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Showing papers on "Filter design published in 1989"


Journal ArticleDOI
TL;DR: The contribution of this work is to quantify and justify the functional relationships between image features and filter parameters so that the design process can be easily modified for different conditions of noise and scale.

284 citations


Journal ArticleDOI
G. K. Kaleh1
TL;DR: By using a pulse-amplitude-modulation representation of binary continuous-phase- modulation signals, the authors develop a novel optimum Viterbi sequence detector and a near-optimum Viterba receiver with low complexity.
Abstract: By using a pulse-amplitude-modulation representation of binary continuous-phase-modulation signals, the authors develops a novel optimum Viterbi sequence detector and a near-optimum Viterbi receiver with low complexity. For modulation index 0.5, where a linear receiver can be used, a minimum-mean-squared-error linear receiver filter is derived. The performance of all of these is analyzed, using the Gaussian minimum-shift-keying signal (GMSK) for illustration. It is shown that a GMSK receiver consisting of two matched filters and a four-state Viterbi algorithm performs with less than 0.24-dB degradation compared with the optimal receiver. The linear receiver is optimum for all values of E/sub b//N/sub 0/ (bit-energy-to-noise one-sided spectral density ratio). A design method for its filter is given. The filter is equivalent to a cascade of a matched filter and a Wiener filter estimator. Both upper and lower bounds for the bit-error probability are calculated. Simulation results which confirm the analysis are given. >

236 citations


Journal Article
TL;DR: In this article, a method for designing an equalization filter for a sound-reproduction system by adjusting the filter coefficients to minimize the sum of the squares of the errors between the equalized responses at multiple points in the room and delayed versions of the original electrical signal is presented.
Abstract: A method is presented for designing an equalization filter for a sound-reproduction system by adjusting the filter coefficients to minimize the sum of the squares of the errors between the equalized responses at multiple points in the room and delayed versions of the original electrical signal

176 citations


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


Proceedings ArticleDOI
03 Oct 1989
TL;DR: A six-step SPUDT synthesis procedure is presented which includes a new method for determining transduction and reflection weighting functions that satisfy a desired amplitude and phase response, and achieve low insertion loss, achieve low triple transit, and allow use of simple matching networks.
Abstract: A comprehensive discussion of the design synthesis challenges for single-phase unidirectional transducers (SPUDTs) is presented. The complexity of SPUDT filter design results from the fact that a SPUDT deliberately includes reflections internal to the transducer to cancel the effects of regeneration reflection. This destroys the validity of the impulse-response model which is the basis for all conventional SAW (surface acoustic wave) transducer designs. Although a recent generalization of the impulse response model includes effects of uniform internal reflections, it is of limited usefulness, because a high-performance SPUDT filter requires both weighted reflection and weighted transduction. Presented is a six-step SPUDT synthesis procedure which includes a new method for determining transduction and reflection weighting functions that satisfy a desired amplitude and phase response, achieve low insertion loss, achieve low triple transit, allow use of simple matching networks, and properly account for the interactions between transduction and reflection effects under matched conditions. Fundamental insertion-loss limitations for SPUDT filters are discussed and areas for future SPUDT research are outlined. Good experimental performance is demonstrated for a bandpass filter that uses a new SPUDT structure, the EWC/SPUDT, the geometry of which is also shown. >

169 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
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: Comparison to the McClellan-Parks algorithm for minimax equiripple filters shows that both are optimal in the sense of different minimum norms of the error function, but much better performance is obtained with the proposed approach in most of the frequency band, except in the narrowband region near the cutoff edge.
Abstract: An effective approach is proposed for designing higher order differentiators by the eigenfilter method By minimizing a quadratic measure of the error in the frequency band, an eigenvector of an appropriate matrix is computed to get the filter coefficients This method is not only simple and fast, but also optimal in the least-squares sense Comparison to the McClellan-Parks algorithm for minimax equiripple filters shows that both are optimal in the sense of different minimum norms of the error function, but much better performance is obtained with the proposed approach in most of the frequency band, except in the narrowband region near the cutoff edge >

112 citations


Journal ArticleDOI
TL;DR: Two simple methods for retrieving a single sinusoid corrupted with noise are proposed, based on the lattice form realization of an adaptive infinite-impulse-response (IIR) notch filter, which have considerable potential in adaptive notch filter applications, especially when the input signal-to-noise ratio is low.
Abstract: Two simple methods for retrieving a single sinusoid corrupted with noise are proposed. They are based on the lattice form realization of an adaptive infinite-impulse-response (IIR) notch filter. The IIR filter is a cascade of second-order all-pole and all-zero filters, and the coefficients of the finite-impulse-response (FIR) section are adapted. The proposed algorithms keep the poles of the filter inside the unit circle. The computer simulation results show that the algorithms have considerable potential in adaptive notch filter applications, especially when the input signal-to-noise ratio is low. >

111 citations


Journal ArticleDOI
TL;DR: The proposed algorithm deals directly with the complex error function, which depends linearly on the coefficients of the filter to be designed, and is minimized in the Chebshev sense using a generalization of the Remez exchange algorithm.
Abstract: The long-standing problem of approximating a complex-valued desired function with a finite impulse-response (FIR) filter is considered. It is formulated as an equalization to be solved using complex-valued filters. The proposed algorithm deals directly with the complex error function, which depends linearly on the coefficients of the filter to be designed. The magnitude of this error function is minimized in the Chebshev sense using a generalization of the Remez exchange algorithm. The method can be used to design complex- or real-valued-selective systems as well. The well-known design of optimal FIR filters with linear phase is included here as a special case. >

102 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. >

Patent
Ygal Arbel1
12 Sep 1989
TL;DR: In this article, a full-duplex digital speakerphone with room echo cancellation adaptive filter was proposed, where the adaptive filter coefficient initialization is performed in a half-duplication mode and switches to fullduplex when filter coefficients are adapted.
Abstract: A full-duplex digital speakerphone 10 includes a transmit signal path having an output coupled to a telephone trunk and a receive signal path having an input coupled to the telephone trunk and an output coupled to a loudspeaker. The speakerphone further includes a room echo cancellation adaptive filter 56 and a trunk echo cancellation adaptive filter 66. Serially coupled within the transmit signal path is a selective suppression block 50 for suppressing a component of a Mu-Law or an A-Law quantization error signal. A second selective suppression block 52 is serially coupled within the receive signal path. Suppression of non-linearities due to Mu-Law or A-Law signal conversion is also accommodated by providing a non-linear signal processing block 40 at an input to an adaptive filter and an optional non-linear signal processing block at an output of the adaptive filter. Each of the blocks emulates and compensates for signal converter non-linearity. The speakerphone facilitates adaptive filter coefficient initialization by beginning a call in a half-duplex mode and switching to full-duplex when filter coefficients are adapted. The speakerphone also has a variable adaptation step size which is a function of a short-term estimate of signal power within the associated transmit or receive signal paths.

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. >

Journal ArticleDOI
TL;DR: In this article, a closed-form least-squares solution to the design problem of two-dimensional real zero-phase finite-impulse-response (FIR) filters with quadrantally symmetric or antisymmetric frequency response is obtained.
Abstract: A closed-form least-squares solution to the design problem of two-dimensional real zero-phase finite-impulse-response (FIR) filters with quadrantally symmetric or antisymmetric frequency response is obtained. An in-depth study of the matrices involved in the development of the design technique reveals a number of useful properties. It is shown that these properties lead to an optimal analytical solution for the filter coefficients, making it unnecessary to use the time-consuming methods of optimization, matrix inversion, and iteration. Because of the reduced order of the matrices involved, their specific characteristics, and the analytical approach, the computational complexity is greatly reduced. Simplicity and efficiency of the design technique is illustrated through examples. The results in terms of error in frequency response compare favorably with those obtained by using other techniques. It is shown that the design time using the proposed technique is significantly smaller than that required by the I/sub p/-optimization technique or weighted least-squares technique using Harris' ascent algorithm or modified Lawson's algorithm. >

Proceedings ArticleDOI
14 May 1989
TL;DR: A time-domain input-shaping technique that works like a filter that is designed to eliminate decaying sinusoidal resonance responses is compared to a variety of filtering techniques and is shown to perform more effectively than any of them.
Abstract: The authors evaluate a signal generation technique for commanding high-performance machines to move without exciting residual vibration. This time-domain input-shaping technique works like a filter that is designed to eliminate decaying sinusoidal resonance responses. It is compared to a variety of filtering techniques and is shown to perform more effectively than any of them. In particular, frequency-domain filter design techniques are less effective for shaping vibration-reducing inputs to systems. The three-impulse shaping sequence offers significant performance advantages in that moves require less time and result in little or no residual vibration. In addition, the response is not sensitive to variations in the system parameters. >

Patent
14 Mar 1989
TL;DR: In this paper, a technique for the sub-band decomposition and reconstruction of video signals is described, where each field of a video signal is decomposed into sub-bands utilizing a polyphase filter bank unit including an infinite impulse response all-pass filter with coefficients that are powers of two.
Abstract: A technique for the sub-band decomposition and reconstruction of video signals is disclosed. In an illustrative embodiment of the present invention, each field of a video signal is decomposed into sub-bands utilizing a polyphase filter bank unit including an infinite impulse response allpass filter with coefficients that are powers of two. To reconstruct the original image, an FIR is utilized which approximates an IIR allpass filter This eliminates the need for a full field memory in the reconstruction process.

Journal ArticleDOI
TL;DR: In this article, a mismatched filter design is introduced whereby the receiver filter coefficients are optimized with the property that all sidelobes of the cross-correlation function (CCF) are zero.
Abstract: A mismatched filter design is introduced whereby the receiver filter coefficients are optimized with the property that all sidelobes of the cross-correlation function (CCF) are zero. Some results are given for binary sequences and the associated mismatched filters. For each length N>2 there exist at least one sequence and an associated mismatched filter with an ideal impulselike CCF, which means that there are no sidelobes at all in the receiver output signal. >

Proceedings ArticleDOI
30 Jun 1989
TL;DR: In this paper, a robust, nonlinear, order statistic type filter is proposed for point-like feature detection in infrared systems, known as median subtraction filtering, which exhibits high-pass filter characteristics without the usual ringing associated with linear highpass filters.
Abstract: The nonstationarity of infrared interference backgrounds which prevents the implementation of the usual optimum linear filtering techniques makes clutter suppression signal processing for point target detection in infrared surveillance systems a challenging and difficult problem. Hence, more robust filtering schemes are sought which will perform well in structured backgrounds where the underlying probability distribution defining that structure is not well known or characterized. This paper investigates a promising candidate spatial filter for point-like feature detection in infrared systems. The technique, known as median subtraction filtering, is a robust, nonlinear, order statistic type filter which exhibits highpass filter characteristics without the usual ringing associated with linear highpass filters. A quantitative analysis of the statistical properties of the median subtraction filter is presented, including analytic expressions for the output distribution of the filter (thus analytic expressions for the probability of detection and probability of false alarm), its autocorrelation function and spectral density function. Performance results of a signal processing simulation comparing a median subtraction filter with an adaptive linear filter of the LMS type using actual infrared video as input are also included.

Journal ArticleDOI
R.L. Cupo1, Richard D. Gitlin2
TL;DR: These systems, which adapt their structure to match the spectral properties of the impairments, avoid the conflict between a wide bandwidth ( to track fast jitter) and a narrow bandwidth (to minimize output noise) inherent in most carrier recovery loops.
Abstract: Adaptive or predictive carrier recovery systems, which are essential in high-performance quadrature-amplitude-modulated (QAM) data communications systems to correct for phase jitter and frequency offset, are considered. Analytical and experimental results are presented for two structures that implement a predictive carrier recovery system. These systems, which adapt their structure to match the spectral properties of the impairments, avoid the conflict between a wide bandwidth (to track fast jitter) and a narrow bandwidth (to minimize output noise) inherent in most carrier recovery loops. Such a system increases the likelihood that very bandwidth-efficient modems (e.g., 7 b/s/Hz for 19.2 kb/s voiceband modem applications) can provide reliable transmission in the presence of severe phase jitter and frequency offset. In particular, the predictive carrier recovery systems can track sinusoidal jitter present at more than one frequency as well as generalized time-varying phase jitter processes. Both finite-impulse-response (FIR) and infinite-impulse-response (IIR) adaptive phase tracking systems are considered. Prior limitations on adaptive IIR filters are overcome by designing a structure that is guaranteed to be stable and to possess only a global minimum as the filter coefficients converge to their desired values. >

Patent
22 Mar 1989
TL;DR: In this article, a hybrid despread and demodulation receiver for low symbol rate communications employs a passive matched filter to remove a "short" coding portion of a composite spreading code that has been used to spread the data signal.
Abstract: A hybrid despread and demodulation receiver for low symbol rate communications employs a passive (SAW) matched filter to remove a "short" coding portion of a composite spreading code that has been used to spread the data signal. The composite spreading sequence is formed by multiplying different length coding sequences, thereby obtaining an overall signal processing operator the duration or symbol span of which is sufficient to maintain a high signal processing gain, but is considerably longer than can be processed using a practical sized passive (e.g. SAW) filter design. The design of the receiver takes advantage of the fact that the relatively short sequence can be despread using a practical SAW structure and is comprised of a hybrid signal processor, the front end of which contains a compact SAW matched filter and the downstream end of which is implemented using analog processing components. The matched filter removes the relatively short spreading sequence from the received signal and feeds its output to a mixer, which combines the output of the matched filter with the longer coding sequence to complete the despreading process. The despread signal is then differentially coherently decoded and coupled to an integrate and dump circuit, which accumulates the energy in successive long code symbol intervals in order to determine the value of the respective data bits.

Journal ArticleDOI
TL;DR: The theory for just such an optimal filter design given an arbitrary region of realizability is presented and a fast algorithm is presented to implement the theory.
Abstract: Almost all coherent pattern recognition architectures are based on optical correlation of the input with a designed filter. However, the filter can be implemented via many different media, and each medium will impose different realizability constraints on the filter. That is, different media will have different regions of physical realizability. In the past, there has not been much work addressing the problem of designing an optimal filter given an arbitrary region of realizability. This paper presents the theory for just such an optimal filter design. A fast algorithm is presented to implement the theory. The algorithm is demonstrated with two examples.

Journal ArticleDOI
TL;DR: Numerical evaluations using the image of a tank indicate that using such a filter can provide an improvement in SNR of ~5 dB over the conventional binary phase-only filter (BPOF), superior to the 1-dB improvement obtained for that image by varying the threshold line angle (TLA) in filter binarization.
Abstract: An efficient algorithm for designing a ternary valued filter yielding the highest signal to noise ratio (SNR) is utlined. Numerical evaluations using the image of a tank indicate that using such a filter can provide an improvement in SNR of ~5 dB over the conventional binary phase-only filter (BPOF). This is superior to the 1-dB improvement obtained for that image by varying the threshold line angle (TLA) in filter binarization. Simulation results are presented. They agree with the numerically computed SNRs.

Patent
27 Dec 1989
TL;DR: In this article, a normalized frequency domain Least Means Square filter is proposed, where the power estimate is incorporated directly into the filter algorithm as a data-dependent, time-varying stochastic feedback coefficient.
Abstract: A normalized frequency domain Least Means Square filter. The feedback coefficient for each frequency bin is adjusted separately in the filter by use of an input power estimate. The power estimate is incorporated directly into the filter algorithm as a data-dependent, time-varying stochastic feedback coefficient. The filter is particularly useful in applications in which the input signal has large spectral variations, which could lead to filter instabilities in some frequency bins if a single feedback coefficient were employed in all frequency bins.

Proceedings ArticleDOI
13 Dec 1989
TL;DR: In this paper, an approach for using optimally robust detection filters to generate analytic redundancy is presented, where the design of the filter is formulated as an optimization problem and its solution shows that the optimally-robust detection filter consists of a bandpass filter and a linear system which is obtained by solving a general eigenvalue problem.
Abstract: An approach is presented for using optimally robust detection filters to generate analytic redundancy. By introducing an appropriate criterion the design of the filter is formulated as an optimization problem. Its solution shows that the optimally robust detection filter consists of a bandpass filter and a linear system which is obtained by solving a general eigenvalue problem. The algorithm for designing this filter is therefore computationally simple and systematic. Investigations on the physical core of the fault detection procedure are carried out. It is shown that the quality of the fault detection is dependent on the coupling strength of the faults with respect to the system output. It is demonstrated that the optimal detection filter presented is a generalization of the detection filter of R.V. Beard (1971) and H.L. Jones (1973). >

Journal ArticleDOI
TL;DR: Experimental results show that the modified Wiener filter outperforms its linear counterpart (neglecting the impulse-response noise) and gives better restoration results than the Backus-Gilbert technique.
Abstract: The restoration of images distorted by systems with noisy point spread functions and additive detective noise is considered. The criterion considered for the restoration is based on the Wiener technique. The proposed Wiener-based filter is iterative in nature. The overall computation of this modified Wiener filter can be carried out in the frequency domain using the fast Fourier transform and circulant matrix approximation. Experimental results show that the modified Wiener filter outperforms its linear counterpart (neglecting the impulse-response noise). The modified Wiener filter also gives better restoration results than the Backus-Gilbert technique. The Wiener-based filter is found to be computationally robust and inexpensive. >

Proceedings ArticleDOI
01 Oct 1989
TL;DR: In this paper, an extended Kalman filter is used to identify the parameters of an induction motor using measurements of the stator voltages, currents, and rotor speed, and the results demonstrate that the filter is capable of identifying the parameters.
Abstract: An extended Kalman filter is used to identify the parameters of an induction motor using measurements of the stator voltages, currents, and rotor speed. A model of the induction motor in the state space and the Kalman filter algorithm are shown. This filter is applied to the parameter identification of an inverter-fed induction motor. A simple and practical method of setting the covariance matrices of the noises, which are important in the Kalman filter algorithm, is proposed. The starting values of the state and parameter vectors as well as the covariance matrix of the estimation error are then shown, and, finally, the results of parameter identification are shown. The results demonstrate that the filter is capable of identifying the parameters. >

Journal ArticleDOI
TL;DR: An analytical method is developed for estimating the probability density function (pdf) of this envelope for different kinds of filter responses and for realistic combinations of phase noise severity and filter bandwidth.
Abstract: Consideration is given to the case of an optical pulse containing phase noise which is passed through either an optical filter or (following heterodyne lightwave detection) an electrical filter Because of the phase noise, the envelope of the filter output at any instant is a random variable An analytical method is developed for estimating the probability density function (pdf) of this envelope for different kinds of filter responses and for realistic combinations of phase noise severity and filter bandwidth Obvious applications are to detection analyses of coherent lightwave systems, wherein finite laser linewidths constitute an important source of impairment For each of the several types of filters considered, the envelope PDF can be accurately fitted by an exponential function approximation, where the decay constant is related in a simple way to known system parameters >

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
23 May 1989
TL;DR: In this paper, a differential mode rejection network (DMRN) is proposed, which separates common mode noise from differential mode noise in a line impedance stabilization network (LISN)-based conducted emissions setup.
Abstract: A differential mode rejection network (DMRN), a device which separates common mode noise from differential mode noise in a line impedance stabilization network (LISN)-based conducted emissions setup, is described. Although it is not a filter, its function is analogous to that of a filter. The DMRN filters out differential mode, and passes common mode unfiltered. The differential mode is attenuated by more than 50 dB, and the common mode noise is attenuated by less than 4 dB. Applications of the DMRN are discussed for filter design and troubleshooting, and electromagnetic interference source suppression is briefly described. Mathematical analysis and hardware implementation of the devices are explained. >

Journal ArticleDOI
P.A. Ruetz1
TL;DR: In this paper, a set of real-time 20-MHz digital signal processor (DSP) chips has been designed, fabricated, and tested, including a 64-tap programmable FIR (finite impulse response) filter, a 1024-tap binary filter and template matcher, and an eight-line 512-pixel video line delay.
Abstract: A set of four real-time 20-MHz digital signal processor (DSP) chips has been designed, fabricated, and tested. The chips include a 64-tap programmable FIR (finite impulse response) filter, a 1024-tap binary filter and template matcher, a 64-tap rank-value filter, and an eight-line 512-pixel video line delay. The circuits were implemented in a 1.5- mu m CMOS process and are fully functional with a 20-MHz clock rate. The processors have reconfigurable windows to allow processing on both one-dimensional and two-dimensional data. The FIR filters can be used in multiprocessor systems to increase the window size and the data precision. >