Showing papers on "Prototype filter published in 1986"

Stack filters

Abstract: The median and other rank-order operators possess two properties called the threshold decomposition and the stacking properties. The first is a limited superposition property which leads to a new architecture for these filters; the second is an ordering property which allows an efficient VLSI implementation of the threshold decomposition architecture. Motivated by the success of rank-order filters in a wide variety of applications and by the ease with which they can now be implemented, we consider in this paper a new class of filters called stack filters. They share the threshold decomposition and stacking properties of rank-order filters but are otherwise unconstrained. They are shown to form a very large class of easily implemented nonlinear filters which includes the rank-order operators as well as all compositions of morphological operators. The convergence properties of these filters are investigated using techniques similar to those used to determine root signal behavior of median filters. The results obtained include necessary conditions for a stack filter to preserve monotone regions or edges in signals. The output distribution for these filters is also found. All the stack filters of window width 3 are determined along with their convergence properties. Among these filters are found two which we have named asymmetric median filters. They share all the properties of median filters except that they remove impulses of one sign only; that is, one removes only positive going edges, the other removes only negative going edges, while the median filter removes impulses of both signs. This investigation of the properties of stack filters thus produces several new, useful, and easily implemented filters.

Frequency-response masking approach for the synthesis of sharp linear phase digital filters

Abstract: If each delay element of a linear phase low-pass digital filter is replaced by M delay elements, an (M + 1) -band filter is produced. The transition-width of this (M + 1) -band filter is 1/M that of the prototype low-pass filter. A complementary filter can be obtained by subtracting the output of the (M + 1) -band filter from a suitably delayed version of the input. The complementary filter is an (M + 1) -band filter whose passbands and stopbands are the stopbands and passbands, respectively, of the original (M + 1) -band filter. If the frequency responses of the original ( M + 1) -band filter and its complementary filter are properly masked and recombined, narrow transition-band filter can be obtained. This technique can be used to design sharp low-pass, high-pass, bandpass, and bandstop filters with arbitrary passband bandwidth.

Continuous-time MOSFET-C filters in VLSI

Abstract: The desirable features of fully integrated, VLSI-compatible continuous-time filters are discussed. A recently proposed integrated continuous-time filter technique is reviewed, in which MOS transistors are used in place of resistors along with the nonlinearity cancellation and on-chip automatic tuning. The filters obtained using this technique are compared to switched-capacitor filters, digital filters, and continuous-time filters using different techniques. Representative experimental results are given, demonstrating the high performance that can be achieved.

Varactor Tuned Bandpass Filters Using Microstrip-Line Ring Resonators

Abstract: This paper describes the fundamental characteristics and experimental results of newly developed tunable bandpass filters using rnicrostrip-line ring resonators. The experimental filter has a steeper attenuation slope and less circuit instability than conventional filters, and it seems to be a suitable filter for MICs or MMICs.

Continuous-time MOSFET-C filters in VLSI

Abstract: The desirable features of fully integrated, VLSI-compatible continuous-time filters are discussed. A recently proposed integrated continuous-time filtering technique is reviewed, in which MOS transistors are used in place of resistors along with nonlinearity cancellation and on-chip automatic tuning. The filters obtained using this technique are compared to switched-capacitor (SC) filters, digital filters, and continuous-time filters using different techniques. Representative experimental results are given, demonstrating the high performance that can be achieved.

Passive and Active Filters: Theory and Implementations

Abstract: Fundamentals of Network Synthesis Filter Approximation and Frequency Transformations Passive Filter Synthesis Design of Resistively Terminated Networks Active Filter Synthesis: Fundamentals Sensitivity The Active Biquad Realization of Active Two-Port Networks Design of Broadband Matching Networks Theory of Passive Cascade Synthesis General Theory of Compatible Impedances. Appendixes.

Active filter for reactive power and harmonics compensation

Abstract: This paper describes a three-phase active filter aimed at the compensation of reactive power and current harmonics of a symmetrical load. The filter employs an inductor for energy storage and a two-quadrant PWM bridge converter.

Signal analysing and synthesizing filter bank system

Abstract: In a signal analysing and synthesizing filter bank system, the analysing bank receives a signal sampled at the rate fe and produces N contiguous subbank signals sampled at the rate fe /N. From the subband signals the synthesizing bank must recover the incoming signal. These filter banks are formed by modulation of a prototype filter by sinusoidal signals which, for subband k (O≦k≦N=1), have a frequency (2k30 1)fe /(4N) and respective phases +(2k+1)π/4 and -(2k+1)π/4 for the analysing and synthesizing banks. These signals are furthermore delayed by a time delay (Nc -1)/2fe), where Nc is the number of coefficients of the prototype filter. Preferably, the analysing bank is realized by the cascade arrangement of an N-branch polyphase network (12) and a double-odd discrete cosine transform calculating arrangement (14) and the synthesizing bank is realized by the cascade arrangement of a double-odd discrete cosine transform calculating arrangement (15) and an N-branch polyphase network (17).

The use of Symmetrics in the Design of Multidimensional Digital filters

Abstract: Symmetries have been successfully used in the design of two-dimensional digital filters. This results in a reduction of the required optimization and in better response characteristics of the filters designed. Since the number of parameters of a filter increases by an order of magnitude for every increase by one of the filter dimensionality, the use of symmetries in the design of multidimensional filters is essential. This paper introduces various multidimensional symmetries and applies them to the computer-aided design of some useful three-dimensional digital filters.

Switched-capacitor high-Q bandpass filters for IF applications

Abstract: Experimental results for a sixth-order switched-capacitor bandpass filter with a selectivity Q of 55 at a center frequency of 3.1 MHz are presented. A simple noise analysis of active bandpass filters composed of coupled identical resonators is introduced to explain the dynamic range reduction in high-Q active filters resulting from loose high-Q couplings between resonators. Theoretical predictions of noise in coupled resonator-type bandpass filters agree well with measured results. The prototype chip occupies 2 mm/SUP 2/ and dissipates 45 mW with a single 5-V supply.

Analysis of coefficient quantization errors in state-space digital filters

Abstract: In this paper, we examine the sensitivity of state-space digital filters to coefficient quantization errors. A cost function using an L p norm criteria is used to measure the deviation in frequency response of the filter. The cost function is used to indicate the connection between roundoff noise and filter sensitivity to coefficient quantization errors. Then based on this cost function it is shown that low roundoff noise state-space digital filters, filters in balanced coordinates, and filters based on polynomials orthogonal on the unit circle have low sensitivity to coefficient quantization errors. Furthermore, the sensitivity of the filters is shown to depend on the second-order modes thereby exhibiting certain robustness and immunity to filter bandwidth.

Active filters for 1 MHz power circuits with strict input/output ripple requirements

Abstract: As the switching frequency of a dc-dc converter is raised from 100 kHz to 1 MHz, one would expect the energy storage elements of the circuit to be correspondingly reduced in size. If the circuit must meet input and output ripple requirements of the nature of MIL-STD-461B CEO3, however, the advantage of a higher switching frequency is significantly offset by the higher filter attenuation levels required. This paper discusses the use of active filters, in conjunction with minimal passive filters, to achieve the required ripple levels with a much smaller overall volume. Working active filters with gains in excess of 100 at 1 MHz are also presented.

Non linear adaptive filters

Abstract: A non-linear adaptive filter is described having a linear filter connected in parallel with a non-linear filter. The linear filter provides fast adaption until it has modelled the linear contribution of each coefficient. Thereafter the non-linear filter continues to adapt until the error signal has been reduced to an acceptable level. In a preferred embodiment a plurality of unit delay devices provide taps to sub-processing units each of which is arranged to be adapted according to a linear algorithm during an initial part of a training period, and to a non-linear algorithm after said initial part. The filter has particular use in echo cancellers for data modems.

Computer-Generated and Phase-Only Synthetic Discriminant Function Filters,

Abstract: The laboratory realization of smart matched spatial filters (specifically snythetic discriminant functions) is considered and studied via computer simulation. The realization of these distortion-invariant correlator functions by computer-generated hologram and other methods is considered. Multiclass filters are addressed in this work. Phase-only filters and filters with a quantized number of amplitude and phase levels are considered. Phase-only filters are found to perform well for some imagery, but filters with some amplitude response are preferable. We also considered the number of phase and magnitude levels required and found four levels of each to be adequate.

Multimemory block structure for implementing a digital adaptive filter using distributed arithmetic

Abstract: A new structure of adaptive transversal filters with a large number of taps is described. It is based on the use of the distributed-arithmetic technique without any multiplier in the realisation of the filter function. In this structure, the N filter taps are divided into M blocks, each with R taps. These M blocks operate simultaneously and thus achieve a high-speed signal processing capability. This type of adaptive filter can easily be implemented by using microprocessor or transistor-transistor logic integrated circuits. A simplified hardware prototype module suitable for 8- and 16-point transversal adaptive filters, using microprocessor and simple peripheral interface circuitry, is presented. Results from this prototype demonstrate the basic feasibility of this structure for implementing digital adaptive filters with a large number of taps.

Broadband RFI power line filter

Abstract: A broadband RFI power line filter in which emission filters designed for the 10 KHz to 30 MHz range are combined with high frequency filter elements so as to produce a filter in the 10 KHz to 1000 MHz range. The filter is divided into sections which are constructed in separate electrically isolated compartments so as to increase the attenuation which can be obtained with filters which are not separated from each other and electrically shielded compartments and the level of attenuation which must be achieved from each compartment is reduced in the invention.

The analog median filter

Abstract: Discrete median filters are a special class of ranked-order digital filters used for smoothing signals. In this paper, the analog median filter is defined and proposed for analysis of the standard discrete median filter in cases with a large sample size or when the associated statistics would be simpler in the continuum. Discrete filters are shown to be a subclass of analog filters. Also, an equivalence among analog filters and limits of discrete filters is established. Finally, several stochastic interpretations of the analog median filter are presented including necessary and sufficient conditions on input processes which guarantee the existence of output distributions for multiple passes of the analog median filter

Analytic design of optimal FIR narrow-band filters using Zolotarev polynomials

Abstract: An analytic method for designing narrow-band FIR filters using Zolotarev polynomials, which are extensions of Chebyshev polynomials, is proposed. These filters are optimal in the Chebyshev sense. Bandpass and bandstop narrow-band filters, as well as low-pass and high-pass filters, can be designed by this method. The design procedure and related formulas are presented. Design examples are included to show the properties of these filters.

Design of circularly symmetric low-pass two-dimensional FIR digital filters

Abstract: A technique for choosing the coefficients of the first-order McClellan transformation for the design of circularly symmetric twodimensional FIR filters from one-dimensional filters is described. It is shown that, by a simple analytic technique, it is possible to determine the coefficients so as to make the passband boundary of a two-dimensional filter to approximate a circle with a high degree of accuracy. The behavior of the transformed filter in the transition region is discussed and a design example to meet a given set of specifications is given.

Design and Implementation of Linear Phase FIR Filters for Biological Signal Processing

Abstract: This paper presents a new method for designing digital linear phase, finite impulse response filters with loose frequency response characteristics, but with good time resolution as is required in biological signal conditioning. The design is very simple and has been used with success in the microcomputer implementation of filters for the automated processing of electroencephalographic (EEG) data. Examples and a discussion of possible filter implementations are included.

Theory and realization of M-D nonlinear digital filters

Abstract: A class of M-D nonlinear digital filters based on the volterra series is presented. Both the theoretical and the implementation standpoints are considered, along with a performance example.

Suspended Substrate Filters and Multiplexers

Abstract: The different configurations which may be used for broadband lowpass and highpass filters will be discussed for applications from 0.5 - 40 GHz. Diplexer design and their application to broadband multiplexers will be considered with examples of practical devices. Due to the limitations of loss, narrow band bandpass and bandstop designs have restricted applications. However, one application in contiguous switched multiplexers will be considered in detail where the dissipation loss characteristic plays a very important role in the overall device performance. Such an N channel switched multiplexer may assume anyone of 2N states within 30 ns. Finally, other special classes of networks such as constant phase difference networks will be discussed with applications to frequency discriminators.

Recursive center-frequency adaptive filters for the enhancement of bandpass signals

Abstract: In some signal enhancement and tracking applications, where a priori information regarding the signal bandwidth and spectral shape is available, it is suggested to use a recursive center-frequency adaptive filter instead of a fully adaptive filter. A new adaptive algorithm, namely, the recursive maximum-mean-squares (RMXMS) algorithm, is developed based on the gradient ascent technique for the implementation of these filters. An adaptation mechanism based on the Gauss-Newton algorithm is also presented. This class of filters is found to have several advantages which include faster convergence and lesser computational complexity compared to the fully adaptive filters.

Design of doubly-complementary IIR digital filters, using a single complex allpass filter

Abstract: It is shown that a wide class of real-coefficient, doubly-complementary IIR transfer-function pairs can be implemented by means of a single complex allpass filter. For a real input sequence, the real part of the output sequence of the complex allpass filter corresponds to one of the transfer functions G(z) (for example, low-pass), whereas the imaginary part of the output sequence corresponds to its "complementary" filter H(z) (for example, highpass). Since the resulting implementation is structurally lossless, G(z) and H(z) have very low passband-sensitivity. Numerical design examples are included to demonstrate the ideas.

Evanescent Mode-Serrated Ridge Waveguide Bandpass Harmonic Filters

Abstract: A new class of bandpass harmonic filters with spurious free and extremely wide stopband response is presented. The design concept is based on using evanescent mode waveguide sections as impedance K-inverter, while using serrated ridge waveguide sections as series resonators. Measured results of narrow and wideband filters show an excellent agreement with the theoretical design.

Equaliser with adjustable band filters and a digital filter suitable for use in the equaliser

Abstract: A manually operated or automatic equalizer (1) with adjustable band filters for varying the frequency characteristic of an electric signal presented to an input (2) and for applying a frequency characteristic-adapted electric signal to an output (3). The equalizer has a plurality of band filters (F1 to Fn) with their bands located adjacent to one another in a given frequency range. The distance (frequency separation) between the central frequencies of filters having neighboring bands is greater than the distance between the central frequencies of one third octave filters having neighboring bands located at at least approximately the same frequencies as those of the band filters (Fi). At least those band filters (F1 to Fm) whose bands are located in the low-frequency part (f<450 Hz) of the frequency range have a central frequency (fci) which is variable. A digital filter which can be used as a band filter in the equalizer is also described.

Some new aspects of filters for filter banks

Abstract: The present paper deals with the design of filters for use in filter banks, i.e., with certain constraints on their transition bands. The goal is to achieve a baseband frequency response of a back-to-back analysis/synthesis filter bank system that is ideally flat in magnitude and linear in phase. Using the proposed method, this is possible for conventional polyphase filter banks, single-sideband (SSB) filter banks, and generalized quadrature mirror filter (GQMF) banks. The method provides several options, one of them even yielding subsystems with linear phase. In contrast to most previous approaches to this topic, the described design procedure is an analytical one and is not based on optimizations or iterative algorithms. Furthermore, two families of window sequences are proposed for use in the design procedure mentioned above, but are also suitable for any windowed Fourier filter design.

Digital filter for processing two-dimensional digital image

Abstract: Digital filters can be very effectively used for filtration of two-dimensional image. A special one of the digital filters operates to add together all data over a filter length, multiply the addition result by a predetermined set value, and to effect addition or subtraction between the multiplication result and data at a point of interest to be filtered. In case of the addition, the filter works as a low-pass filter. In case of the subtraction, the filter provides for a high-pass filtration. Such digital filter has such a problem that the number of additions is increased as the filter length is larger. Therefore, it is essential in the digital filters to reduce the necessary number of additions.

Rotation Invariant Composite Filters

Abstract: The Composite filter of Caulfield and Maloney, a linear combination of matched filters, is an optimum estimator in the mean-square sense. Composite filters may be made rotation invariant by using Circular harmonic filters as components. With a finite number of Circular harmonic filters, the output of the composite filter is invariant under rotation of the input only if all Circular harmonic filters have the same order.