Showing papers on "Filter design published in 1979"

The processing of hexagonally sampled two-dimensional signals

Abstract: Two-dimensional signals are normally processed as rectangularly sampled arrays; i.e., they are periodically sampled in each of two orthogonal independent variables. Another form of periodic sampling, hexagonal sampling, offers substantial savings in machine storage and arithmetic computations for many signal processing operations. In this paper, methods for the processing of two-dimensional signals which have been sampled as two-dimensional hexagonal arrays are presented. Included are methods for signal representation, linear system implementation, frequency response calculation, DFT calculation, filter design, and filter implementation. These algorithms bear strong resemblances to the corresponding results for rectangular arrays; however, there are also many important differences. Some comparisons between the two methods for representing planar data will also be presented.

MOS switched-capacitor filters

Abstract: In the past several years, much progress has been made in bringing the economies of integrated-circuit technology to bear on the realization of voiceband frequency selective filters. This paper will review one approach to this problem, the use of switched-capacitor techniques. The paper emphasizes the practical aspects of switched-capacitor filter design under the constraints imposed by MOS integrated-circuit technology. The basic operation of switched-capacitor filters is reviewed, followed by a discussion of the properties of the various circuit building blocks in MOS technology. Finally, a summary of several filter organizations which appear to be well suited to switched-capacitor implementation is presented.

Optimal synthesis of second-order state-space structures for digital filters

Abstract: Sufficient conditions are derived for a second-order statespace digital filter with L_2 scaling to be optimal with respect to output roundoff noise; and from these, a simple synthesis procedure is developed. Parallel-form designs produced by this method are equivalent to the block-optimal designs of Mullis and Roberts. The corresponding cascadeform designs are not equivalent, but they are shown, by example, to be quite close in performance. It is also shown that the coefficient sensitivities of this structure are closely related to its noise performance. Hence, the optimal design has low-coefficient sensitivity properties, and any other low-sensitivity design is a good candidate for near-optimal noise performance. The uniform-grid structure of Rader and Gold is an interesting and useful case in point.

An Optimal Frequency Domain Filter for Edge Detection in Digital Pictures

Abstract: Edge detection and enhancement are widely used in image processing applications. In this paper we consider the problem of optimizing spatial frequency domain filters for detecting edges in digital pictures. The filter is optimum in that it produces maximum energy within a resolution interval of specified width in the vicinity of the edge. We show that, in the continuous case, the filter transfer function is specified in terms of the prolate spheroidal wave function. In the discrete case, the filter transfer function is specified in terms of the sampled values of the first-order prolate spheroidal wave function or in terms of the sampled values of an asymptotic approximation of the wave function. Both versions can be implemented via the fast Fourier transform (FFT). We show that the optimum filter is very effective for detecting blufred and noisy edges. Finally, we compare the performance of the optimum edge detection filter with other edge detection filters using a variety of input images.

A convergence proof for a hyperstable adaptive recursive filter (Corresp.)

Abstract: Hyperstability, a concept from nonlinear stability theory, is used to develop a real-time adaptive recursive filter useful in a nonstationary environment.

Optimum second order digital filter

Abstract: A second order digital filter utilizing six processor operations, two add instructions, two shift instructions and two store instructions. No multipliers are required. The filter is used as a digital filter in a servo loop having a Z transform of, G(Z)=4 (1-Z-1)+Z-2.

New digital filter for the analysis of experimental data

Abstract: The extremely simple mathematical technique called ’’straightening through smoothing,’’ which is a numerical frequency filter, is generalized in order to provide a transmission function having any shape. This frequency filter requires such a small memory that it can be performed using a minicomputer or even a programmable hand held calculator and the number of channels used is not limited to a power of 2, as in the case of the fast Fourier transform. For some filtering functions the number of operations required is smaller than with the fast Fourier transform.

A new approach for the design of digital interpolating filters

Abstract: The paper presents a new class of interpolators characterized by the property that the mean power of their error sequence, as a function of frequency, approximates zero in the Chebyshev sense. The design method, which is based on some observation and newly found general properties, will be described in some detail. An example shows the special properties of these interpolators in comparison with former results. Measurements done with a practical implementation are presented as well. A design chart for these filters is provided.

Optimum Filter Design for a Single-Phase Solid-State UPS System

Abstract: A systematic design procedure for the output filter of a singlephase uninterruptible power supply (UPS) system is developed. The basic specifications for the UPS system are first established. Four different output filter configurations are then analyzed and compared for sinusoidal pulsewidth and single-pulse modulated inverter output (i.e., filter input) voltage. On the basis of the above comparison, ``optimum'' filters are selected for both modulation techniques. Using a minimization function for filter cost and size, a set of filter design parameters corresponding to each type of modulation are obtained on the per unit basis. The theoretical results are verified on an experimental breadboard utilizing a current commutated thyristor inverter. Finally, the overall filter design procedure is outlined and a design example is presented.

Statistical design of autoregressive-moving average digital filters

Abstract: Procedures are presented for the systematic design of digital filters that contain poles and zeros. The procedures are simple, fast, and effective. All of the important algorithms are of the Levinson-type. The first key idea in the paper is that one may begin a design by posing a linear prediction problem for a stochastic sequence. The second is that a high-order "whitening" filter may be constructed for this sequence and "inverted" to yield a high-order all-pole filter whose spectrum approximates the spectrum of the stochastic sequence. The third key idea is that the all-pole filter may be used to generate consistent unit-pulse and covariance sequences for use in the Mullis-Roberts algorithm. This algorithm is then used to obtain a low-order digital filter, with poles and zeros, that approximates the high-order all-pole filter. The results demonstrate that the Mullis-Roberts algorithm, together with the design philosophy of this paper, may be used with profit to reduce filter or stochastic model complexity and to design spectrum-matching digital filters.

Loss sensitivities in singly and doubly terminated filters

Abstract: The loss of a conventional doubly terminated filter is far less sensitive than the loss of a singly terminated filter to errors in the component values. This property, first pointed out in 1966, has motivated many new high quality RC-active and digital filter design techniques. It is not so widely appreciated, however, that this superiority of the doubly terminated filter also holds for errors in the terminating resistances as well as for errors in the components inside the filter. The nature of these sensitivities in both types of filter is discussed in some detail, and Illustrated by means of a numerical example.

Input-Filter Design for Switching Regulators

Abstract: The interaction between the input filter and the control loop of switching regulators often results in detrimental effects, such as loop instability, transient response, and audio-signal-rejection rate, etc. A small-signal average model is derived to investigate these effects. Design constraints of an input-filter and switching-regulator system are formulated. An optimum low-pass and light-weight filter configuration is proposed.

Decimator apparatus for decreasing the word rate of a digital signal of the type employed in digital telephone systems

Abstract: There is disclosed a decimator structure which incorporates the cascade of an FIR filter with a low pass recursive filter. The input to the decimator is obtained from a high rate analog to digital converter. The output from the decimator is a low rate digital signal having an increased word length. The decimator serves to reduce the word rate and increase the word length of the output digital signal of the analog to digital converter. In this manner, the low rate digital signal at the output of the decimator can be easily accommodated by the telephone system. The decimator described does not require conventional hardware multipliers and hence, affords efficient and economical circuit components which can be implemented by conventional integrated circuit techniques. A register further reduces the pulse rate for transmission.

Optimal design of FIR digital filters with monotone passband response

Abstract: The application of linear programming to the design of FIR digital filters with constraints on the derivative of the frequency response is described. Numerical considerations in the implementation are discussed and a program is given with examples for the design of filters with optional monotone response in passbands. The method provides the user with an additional degree of flexibility over the Remez exchange algorithm.

Miniature programmable transversal filter using CCD/MOS technology

Abstract: This paper describes the operational features and performance of an integrated-circuit programmable sampled-analog data filter in transversal form using CCD/MOST technology. Reasons behind the particular choice of filter architecture for a prototype realization and its comparison with other reported designs in this technology are discussed, with particular emphasis placed on a novel MOST multiplier array implementation. The performance characteristics of a prototype 64-point filter design based on this approach is detailed in the context of frequency- and matched-filtering, and a module of 256 points using four cascaded filters is also described. Techniques for optimizing the inherent performance limits of these filter types under microprocessor control are suggested, via the iterative adaption of the filter impulse response, and results are given to show the improvement obtained. Finally, the potential of this miniature integrated-circuit filter for sonar type applications is briefly discussed.

Band-pass filter circuit

Abstract: A band-pass filter circuit comprising a band-pass filter for removing noise modulation components from an input signal comprising a carrier sine wave, a phase detector for detecting a phase difference between input and output signals of the band-pass filter, a loop filter supplied with the output from the phase detector, and automatic control means for effecting control by the output from the loop filter so that a difference between the frequency of the input sine wave and the center frequency of the band-pass filter is reduced to zero. For carrier recovery in a burst mode, the loop filter is selected for each particular case so that high-speed pulling-in is possible even if a narrow-band filter is employed as the band-pass filter for the removal of noise components. Also, high-speed and high-precision pulling-in is achieved when many bursts of different frequencies are applied in one frame period. Further, when common and individual frequency variations occur at the same time, high-speed pulling-in is also achieved for each burst.

Charge-transfer filtering

Abstract: Linear discrete-time (sampled-data) filters can be implemented in monolithic form on an MOS chip using a variety of recently developed and newly emerging techniques without requiring analog-to-digital conversion. An analog signal sample can be represented by an isolated quantity of charge and such packets of charge can be stored, transferred, and manipulated in other ways to perform signal processing operations. Best known of these techniques is the use of a charge-coupled device (CCD) to operate as an analog shift register. A simple modification of the CCD shift register allows the realization of transversal filters. Other techniques can implement recursive filter operations offering a great flexibility for filter design. The possibilities and limitations of charge-transfer filtering are reviewed and examined.

Adaptive digital echo cancellation circuit

Abstract: The present invention discloses an adaptive digital echo cancellation circuit including a finite impulse response digital filter, wherein the digital filter coefficients are adapted by continuous updating to compensate for telephone subscriber line echo conditions to enable the digital filter to continuously simulate the instantaneous subscriber line echo. The continuous coefficient update is provided by a correlator which includes provision for introducing non-linearities into the PCM transmitted and received signals, in parallel with the speech path, to derive a simplified digital representation of the PCM speech, thereby reducing the signal processing hardware required by the correlator to derive the updated filter coefficients.

Practical design of switched-capacitor networks for integrated circuit implementation

Abstract: Switched-capacitor (s.c.) networks are of great interest owing to their compatibility with m.o.s.i.c. technology. It is shown how s.c. networks can be derived directly from continuous active RC networks. The corresponding network functions in z are obtained from those in s by using a simple `p-transformation?. It is shown that s.c. networks can also be designed directly. The resulting networks are often more versatile than those derived from another circuit type or technology. Numerous examples of both approaches to s.c. network and filter design are given and experimental results presented. Excellent agreement between theory and practice has been obtained.

Filter and system incorporating the filter for processing discrete samples of composite signals

Abstract: A system for processing discrete digitized samples representing composite signals utilizing a filter which eliminates a periodic signal component from the composite signal. The filter receives and stores consecutive digital sample representations of the composite signal and, for each received sample representation, provides a digital average representation of the values of a selected number of the received digital sample representations which define a zero average value of the periodic signal component. In one embodiment of the signal processing system, the filter is arranged in circuit with digital delays and digital signal combining and differencing circuits to form a digital color television signal dropout compensator, which is adaptable for use in NTSC, PAL, PAL-M, or other television standard systems. In a dropout compensator adapted for NTSC color television signals, the filter receives the digital composite television signal and eliminates the chrominance component therefrom, leaving only the luminance component at its output. A following digital subtractor is coupled to subtract the luminance component provided by the filter from the received digital composite television signal and provide the chrominance component at its output. The separated chrominance component is phase adjusted on consecutive television lines and recombined with the separated luminance component provided by the filter for substitution in the television signal in place of the dropout affected portion thereof. The dropout compensator also includes a digital delay of one horizontal line period through which the television signal components are passed to provide the delay necessary for substituting television signal information from a prior horizontal line.

Optimal data transmission filters

Abstract: A common requirement in the design of multilevel data transmission filters is to provide a relatively large impulse response at t= t_M and near-zero responses at t = t_M + k ( k = any integer), in order to minimize intersymbol interference. This paper gives an analytic design procedure for a realizable filter that meets this requirement in an optimal manner and also optimizes other frequency and time domain criteria. The results are then extended to handle nonimpulsive inputs (which includes nonideal channels).

Tracking filter for FM threshold extension

Abstract: A tracking filter suitable for use in the IF stage of FM receiver circuitry incorporates a bandpass filter the components of which yield a characteristic that permits the filter to provide substantially the entirety of a required 90° phase shift for phase detection control circuitry that rapidly actively tunes capacitive components of the filter without imparting unacceptable group delay behavior to the configuration. The filter is a two-pole pair, single zero at the orgin, bandpass filter inserted in the signal path between a pair of signal power splitters. Part of the input signal is applied from the first splitter to one input of a phase comparator the other input of which is derived from the output signal via the second power splitter. The output of the phase comparator is applied through a broadband control amplifier to actively adjust the center frequency of the bandpass filter. Advantageously, this configuration enables the filter to have its center frequency effectively coincide with the instantaneous frequency of the deviating input signal, whereby an increase in performance over conventional filter approaches is obtained.

Determination of resistivity sounding filters by the wiener‐hopf least‐squares method*

Abstract: It has been found that the Wiener-Hopf least-squares method is a very successful tool for the determination of resistivity sounding filters. The values of the individual filter coefficients differ quite appreciably from those obtained by the Ghosh procedure. These differences in the filter coefficients, however, have only a negligible effect on the output of the filter. It seems that these differences in the coefficients correspond to a filter function of a rather narrow frequency band around the Nyquist frequency, which is only very weakly present in the input and output functions.

Frequency-responsive filter for flowmeter transmission system

Abstract: A transmission system for a vortex-shedding or swirl-type flowmeter whose meter signal lies in the low-frequency range and is therefore of low resolution, the meter signal being subject to jitter. In order to provide a jitter-free, high resolution output signal, the system includes an input amplifier responsive to the meter signal, the output of which is applied through a tracking filter to a Schmitt trigger. The trigger converts the meter signal into pulses of the same frequency which are fed into a frequency multiplier to produce a relatively high-frequency output signal of good resolution. The tracking filter serves to characterize the amplifier response to reduce the amplification gain thereof below a selected frequency level. The tracking filter operates in conjunction with a comparator assembly which compares an analog voltage whose magnitude depends on the meter signal frequency with a series of progressively increased reference voltages, each representing a predetermined frequency level. The assembly functions to render the filter operative in a stepwise manner, whereby when the meter frequency reaches any one of the predetermined frequency levels, the filter then acts effectively to reduce the amplification gain of the input amplifier to attenuate frequencies below that level, thereby minimizing the transmission of noise components in the meter signal.

Least-mean-square spatial filter for IR sensors.

Abstract: A new least-mean-square filter is defined for signal-detection problems The technique is proposed for scanning IR surveillance systems operating in poorly characterized but primarily low-frequency clutter interference Near-optimal detection of point-source targets is predicted both for continuous-time and sampled-data systems

Optimal design and comparison of wide-band digital on-line differentiators

Abstract: The problem of wide-band digital on-line differentiators is of increasing importance. The frequency approach is used to design and compare different filters. First, an error criterion that represents numerical error is derived, combining amplitude and phase errors. Conditions that ensure good low frequency behavior are included. Two design procedures to obtain the filter coefficients are discussed: an indirect approach based on modeling with polynomials and modified splines and a direct approach using an optimization scheme. Finally, the frequency characteristics of the main optimal and suboptimal wide-band differentiators are compared. The full-band requirement restricts improvements to about 33 percent with respect to the first-order difference formula.

New integer programming scheme for nonrecursive digital filter design

Abstract: A new mixed-integer linear programming objective function for optimising an f.i.r. linear-phase digital filter is presented. In comparison with the conventional objective function, the new one has the advantages of reducing the number of delays and/or the coefficient wordlength.

Maximum entropy (adaptive) filtering applied to radar clutter

Abstract: An adaptive digital filtering scheme is presented which uses a lattice structure for adaptive prediction and elimination of radar clutter, using Burg's algorithm for the computation of the lattice coefficients. This method computes a minimum-phase prediction error filter directly from the radar data, while not having the end-bias problems common to many filtering schemes. This permits quick adaptation to changing clutter conditions. An integration decay constant allows the filter to adapt to longer duration signals (clutter) while passing shorter duration signals (targets). The filter design is described and experimental results are discussed.

Variable digital filters having a recursive structure

Abstract: Variable digital filters allow the frequency characteristics of a filter to be manipulated. Frequency transformations have been successfully used to obtain a variable filter from a nonrecursive prototype. However, this technique cannot be applied directly to filters having a recursive structure. A variation of this method is presented which allows variable filters to be obtained from both recursive and nonrecursive prototypes. This method has special advantages when the prototype has an equal number of poles and zeros (e.g., digital filters obtained by the bilinear transformation from an analog filter).