Showing papers on "Filter design published in 1984"

Adaptive notch filtering for the retrieval of sinusoids in noise

Abstract: In this paper, an adaptive notch filter is developed (employing a frequency domain and time domain analysis) for the enhancement and tracking of sinusoids in additive noise, colored or white. The notch filter is implemented as a constrained infinite impulse response filter with the constraint enforced by a single parameter termed the debiasing parameter. The resulting notch filter requires few parameters, facilitates the formation of the desired band rejection filter response, and also leads to various useful implementations (cascade, parallel). For the adaptation of the filter coefficients, the stochastic Gauss-Newton algorithm is used. The convergence of this updating procedure is established by studying the associated differential equation. Also, it is shown that the structure present in the problem enables truncation of the gradient, thereby reducing the complexity of adapting the filter coefficients. Simulation results are presented to substantiate the analysis, and to demonstrate the potential of the notch filtering technique.

Multirate filter designs using comb filters

Abstract: Results on multistage multirate digital filter design indicate most of the stages can be designed to control aliasing with only slight regard for the passband which is controlled by a single stage compensator. Because of this, the aliasing controlling stages can be made very simple. This paper considers comb filter structures for decimators and interpolators in multistage structures. Design procedures are developed and examples shown that have a very low multiplication rate, very few filter coefficients, low storage requirements, and a simple structure.

A flexible sampling-rate conversion method

Abstract: A digital resampling method is proposed which allows non-uniform and time-varying resampling. The method is based on interpolated look-up in a large table of filter coefficients. One filter table handles all conversion factors. Formulas are given for determining the size of look-up table needed for a given precision requirement.

Performance Analysis of Narrow-Band Interference Rejection Techniques in DS Spread-Spectrum Systems

Abstract: Linear least squares estimation (LLSE) techniques can provide an effective means of suppressing narrow-band interference in direct sequence (DS) spread-spectrum systems. In the results presented here, analytical expressions for bit error rate are derived for two DS spread-spectrum systems under the conditions of either tone or narrowband Gaussian interference. It is shown that the most common LLSE filter design can lead to performance inferior to that of various other filter designs. However, results are also presented demonstrating that an LLSE filter design motivated by the structure of the maximum-likelihood receiver leads to consistently superior performance. The performance of a system using this new design criterion is compared with that of an approximation to the maximum-likelihood (ML) receiver for the tone interference model and with that of the exact ML receiver for the Gaussian interference. Finally, it is shown that the bit error rate estimate obtained from application of a Gaussian approximation for the test statistic is overly pessimistic for the systems studied here.

New quadrature mirror filter structures

Abstract: This paper introduces new quadrature mirror filter (QMF) structures for the frequency domain analysis and synthesis of digital signals. The conventional QMF technique is first extended to cover complex quadrature mirror filters (CQMF) in which a digital signal is split into N adjacent complex subbands where the real and imaginary parts are subsampled by 1/2N with respect to the original signal. The computational complexity of QMF banks is then analyzed and a new scheme which reduces the computational complexity by about a factor of two over conventional QMF implementations is proposed. Finally, the filter design tradeoffs are discussed and the microprogramed implementation of QMF banks is evaluated.

An adaptative, nonlinear edge-preserving filter

Abstract: An adaptative filter whose main feature is to preserve edges and impulses present in the signal is analyzed by the computation of the mean-square error (MSE) of its output sequence. The filter in its more general form is highly nonlinear, resembling the M-type estimators used in robust statistics. A simplified form used here allows the exact computation of the MSE when the filter length is finite. This MSE can be compared to the ones obtained for a median filter and a mean filter. It is shown that for a wide range of the filter and signal parameters such as filter length, edge heights, and impulse width, the performance of the filter proposed in this paper is superior to the other filters mentioned above. An additional advantage of the simplified version of the filter is that in most cases, its computation amounts to a linear adaptative averaging. This contrasts with the amount of calculation required to implement the median filter and any other filter based on the order statistics of the measured samples.

Analysis and performance evaluation of an adaptive notch filter

Abstract: An adaptive notch filter is derived by using a general prediction error framework. The proposed infinite impulse response filler has a special structure that guarantees the desired transfer characteristics. The filter coefficients are updated by a version of the recursive maximum likelihood algorithm. The convergence properties of the algorithm and its asymptotic behavior are discussed, and its performance is evaluated by simulation results.

Optimal design of periodically time-varying and multirate digital filters

Abstract: A description of time-varrying digital filters is developed that uses a transfer function of two frequency variables. This proves to be particularly powerful for periodically time-varying systems and results in the sources and effects of aliasing and folding being explicitly shown. This method is used to allow the optimal frequency domain design of these systems using time-invariant methods. An application to a narrow-band filter design gives up to a factor of three improvement in efficiency.

Additional properties of the phase-only correlation filter

Abstract: Computer simulations of the scaling and rotation sensitivity of a phase-only filter were performed, showing that it is much more sensitive to such input variations than is a classical matched filter Values of the peak correlation spot power versus both rotation angle and scale factor are presented for both filter types Several theorems are derived for calculating the optical efficiency of any filter in both input space and frequency space

Broad-Band Millimeter-Wave E-Plane Bandpass Filters (Short Papers)

Abstract: An accurate method to obtain starting estimates for an E-plane bandpass filter CAD program recently developed by Shih, Itoh, and Bui is presented. Results agree very well with exact values for filter design with relative bandwidths exceeding 10 percent at W-band and D-band and 20 percent at lower frequencies.

Process and apparatus for translating the sampling rate of a sampling sequence

Abstract: An input sampling sequence with an input sampling rate is translated into an output sampling sequence with a selectable output sampling rate in a sampling filter. A circuit for forming the time difference of the sampling points forms from the given input sampling rate and the desired output sampling rate a signal corresponding to the time difference and this is used in a translation circuit for converting into data for characterizing the filter coefficients. The sampling filter is conditioned by the selected coefficients from the translation circuit. The invention is suitable for the transmission of sampled data, particularly sampled rates between two systems operating at different clock frequencies.

Fast block data processing via a new IIR digital filter structure

Abstract: A new structure for realizing IIR digital filters is introduced based on the idea of processing sequences by blocks. It is shown that this structure exhibits high inherent parallelism that is ideally suited for VLSI implementation or multimicroprocessor systems. This enables high-speed processing with the number of multiplies and additions per output sample much less than those of the block-state or canonical filter realizations. The roundoff noise level in the output of the new filter structure is derived and compared to the noise levels of the block-state and canonical forms. Further, it is shown that the new filter structure will present no scaling problems if the canonical filter is scaled. Finally, the extension of the new filter structure to the realization of periodically time-varying digital filters is also presented.

Design of three- dimensional recursive digital filters

Abstract: Three types of three-dimensional (3-D) recursive digital filters are first defined under different radial symmetry constraints in the 3-D coordinate axes: the symmetrical filter, the nonsymmetrical filters I and II. A design technique is then outlined for these 3-D recursive digital filters whose magnitude responses can be decomposed into several cubic pass- and stopbands. The filter designed by the present approach can be realized by cascading and paralleling of the well-known one-dimensional (1-D) component transfer functions so that the stability test is simple and the filter can be implemented easily. Three examples are included to illustrate the design procedure for each type of 3-D filter.

A new structure for narrow transition band, lowpass digital filter design

Abstract: In this paper, a new class of lowpass linear phase FIR filters is introduced. It is shown that lowpass filters with narrow transition bands can be realized efficiently by a structured form composed mainly of a few small FIR filters. The modular structure is suited for an implementation by a fast short convolution algorithm or a few single-chip filter IC's. A design procedure is considered in some detail and illustrated by an example. The properties and performance of the filter are discussed by analysis as well as design results. A brief discussion on its implementations concludes the paper.

Detailed analysis of nonidealities in MOS fully integrated active RC filters based on balanced networks

Abstract: The paper gives a detailed account of the effects produced by practical nonidealities in fully integrated balanced RC filters, which are implemented using MOS transistors as replacements for resistors (MOSFET-capacitor continuous-time filters). It is found that the operation of such filters is insensitive to all typical nonidealities, with the exception of the intrinsic distributed parasitic capacitances of the transistors; there are cases when the latter have to be taken into account in the filter design.

Interpolating filter arrangement with irrational ratio between the input and the output sampling frequencies

Abstract: Interpolating time-discrete filter arrangement for increasing the sampling frequency of a time-discrete signal from f 1 to f u , f u not being a rational multiple of f i . For the generation of the required filter coefficients this filter arrangement comprises clock pulse generators which produce input clock pulses ki(q) at a rate f i and output clock pulses ku(n) at a rate f u . It also comprises a coefficients generator (4) in which a deviation component d(q) is calculated which indicates the relationship between the time interval T d(q) located between an input clock pulse k i (q) and the immediately preceding or the immediately subsequent output clock pulse ku(n) and the time interval T u =1/f u between two consecutive output clock pulses ku(n). In response to this deviation component the filter coefficients generator produces W filter coefficients, the w th filter coefficient (w = 0, 1, 2, 3, ... W-1) being equal to a (d(q),w) and being defined by the expression Herein the function h(v) represents the impulse response of a FIR filter and v a continuous variable in the interval -∞ < v < ∞.

On the design of optimal state-space realizations of second-order digital filters

Abstract: Simple design equations are derived for optimal (minimum roundoff noise) state-space realizations of second-order digital filter sections with complex poles. These relations permit easy computation of filter coefficients and unit noise gain directly from the filter transfer function. New parameters are identified that are easily computed from the transfer function, are invariant under certain frequency-scaling transformations, and which can be used to simply describe all of the design variables of optimal second-order filters, including state covariance matrix, unit noise matrix and second-order modes. The design relations are illustrated by computing the unit noise gains and eigenvalue sensitivities of optimal parallel realizations of Chebyshev lowpass filters.

Performance of minimum-variance deconvolution filter

Abstract: Recently, we observed zero phase and undershoot patterns in data processed by a minimum-variance deconvolution (MVD) filter. These observations motivated a careful analysis of the MVD filter, which, as we demonstrate in this paper, explains both the zero phase and undershoot patterns. This analysis also connects the MVD filter with the well-known prediction-error filter [6], and Berkhout's two-sided least-squares inverse filter [7]. We show that the performance of the MVD filter depends heavily on the bandwidth of the source wavelet and signal-to-noise ratio, and only slightly on data length.

An Efficient 200-290-GHz Frequency Tripler Incorporating a Novel Stripline Structure

Abstract: This paper describes a broadly tuneable frequency tripler which can provide more than 2-mW output power at any frequency between 200 and 290 GHz. It is derived from an earlier narrow-band prototype design, with the major improvements being the use of a new low-pass filter design implemented using a novel suspended substrate stripline structure, an optimized waveguide transformer, and a lower loss contacting output backshort.

Design of 2-D separable-denominator recursive digital filters

Abstract: This paper presents a technique for finding the coefficients of a two-dimensional (2-D) recursive digital filter having a separable denominator which gives the best approximation, in the least squares sense, to a desired 2-D impulse response over a finite interval. All the coefficients in the filter are found by iteratively solving linear equations. Since the resulting filter has a separable denominator, it is easy to check the stability and the implementation is simpler. Two examples are given to illustrate the utility of the proposed technique.

Computer tomography system and method for operating same

Abstract: A computer tomography system has an X-ray source and a radiation receiver having an array of individual detectors each forming an electrical signal corresponding to the received radiation intensity, the X-ray source being rotated about a subject for transradiating a layer of the subject from different directions, and a measured value processing circuit to which the output signals of the individual detectors are supplied and which identifies therefrom attenuation values of predetermined points in the transradiated plane of the subject for generating a display image. The measure value processing circuit has a convolution computer, and further has an adaptive digital filter and a filter control unit to which the signals from the array are supplied before being operating on by the convolution computer. The filter control unit controls the transfer function of the filter as a function of the filter input signal so as to substantially reduce image artifacts due to quantum and electronics noise.

Flexible filter disposed within flexible conductor

Abstract: There is shown and described a tubular flexible filter membrane disposed within a flexible conductor such as IV tubing. This flexible filter membrane is sufficiently smaller in its outer diameter than the inner diameter of the flexible conductor so that a space is provided for the flow of fluid. This flexible filter is hydrophilic and has one end closed to exclude fluid flow to the interior of the filter. The other end of this filter membrane is secured to a tubular conductor and at this secured portion is made non-conductive as to gas and fluid flow through the filter wall. A method of attaching this filter membrane end by heating the thermoplastic filter membrane so as to cause thinning of the wall of the filter membrane and making this thinned portion non-conductive as to gas flow is shown. Potting, using polyurethane compounds, may be used but is time-consuming and costly. The flexible conductive tubing is usually produced by extruding dies and methods so ribs and/or flutes may be formed and are contemplated although not illustrated. Heating of the end of the filter membrane is shown by two modes of apparatus. When this filter design is used in conjunction with a drainage system, a vent as well as anti-reflux devices are not required in the collection bag.

The use of wave filter design in Kalman filter state estimation of the automatic steering problem of a tanker in a seaway

Abstract: The problem of automatic steering control of a large tanker in a seaway is formulated within the framework of linear quadratic Gaussian (LQG) control theory. Wave disturbances are characterized by shaping filters, and Kalman filters are designed using these disturbance noise models. LQG controllers are designed to minimize a performance criterion commonly thought to be representative of propulsion losses due to steering. Performance of the controllers is determined by simulation results, which apply for deep water and are based on data from scale model tests.

Detecting signals within a passband on a telephone line

Abstract: An improved combination of a hardware and digital signal processing filter for detecting pick-up of a telephone call, solely through audio information on the telephone line. The apparatus employs a high gain band pass filter (28) with no automatic gain control, the output of which goes to a window comparator (30). The output from the window comparator (31) goes to a digital high pass filter (32) and from there to an integrator 37 for providing a digital output signal (40) indicative of the presence or absence on the telephone line of a signal exceeding a predetermined magnitude within the filter pass band. The digital signal is then processed by an intelligent digital filter having a set of predetermined threshold values of durations for states of the digital output signal, by which determinations of pick-up are made. The digital filter is adaptive and learns the durations high and low states of the digital output signal as they occur, subsequently checking for deviations from previously learned valid values. The digital filter includes a digital phase lock loop which will lock into a periodic but asymmetric pattern in the digital output signal and declare pick-up when lock is lost.

Wide frequency band coupling network for an active ripple and noise filter

Abstract: A ripple and noise filter for high voltage DC converters or power supplies The filter includes a common terminal, an input terminal and an output terminal A voltage sensing circuit is connected between the output and common terminals of the filter and is adapted to produce a sense signal which is proportional to the output voltage at the output terminal An active filtering circuit, preferably an amplifier having a frequency operating range which covers the frequency spectrum of the ripple and noise voltages, includes an input for receiving the sense signal and an output at which a correction signal is generated The correction signal has a magnitude and waveform such that coupling the correction signal onto the output terminal of the filter eliminates or reduces the ripple and noise A passive coupling element, preferably a capacitor, isolates the amplifier from the output terminal and couples the correction signal thereto A passive filter, also a capacitor, is connected across the output and common terminals of the filter The coupling capacitor and the passive filter capacitor have nearly identical electrical characteristics so that the correction signal is coupled onto the output terminal without distortion over the full frequency operating range of the amplifier

Method and apparatus for converting an input scanning sequence into an output scanning sequence

Abstract: Method and apparatus for converting an input scanning sequence into an output scanning sequence, where scan values of an input scanning sequence occurring with an input scanning frequency are converted into an output scanning sequence whose scan values occur with an output scanning frequency. To process scanning sequences with alternating scanning frequencies with limited computing and storage capacities, a single one-stage digital filter is provided for processing the input scanning sequence using a limited number of filter coefficients, further filter coefficients supplied thereto by calculation from existing filter coefficients.

Apparatus and method for adjusting the receivers of data transmission channels

Abstract: Filters are provided at the outputs of data transmission links to overcome attenuation distortion and phase distortion but such filters tend to increase the noise content of received signals. In the present invention a filter coupled to a linear baseband channel is constrained to be an all pass network, thus avoiding any noise enhancement at its output. The filter is adjusted using a channel estimator and an adjustment system to give the channel plus the filter an impulse-response that rises rapidly to its peak, thus simplifying the detection process needed for a satisfactory tolerance to noise. The required response is obtained by finding those roots (zeros) of the z transform of the sampled impulse-response of the channel, which have a modulus greater than some given value which is not less than unity, and adjusting the filter such that, in the z transform of the sampled impulse-response of the channel and filter, the roots are replaced by the complex conjugates of their reciprocals, the remaining roots of the channel z-transform being left unchanged.

A digital linear filter for resistivity sounding with a generalized electrode array

Abstract: This paper presents a digital linear filter which maps composite resistivity transforms to apparent resistivities for any four—electrode array over a horizontally layered earth. A filter is provided for each of three sampling rates; the choice of filter will depend on resistivity contrasts and computational facilities. Two methods of filter design are compared. The Wiener-Hopf least-squares method is preferable for low sampling rate filters. The Fourier transform method is more successful in producing a filter with a high sampling rate which can handle resistivity contrasts of 100 000: 1.