Showing papers on "Prototype filter published in 1981"

Acousto-Optic Tunable Filters

Abstract: This paper reviews the theory, techniques, and applications of acousto-optic tunable filters (AOTF). Two basic types of AOTF are described, i.e., the collinear and the noncollinear configurations. The important device characteristics of the AOTF presented include the bandpass response and spectral resolution, angular aperture, and the filter tuning relation. This review concludes with a discussion of various AOTF applications.

Error surfaces of recursive adaptive filters

Abstract: For an adaptive filter with N adjustable coefficients or weights, the "error surface" is a plot, in N + 1 dimensions, of the mean-squared error versus the N coefficient values. If the adaptive filter is nonrecursive, the error surface is a quadratic function of the coefficients. With recursive adaptive filters, the error surface is not quadratic and may even have local minima. In this correspondence we discuss the nature of the recursive error surface and give examples of conditions under which local minima may exist. We conclude with a discussion of the effects of the nonquadratic error surface on gradient-search algorithms for recursive adaptive filters.

Adaptive frequency sampling filters

Abstract: We present two new structures for adaptive filters based on the idea of frequency sampling filters and gradient based estimation algorithms. These filters have a finite impulse response (FIR) and can be thought of as attempting to approximate a desired frequency response at given points on the unit circle. The filters operate in real time with no batch processing of signals as is the case when using the discrete Fourier transform. They result in a marked reduction in dimension of the timedomain problem of fitting an Nth-order FIR transversal filter to a collection of length 2 transversal filters and further to a collection of N scalar filters. The advantages of this are then discussed.

Some statistical properties of median filters

Abstract: Some statistical properties of median filters are analyzed. It is shown that median filters can remove impulsive plus Gaussian white noise better than Hanning filters when the amplitude of the impulses is large or the energy of the Gaussian noise is relatively low. It is also shown that, unlike linear filters, median filters preserve sharp changes in signals when the noise energy is not too high.

Self-tuned filters

Abstract: A scheme for the tuning of filters is proposed. In it, filters are time-shared between a signal path and a tuning path, so that each filter in a system is individually tuned. An accurate system frequency response can thus be achieved which does not rely on tight matching between filters.

Properties of Digital Smoothing Polynomial (DISPO) Filters

Abstract: Digital filters for spectrometric applications are compared with the classical RC filter. Properties discussed include noise reduction, line shift, and conservation of line moments. For Gaussian and Lorentzian lines, signal deformation and change of half-width as a function of time constant and line width are calculated for several filter types. Using accuracy, sensitivity, and scan speed as criteria, it is shown that a fourth-order digital smoothing polynomial (DISPO) filter is better by typically 1 or even 2 orders of magnitude than the RC filter. Since a real time implementation of these filters is possible, they can directly replace RC filters in all spectrometric applications.

Error surfaces of recursive adaptive filters

Abstract: For an adaptive filter with N adjustable coefficients or weights, the "error surface" is a plot, in N+ I dimensions, of the mean-squared error versus the N coefficient values. If the adaptive filter is nonrecursive, the error surface is a quadratic function of the coefficients. With recursive adaptive filters, the error surface is not quadratic and may even have local minima. In this correspondence we discuss the nature of the recursive error surface and give examples of conditions under which local minima may exist. We conclude with a discussion of the effects of the nonquadratic error surface on gradient-search algorithms for recursive adaptive filters.

Adaptive frequency sampling filters

Abstract: We present two new structures for adaptive filters based on the idea of frequency sampling filters and gradient based estimation algorithms. These filters have a finite impulse response (FIR) and can be thought of as attempting to approximate a desired frequency response at given points on the unit circle. The filters operate in real time with no batch processing of signals as is the case when using the discrete Fourier transform. They result in a marked reduction in dimension of the time-domain problem of fitting an Nth-order FIR transversal filter to a collection of length 2 transversal filters and further to a collection of N scalar filters. The advantages of this are then discussed.

Convergence properties of an adaptive digital lattice filter

Abstract: Convergence properties of a continuously adaptive digital lattice filter used as a linear predictor are investigated for both an unnormalized and a normalized gradient adaptation algorithm. The PARCOR coefficient mean values and the output mean-square error (MSE) are approximated and a simple model is described which approximates these quantities as functions of time. Calculated curves using this model are compared with simulation results. Results obtained for a two-stage lattice are then compared with the two-stage least mean-square (LMS) transversal filter algorithm, demonstrating that it is possible but unlikely for the transversal filter to converge faster than the analogous lattice filter.

Filters comprising reactive components, and a method of determining impedances thereof

Abstract: A filter comprising reactive components which define pass and stop bands has the impedances of the components selected to minimize the sum |Zin-Z2|+|Zout-Z1| in the pass band, where Zin and Zout are respectively the input and output impedances of the filter and Z1 and Z2 are nominal termination impedance at respectively the input and output of the filter. The filter characteristic in the pass band is thereby made substantially independent of the actual termination impedances of the filter in use. The design is exemplified by a 5th order low pass filter for connection in the voice signal path of a telephone subscriber line.

A note on the use of windows for two-dimensional FIR filter design

Abstract: Using a one-dimensional window as a prototype, a two-dimensional window may be formulated having either a square region of support or a circular one. In this paper we compare the effects of using each of these window formulations for 2-D FIR filter design and present formulas for estimating filter order in terms of design specifications, using a Kaiser window as a prototype.

Parallel-Coupled-Line Filters for Inverted-Microstrip and Suspended-Substrate MIC's

Abstract: An improved design procedure is outlined for parallel-coupled-resonator filters in non-homogeneous media such as inverted microstrip and suspended stripline. A modified filter section is used, comprising a parallel-coupled o.c. section in cascade with short un-coupled line sections. Much improved characteristics, both in the pass-band and in the stop-band may be achieved, and the difficulties associated with the unequal even-and odd-mode velocities of the non-homogeneous coupled sections are largely overcome.

Image gradient detectors operating in a partitioned low-pass channel

Abstract: An electronic image processing system, for image enhancement and noise suppression, from signals representing an array of picture elements, or pels. The system is of the kind providing a feature-enhanced output by the addition of outputs of a high-pass filter acting as image-feature detector and a complementary low-pass filter. The low-pass filter also acts as an image-feature detector and includes a prefilter (130 and FIG. 22) and a sub-sampling filter (32) based on a set of weighting patterns in the form of sparse matrices (FIG. 23). The sub-sampling filter in a bandpass channel of the low-pass filter (206 and FIG. 28) comprises a pair of gradient detectors (210, 220, 230, 240 and FIGS. 32, 33, 34, 35) arranged back to back.

Design of active filters independent of first-and second-order operational amplifier time constant effects

Abstract: A practical method of designing active filters in which the transfer function is independent of both first- and second-order operational amplifier time constant effects is presented. Neither matched operational amplifiers nor a tuning procedure dependent on an active parameter is required. The active portion of these filters is universal and readily integrable since it is comprised of conventional operational amplifiers and resistors. The method can be used to design a filter with any realizable transfer function of any order. Several new filters obtained from this method are introduced and evaluated both theoretically and experimentally. These configurations contain popular passive structures and the new universal active circuits. The significant improvements in filter performance of these new filters is demonstrated in this evaluation.

Simplified MOS switched capacitor ladder filter structures

Abstract: A general technique for time-sharing amplifiers to reduce die area in switched capacitor ladder filters is described and illustrated with a fifth-order elliptic low-pass ladder filter requiring only three operational amplifiers. Techniques for synthesizing filters with maximum passband accuracy in the presence of parasitic capacitances are presented, and verified with two versions of the same fifth-order design integrated in a standard NMOS process. Passband accuracies of better than 0.1 dB have been achieved using only 0.3 pF unit-sized capacitors. The dynamic range is 75 dB.

Frequency-domain considerations of LSV digital filters

Abstract: The present paper develops a framework for the analysis and synthesis of linear shift-variant (LSV) digital filters in the frequency domain. First, LSV digital filters are theoretically modeled by the successive use of linear shift-invariant (LSI) filters. On the basis of the model, we present an interpretation of shift-variant spectral modification or filtering. Further, shift-variant digital filtering is discussed in relation to the notions of the short-time spectrum and the generalized frequency function. In addition, we propose an efficient implementation procedure which reduces the number of filter coefficients and the amount of computation. The effectiveness of LSV digital filters in processing time-varying signals is demonstrated by experimental verification.

A synthesis of frequency domain filters for time delay estimation

Abstract: A simple model for two-channel delay estimation filtering is presented. The problem is subdivided into three classes based on initial assumptions. General filters described in the frequency domain are presented as solutions to these specific classes. It is shown that many of these filters, which include the "Wiener" least-squares estimation filter and classical, matched detection filter, can be derived as specific cases of a very general ideal filter form. We call this general ideal filter the weighted distortion balance filter. Relationships between a standard set of ideal filters and some filters previously proposed in the literature for delay estimation are discussed. An illustrative example is presented to compare the delay estimated from the use of various filters.

Linear phase filter with self-equalized group delay

Abstract: In modern communication systems, it has become important to provide filters, and in particular bandpass filters, which can provide substantially uniform group delay across the bandwidth of the filter while still achieving good amplitude response. In this regard, it is particularly desirable to substantially reduce the large variations in the phase characteristics found in conventional bandpass filters at the nominal band edge of the filter. To accomplish this, a filter is provided having at least one pair of lattice arms coupled in parallel to one another between the input and output of the filter. Each of the lattice arms includes a plurality of resonant LC resonators, each of the resonators having a different resonant frequency than the center frequency of the filter. In particular, within the bandwidth of the filter, the exponential damping coefficients for the resonators in each arm are set to decay at the same rate. This desired decay can be accomplished by exponential sizing of the components.

A prototype switched-capacitor voltage-wave filter realized in NMOS technology

Abstract: Presents a novel approach to the realization of monolithic filters. The method is based on using sampled analog signals and is related to the wave digital filter in its design techniques. The eventual monolithic realization in NMOS technology is in the form of a switched-capacitor structure. The design is exact and there is no requirement for a high relative clock frequency. Only unity-gain buffers are required, as opposed to high-gain differential-input operational amplifiers, and so the technique is well suited to CMOS technology. Performance is determined by capacitance ratios and the design is optimally insensitive to parameter variations. Capacitance ratios are moderate relative to those encountered in existing switched-capacitor filters. Results are presented for a prototype integrated circuit design containing fifth- and seventh-order low-pass Chebyshev filters with a designed cutoff at one-eighth clock frequency. The responses achieved for the prototype design show excellent agreement with the theory.

Symmetry exploitation in the design and implementation of recursive 2-D rectangularly sampled digital filters

Abstract: It is well known that exploiting the symmetries existing in the frequency response of 2-D filters results in a reduction in design and implementation complexities. In this paper we explore how to exploit a general class of symmetries in the design and implementation of recursive two-dimensional (2-D) filters. Three types of filters are being studied, namely, causal, factorizable noncausal, and unfactorizable noncausal. The capability of each of these filters to exhibit symmetries is discussed. A method is then proposed to obtain the required form of the filter transfer function for exhibiting the different types of symmetries. Examples are solved both to illustrate the proposed method and to compare the performance of these filter types for the same implementation complexity.

On the stability of digital filters designed using the concept of generalized-immittance convertor

Abstract: A method for designing wave digital filters using the concept of generalized-immittance convertor has been reported recently by Antoniou and Rezk. In this paper is derived a stability criterion for this new type of filter. It is shown that the requirement for ensuring the absence of limit cycles in these filters is the same as the one proposed by Fettweis and Meerkcotter for the conventional wave digital filters.

Error feedback in normal realization of recursive digital filters

Abstract: Normal realizations of narrow-band low-pass filters are shown to be suitable for the use of error feedback to reduce roundoff noise. The use of a simple error feedback structure at each summing node, requiring only a single storage register and no multipliers, reduces the high roundoff noise that is normally associated with narrow-band low-pass filters.

Variable Bandpass Filters Using Varactor Diodes

Abstract: A rectangular waveguide type variable bandpass filter for the 4-GHz bandpass has been proposed and tested. The bandpass width varies from 260 MHz to 1.02 GHz for a filter using varactor diodes. Two microstrip variable bandpass filters for the 6-GHz and 4-GHz bands are also proposed and tested. The passband width varies from 310 MHz to1.24 GHz for a varactor-diode coupled filter, and it varies from 380 MHz to 2.18 GHz for a filter which is composed of low-pass and high pass filters connected in cascade. The center frequency of the three filters can be changed arbitrarily.

Integrated multi-stage electrical filter

Abstract: A multi-legged transformer core is employed to integrate into a single structure the discrete magnetic components required to form a multiple stage inductor-capacitor filter. The device may be employed in low pass filters used with inverters to remove high order harmonic frequency signals. The filter of the present invention is particularly useful in discharge lamp ballast circuits.

Design of finite impulse response digital filters with nonlinear phase response

Abstract: Most of the existing literature on FIR digital filters is concerned with linear-phase (LP) filters. However, several papers have appeared on the subject of nonlinear-phase (NLP) filters, mainly proposing methods for designing minimum-phase filters, or approximating a desired phase response. In this paper, an investigation is made of one such method, based upon a selection of zeros from a prototype LP filter. It is shown that with respect to minimizing the order of a filter subject to given gain response specifications, this is the most efficient method for designing FIR filters. Coefficient quantization error is analyzed for filters generated by this method. A practical comparison is given between the resulting filter and the corresponding minimal order LP filter. It is shown that while most LP filters can be implemented more efficiently than NLP filters by taking into account the symmetry of their coefficients, for filters with very wide passband and for certain special purpose filters such as CCD and those used for filtering a delta-modulated or ADPCM signal, an NLP implementation is usually more efficient. In addition, an alternate design algorithm is proposed for NLP filters which decreases ripple magnitude. The resulting filters, while not of minimal order, can be efficiently implemented by decomposing the filter into LP stopband and NLP passband sections, which is especially attractive for narrow passbands.

Digital filter models for media having linear with frequency loss characteristics

Abstract: Linear with frequency acoustic loss characteristics have been observed for soft biological tissue when probed with diagnostic ultrasound and for Plexiglas when used for underwater applications. Two digital filter models are described that exhibit this linear characteristic. The first filter model, designed using a frequency sampling technique, produces a symmetric unit sample response and a linear with frequency phase characteristic. Several features of this linear phase model, however, are found to be incompatible with the physical mechanisms producing the loss. Upon resolving these incompatibilities, a second filter model is obtained which has a minimum phase transfer function. Experimental verification, using Plexiglas as an acoustic propagation medium, demonstrates the superiority of the minimum phase model for describing the physical mechanisms which produce a linear with frequency loss characteristic.

General spectral transformations for digital filters

Abstract: Constantinides [1] gave a method of transforming a prototype low-pass digital filter into a high-pass, bandpass, or bandstop filter by conformally mapping the complex variable Z-1. We generalize here his method to transformations from prototype low-pass filters to filters containing any number of pass- and stopbands. The computation procedure involves only the solution of a set of simultaneous linear equations, the number of which is related only to the number of band-edges and not to the order of the prototype filter.

Convergence properties of an adaptive digital lattice filter

Abstract: Convergence properties of a continuously adaptive digital lattice filter used as a linear predictor are investigated for both an unnormalized and a normalized gradient adaptation algorithm. The PARCOR coefficient mean values and the output mean-square error (MSE) are approximated and a simple model is described which approximates these quantities as functions of time. Calculated curves using this model are compared with simulation results. Results obtained for a two-stage lattice are then compared with the two-stage least mean-square (LMS) transversal filter algorithm, demonstrating that it is possible but unlikely for the transversal filter to converge faster than the analogous lattice filter.