Showing papers on "Adaptive filter published in 1978"

Adaptive filtering in the frequency domain

Abstract: Adaptive filtering in the frequency domain can be accomplished by Fourier transformation of the input signal and independent weighting of the contents of each frequency bin. The frequency-domain filter performs similarly to a conventional adaptive transversal filter but promises a significant reduction in computation when the number of weights equals or exceeds 16.

A class of all-zero lattice digital filters: Properties and applications

Abstract: A class of minimum- or maximum-phase all-zero lattice digital filters, based on the two-multiplier lattice of Itakura and Saito, is developed. Different lattice forms with different numbers of multipliers are derived, including two one-multiplier forms. Many of the properties of these lattice filters are given, including the important orthogonalization and decoupling properties of successive stages in optimal inverse filtering of signals. These properties lead to important applications in the areas of adaptive linear prediction and adaptive Wiener filtering. As a specific example, the design of a new fast start-up equalizer is presented.

An adaptive lattice structure for noise-cancelling applications

Abstract: This paper describes an adaptive filter structure which may be used in multi-channel noise-cancelling applications. The proposed structure differs from those presented previously in that it incorporates a lattice filter framework, rather than tapped-delay-lines. The successive orthogonalization provided by the lattice offers advantages in adaptive convergence rate which cannot be achieved with tapped-delay-lines. In the sections below, we present an explicit description of the general noise-cancelling lattice structure, together with the appropriate adaptive algorithms.

Evaluation of an adaptive comb filtering method for enhancing speech degraded by white noise addition

Abstract: An intelligibility test was performed to evaluate an adaptive comb filtering method proposed by Frazier [2] for enhancement of degraded speech due to additive white noise. Results indicate that independent of S/N ratio the adaptive comb filtering scheme does not increase speech intelligibility.

Use of the Butterworth low-pass filter for oceanographic data

Abstract: The characteristics of the Butterworth low-pass filter are well known in electrical engineering. Here we discuss its use for oceanographic records and compare its characteristics with other low-pass filters now in use: the cosine-Lanczos filter, the Gaussian filter, and the ideal filter. The Butterworth filter is recursive, i.e., past values of the output are used as input, so a phase shift is introduced unless the data are filtered forward and backward through the same filter. When this is done, the filtered signal differs only slightly from that of other low-pass filters. Because the Butterworth filter uses fewer multiplicative constants for the same effect, there is a reduction in computer time over other low-pass filters; the difference becomes more pronounced as more data points are used.

Stability of general two-dimensional recursive digital filters

Abstract: Two-dimensional recursive filters are defined from a different point of view. A general stability preserving mapping theorem is presented which allows most recursive filters of a particular type to be mapped into any other type of recursive filter. In particular, any type of filter can be mapped into a first-quadrant filter. This mapping is used to prove a number of general stability theorems. Among these is a theorem which relates the stability of any digital filter to its two-dimensional phase function. Furthermore, other stability theorems which are valid for any type of recursive filter are presented. Finally, a number of practical stability tests are developed including one which requires the testing of only several one-dimensional polynomial root distributions with respect to the unit circle.

Block implementation of recursive digital filters--New structures and properties

Abstract: Several new structures for the block implementation of HilR digital filters are proposed. The relation between the pole locations of the block structure to that of the original scalar transfer function is derived. A method to obtain the scalar transfer function from a given block structure is described.

Design of two-dimensional recursive digital filters with specified magnitude and group delay characteristics

Abstract: A technique is proposed for the design of two-dimensional (2-D) recursive filters with a response that best approximates, in the l_p sense, prescribed magnitude and group delay specifications. The filter stability is guaranteed through the use of a frequency transformation. The optimization technique used is that of Davidon-Fletcher and Powell. Several examples are given to illustrate the proposed algorithm.

Adaptive line enhancer

Abstract: An input signal X(j) is fed directly to the positive port of a summing function and is simultaneously fed through a parallel channel in which it is delayed, and passed through an adaptive linear transversal filter, the output being then subtracted from the instantaneous input signal X(j). The difference, X(j)-Y(j), between these two signals is the error signal e(j). e(j) is multiplied by a gain μ and fed back to the adaptive filter to readjust the weights of the filter. The weights of the filter are readjusted until e(j) is minimized according to the recursive algorithm: ##EQU1## where the arrow above a term indicates that the term is a signal vector. Thus, when the means square error is minimized, W.sub.(j+1) =W.sub.(j), and the filter is stabilized.

Channel estimating reference signal processor for communication system adaptive antennas

Abstract: A spread spectrum communication adaptive array antenna processor is disclosed which can acquire and remain synchronized to a pseudo-noise (PN) signal transmitted in a multipath signal environment. The plurality of antennas which receive rf signals are individually associated with mixing circuitry which reduces the received signals to IF frequencies. The IF signals are fed into the adaptive filtering portion of the adaptive signal processor which contains circuits to generate an adaptive weight corresponding to each antenna element. An array signal is formed by summing the products of each IF signal with a filter weight corresponding to each antenna element generated within each respective adaptive loop. The adaptive signal processor utilizes the complex conjugate of the error feedback signal which is then multiplied by each respective IF signal. The complex conjugate of this integrated product forms each filter weight. A channel estimator generates an adaptive reference signal which inclues the essential multipath characteristics of the received signal. By using this reference signal in conjunction with the array signal generated by the adaptive filtering portion of the processor, the adaptive array can form an appropriate main beam without prior knowledge of the signal propagation direction.

A design technique for circularly symmetric low-pass filters

Abstract: The design of two-dimensional (2-D) circularly-symmetric low-pass digital filters by cascading several rotated filters (a rotated filter is defined to be one produced by rotating a one-dimensional (1-D) continuous filter into a two-dimensional continuous filter which is in turn bilinearly transformed into a two-dimensional digital filter) is a well-known and useful technique. An alternate approach which is an extension of the above technique is presented. This new method is based on a spectral transformation from the one-dimensional discrete domain into the two-dimensional discrete domain. This approach retains most of the advantages of the original technique while permitting design entirely in the discrete domain, yielding filters with better stability characteristics, and facilitating frequency response optimization via nonlinear programming.

Adaptive Tracking Filter for Maneuvering Targets

Abstract: A general method of continually restructuring an optimum Bayes-Kalman tracking filter is proposed by conceptualizing a growing tree of filters to maintain optimality on a target exhibiting maneuver variables. This tree concept is then constrained from growth by quantizing the continuously sensed maneuver variables and restricting these to a small value from which an average maneuver is calculated. Kalman filters are calculated and carried in parallel for each quantized variable. This constrained tree of several parallel Kalman filters demands only modest om; puter time, yet provides very good performance. This concept is implemented for a Doppler tracking system and the performance is compared to an extended Kalman filter. Simulation results are presented which show dramatic tracking improvement when using the adaptive tracking filter.

Adaptive lattice methods for linear prediction

Abstract: A general method for adaptive updating of lattice coefficients in the linear predictive analysis of nonstationary signals is presented. The method is given as one of two sequential estimation methods, the other being a block sequential estimation method. The fast convergence of adaptive lattice algorithms is seen to be due to the orthogonalization and decoupling properties of the lattice. These properties are useful in adaptive Wiener filtering. As an application, a new fast start-up equalizer structure is presented. In addition, a one-multiplier form of the lattice is presented, which results in a reduction of computations.

Design of two-dimensional recursive digital filters

Abstract: The present paper develops a design technique for approximating nonseparable frequency characteristics by sums and products of separable transfer functions. This approximation is called the "piecewise separable" decomposition of the characteristic. In the design technique, the desired filter with half-plane symmetry (radial symmetry) is obtained by shifting a low-pass characteristic in the frequency domain, and by combining these shifted characteristics. Also the paper includes design approaches for the four-quadrant symmetry filters. Two examples illustrate the technique of the paper.

Computer-Aided Filter Alignment and Diagnosis

Abstract: The cavity resonant frequencies and coupling values of a wide range of bandpass filters, band-reject filters, and equalizers have been determined in situ by computer-adjusting analytic models to fit the scattering parameters measured on an automatic network analyzer. A higher order mode elliptic filter, a dual-mode quasi-elliptic filter, and a dual-mode band-reject filter are presented as examples. The general relationships between mechanical dimensions and circuit parameters are discussed. The circuit adjustment procedure is outlined, and equations for the sensitivity coefficients of several element types are tabulated.

Convergence of an Adaptive Filter Algorithm

Abstract: A state-space representation of a dynamical, stochastic system is given. A corresponding model, parametrized in a particular way, is considered and an algorithm for the estimation of its parameters is analysed. The class of estimation algorithms thus considered contains general output error methods and model reference methods applied to stochastic systems. It also contains adaptive filtering schemes and, e.g. the extended least squares method. It is shown that if a certain transfer function associated with the true system is positive real, then the estimation algorithm converges with probability 1 to a value that gives a correct input-output model.

On an adaptive algorithm for IIR filters

Abstract: The purpose of this communication is to discuss an IIR adaptive filter algorithm developed by Stearns [1], in terms of an example that appeared in a recent article [2]. The example concerns the approximation of a fixed second-order filter by a first-order adaptive filter, when subjected to a white noise input.

Detection of Narrowband Signals Using Time-Domain Adaptive Filters

Abstract: The detection performance of a conventional narrowband analyzer is compared with two adaptive processor mechanizations based on the Widrow least mean squares algorithm. Comparisons are based on both analysis and extensive digital simulation. With a narrowband signal in stationary, white background noise, the performance of the three systems is shown to be essentially the same. With nonstationary background noise, the performance of the conventional system degrades by an amount proportional to the processing time-bandwidth product. The adaptive systems appear to be less sensitive to the nonstationary background, resulting in a potential performance advantage relative to the conventional system.

An Improved Algorithm for Adaptive Processing

Abstract: The Widrow-Hoff least mean square (LMS) algorithm based on the method of steepest descent is conditionally stable. A modified algorithm is given which is unconditionally stable, capable of better performance when used in adaptive filter processing, and provides a more realistic means for simulating the Applebaum-Howells adaptive loop.

Frequency domain adaptive filter for detection of sonar signals

Abstract: An adaptive filter for sonar signals which operates on the complex spectral components of the received signal, which signal has been transformed into the frequency domain by means such as the Fast Fourier Transform The adaptive filter consists of a plurality of component filters, each of which operates on a single spectral component of the received signals The transfer coefficient of each component filter is described by a complex number and is adaptively adjusted by means of a computational feedback loop The feedback loop compares the product of the transfer coefficient and the complex spectral component of the received signal from a prior frequency transformation cycle, with the present spectral component to obtain an error signal The error signal, in turn, adaptively alters the magnitude and phase of the transfer coefficient A plurality of such component filters operate together to adaptively filter, in the frequency domain, the entire spectrum of the received signal

Linear transformation active filters

Abstract: An approach is presented in this paper for the design of active filters from doubly terminated lossless ladder filters. The method relies-on-the-lnear transformation of port variables from the V-I domain to a new domain in which the filter is realized actively. The resulting active filters are referred to as linear transformation active (LTA) filters. The transformations are applied on an element-by-element basis thus producing a systematic and effective way of achieving new active structures while others like the leapfrog and the wave active filter structures appear as special cases. Examples are presented in the paper to show the efficiency of the method and the importance of the results.

On digital filter structures with low coefficient sensitivities

Abstract: A simple network transformation method is proposed to generate equivalent digital filter structures whose multiplier coefficients are functions of the parameters of the transformation. An application of this proposed approach, illustrated here, is in generating structures with low coefficient sensitivities.

Variable-cutoff two-dimensional lowpass f.i.r. digital filters

Abstract: A recently formulated transformation for implementing 1-dimensional variable-cutoff filters is extended to the design of 2-dimensional variable-cutoff lowpass filters, and it is shown that the cutoff frequencies along the two frequency axes can be independently controlled, each by one parameter. The filters designed with the 1st-order transformation have certain drawbacks, which are shown to be overcome with the 2nd-order transformation with proper constraints on the variable parameters.

Weighting coefficients for certain maximally flat nonrecursive digital filters

Abstract: The weighting coefficients for maximally flat nonrecursive digital ruters have been derived for low-pass filters having equal passband and stopband widths. The solution is in closed form and can be readily evaluated even for large filter order. Asymptotically the coefficients of the finite series expansion converge to the Fourier series coefficients. Hilbert transformers and decimation filters are among the suggested applications for this form of filter.

Separation of NTSC Signals by Minimum Mean Square Error Filters and Adaptive Filters

Abstract: Two techniques for separating a composite NTSC signal into luminance and chrominance components are developed and compared. In the first technique, fixed filters, which are based upon minimum mean square error estimators, are designed. It is shown that, when samples from successive lines are used, these filters have a comb structure in the region of the frequency spectrum in which the luminance and chrominance signals overlap. In the second technique, the composite signal is filtered by several filters, and the outputs of the filters are selected dependent upon the local characteristics of the picture. Both recursive and nonrecursive filters are considered, the criterion for selecting between filters is investigated and the effect of delayed decisions is determined. The various filters are compared using subjective criterion and a number of quantitative criterion. The relationships between the quantitative and qualitative criterion are stated.

The design of optimal multirate bandpass and bandstop filters

Abstract: Narrow-band bandpass and bandstop filters are inherently of high order and require a large computation rate. A multirate filter using decimators and interpolators can be designed to have bandpass or bandstop characteristics, often with a much smaller computation rate. This paper develops rules for designing such a filter by placing constraints on the filter approximation error and the aliasing error. The question of admissible decimation factors is investigated in detail. A method to minimize the computation rate is described. Several examples are presented.

On the design of wave digital lattice filters with short coefficient word lengths and optimal dynamic range

Abstract: Wave digital lattice filters require, in general, fewer multipliers than other digital realizations and have excellent passband sensitivity properties. Because of the variety of solutions which are possible for the realization of the lattice reactances, these filters can be designed with simple multiplier coefficients in spite of the high sensitivity in the stopband. The filter structure which offers the optimal dynamic range is determined by means of the equivalences between parallel and series adaptors and by means of scaling with coefficients equal to powers of two. This is illustrated by some specific examples.

Dynamic filter for a moving head disk storage system

Abstract: A dynamic filter circuit used to process analog position error information in a disk drive head-positioning servo system is disclosed. The filter circuit monitors the velocity of the disk drive carriage and dynamically increases the gain and bandwidth of the filter as the carriage velocity increases. Specifically, an operational amplifier in feedback configuration is used in which the feedback networks are switched in response to the velocity signal. At low carriage velocity the filter has low gain and reduced bandwidth and therefore helps to decrease sensitivity of the system to noise. At high carriage velocities the gain and bandwidth of the filter are increased to minimize the effects of missing servo data.

Adaptive filter network

Abstract: An adaptive filter network comprising a controllable filter having an adjustable cut-off frequency and adapted for varying the pass-band of the network. The adaptive filter network further includes a serial arrangement of an algebraic adder for generating control signals, connected to the controllable filter; a weighting filter for converting the control signal spectrum in response to the load sensitivity variation with frequency; a threshold limiter for setting the noise reduction threshold level of the adaptive filter network; a control signal frequency corrector; and an amplitude detector for shaping control signals applied to the control input of the controllable filter. The network analysis of the input audio signal spectrum and the width of its pass-band is varied depending on the present frequency limit of the input audio signal wanted components.

Digital filter structures having low errors and simple hardware implementation

Abstract: Sensitivity and roundoff errors can seriously limit the application of recursive digital filters in practice, particularly when the filters have poles near z = + 1 . A filter structure, based on digital incremental computers is proposed, which has low sensitivity, good error characteristics, and simple hardware implementation for pole locations close to z = + 1 . Expressions for the roundoff errors are derived and compared to those for conventional structures. A design procedure is suggested to implement the new filter structure given the transfer function. Simulation results are presented.