regulators. A specific example, the buck current mode controlled converter, is examined in detail. Extensive experimental evidence is presented to verify the analytical results. >

Cochlear filter bank with switched-capacitor circuits

Abstract: A parallel dilating-filters switched-capacitor filter bank is described in simulation of the cochlea. Area-saving is achieved by filter-sharing, effective sum-gain amplifier designs, and using area efficient nth-order filter designs, and in particular using a biquadratic filter design using charge-differencing. The structure is easily expandable to include more channels by extending with additional filters and output amplifiers, or by using several chips with different sampling frequencies in parallel connection. An offset-compensated area-efficient switched-capacitor sum-gain amplifier circuit design is described and can be used in the filter bank.

Current-mode multiple-feedback filters

Abstract: A different class of operational-amplifier-based biquadratic filters, the infinite-gain multiple-feedback filters, is transformed into its current version. This class has so far eluded any such transformation because the operational-amplifier is configured without resistive feedback loop. The current-mode multiple-feedback filters contain also a single operational-amplifier as the active element, and therefore an absolute comparison between voltage and current versions is feasible. The authors demonstrate, both theoretically and practically, that the current-mode multiple-feedback filters can extend the useful bandwidth with respect to their classical voltage-mode counterparts up to nearly the GB product of the operational-amplifier. This result is particularly interesting because voltage- and current-mode multiple-feedback filters contain exactly the same active and passive components. >

Complex correction of data acquisition channels using FIR equalizer filters

Abstract: The common conviction about FIR (finite impulse response) digital filters is that the number of necessary taps, to reach the same performance as provided by IIR (infinite impulse response) digital filters, is usually too large. Moreover, the standard FIR filter design algorithm (Remez exchange) allows the design of linear-phase filters only. Therefore, IIR filters are often preferred over FIR ones, without any further investigations. This paper presents a case study of complex (amplitude and phase) equalization of the passband of a commercial anti-aliasing filter. The novelty is the usage of an FIR filter for this purpose (or an FIR one, combined with a low-order IIR filter), and a thorough discussion of the special design aspects. It turns out that for the given anti-aliasing filter (a Cauer filter of order 11) an FIR filter of length 60...100 can perform as well as a 26/26 (numerator order/denominator order) IIR one. The properties are even better if a low-order IIR filter is used in combination with an FIR one (orders, e.g., 1/1+40/0). Because of the absence of stability problems and the ease of implementation, the use of FIR filters is suggested. >

CAD of T-septum waveguide evanescent-mode filters

Abstract: A mode-matching-based design for an evanescent-mode waveguide filters with T-septum shaped metal inserts is presented. Owing to the wideband characteristics of the T-septum waveguide, the proposed design constitutes a significant improvement over common evanescent-mode filters with respect to both size reduction and stopband behavior. The theoretical approach is verified for the example of a three-resonator 8.8-GHz filter prototype of less than 3/4-in length. The second passband is beyond 27 GHz. Since the design procedure takes higher-order mode interactions into account, good agreement between theory and experiment is obtained over the entire measurement range between 8.2 and 40 GHz. >