Showing papers on "Filter design published in 1991"

Design and implementation of an extended Kalman filter for the state estimation of a permanent magnet synchronous motor

Abstract: Practical considerations for implementing the discrete extended Kalman filter in real time with a digital signal processor are discussed. The system considered is a permanent magnet synchronous motor (PMSM) without a position sensor, and the extended Kalman filter is designed for the online estimation of the speed and rotor position by only using measurements of the motor voltages and currents. The algorithms developed to allow efficient computation of the filter are presented. The computational techniques used to simplify the filter equations and their implementation in fixed-point arithmetic are discussed. Simulation and experimental results are presented to demonstrate the feasibility of this estimation process. >

Analysis of the properties of median and weighted median filters using threshold logic and stack filter representation

Abstract: The deterministic properties of weighted median (WM) filters are analyzed. Threshold decomposition and the stacking property together establish a unique relationship between integer and binary domain filtering. The authors present a method to find the weighted median filter which is equivalent to a stack filter defined by a positive Boolean function. Because the cascade of WM filters can always be expressed as a single stack filter this allows expression of the cascade of WM filters as a single WM filter. A direct application is the computation of the output distribution of a cascade of WM filters. The same method is used to find a nonrecursive expansion of a recursive WM filter. As applications of theoretical results, several interesting deterministic and statistical properties of WM filters are derived. >

A class of fast Gaussian binomial filters for speech and image processing

Abstract: The authors present an efficient, in-place algorithm for the batch processing of linear data arrays. These algorithms are efficient, easily scaled, and have no multiply operations. They are suitable as front-end filters for a bank of quadrature mirror filters and for pyramid coding of images. In the latter application, the binomial filter was used as the low-pass filter in pyramid coding of images and compared with the Gaussian filter devised by P.J. Burt (Comput. Graph. Image Processing, vol.16, p.20-51, 1981). The binomial filter yielded a slightly larger signal-to-noise ratio in every case tested. More significantly, for an (L+1)*(L+1) image array processed in (N+1)*(N+1) subblocks, the fast Burt algorithm requires a total of 2(L+1)/sup 2/N adds and 2(L+1)/sup 2/ (N/2+1) multiplies. The binomial algorithm requires 2L/sup 2/N adds and zero multiplies. >

The analysis and design of multidimensional FIR perfect reconstruction filter banks for arbitrary sampling lattices

Abstract: A general analysis of multidimensional multirate filter banks is presented. The approach is applicable to discrete signal spaces of any dimension, to multirate systems based on arbitrary downsampling and upsampling lattices, and for filter banks with any number of channels. A new numerical design procedure is also presented for multidimensional multirate perfect reconstruction filter banks, which is based on methods of nonlinearly constrained numerical optimization. An error function that depends only on the analysis filter impulse response coefficients is minimized, subject to a set of quadratic equality constraints that involve both the analysis and synthesis filter coefficients. With this design framework, it is possible to design a wide variety of filter banks that have a number of desirable properties. The analysis and synthesis filters that result are finite impulse response (FIR) and of equal size. In addition, both paraunitary and nonparaunitary filter banks can be designed with this method. Unlike paraunitary filter banks, nonparaunitary filter banks are capable of performing analysis bank functions more general than band-splitting with flat passband filters. >

On optimal infinite impulse response edge detection filters

Abstract: The authors outline the design of an optimal, computationally efficient, infinite impulse response edge detection filter. The optimal filter is computed based on Canny's high signal to noise ratio, good localization criteria, and a criterion on the spurious response of the filter to noise. An expression for the width of the filter, which is appropriate for infinite-length filters, is incorporated directly in the expression for spurious responses. The three criteria are maximized using the variational method and nonlinear constrained optimization. The optimal filter parameters are tabulated for various values of the filter performance criteria. A complete methodology for implementing the optimal filter using approximating recursive digital filtering is presented. The approximating recursive digital filter is separable into two linear filters, operating in two orthogonal directions. The implementation is very simple and computationally efficient. has a constant time of execution for different sizes of the operator, and is readily amenable to real-time hardware implementation. >

Iterative Wiener filters for image restoration

Abstract: The iterative Wiener filter, which successively uses the Wiener-filtered signal as an improved prototype to update the covariance estimates, is investigated. The convergence properties of this iterative filter are analyzed. It has been shown that this iterative process converges to a signal which does not correspond to the minimum mean-squared-error solution. Based on the analysis, an alternate iterative filter is proposed to correct for the convergence error. The theoretical performance of the filter has been shown to give minimum mean-squared error. In practical implementation when there is unavoidable error in the covariance computation, the filter may still result in undesirable restoration. Its performance has been investigated and a number of experiments in a practical setting were conducted to demonstrate its effectiveness. >

A new class of nonlinear filters for image enhancement

Abstract: Two types of very simple two-dimensional nonlinear filters are introduced and applied to image contrast enhancement. The first type is based on a generalization of the Teager's algorithm. A theoretical analysis has shown that this type of nonlinear filter works like a local-mean-weighted highpass filter. Based on this analysis, a second type of nonlinear filter has been developed which works like local-mean-weighted bandpass filter. The proposed image contrast enhancement technique is based on combining the original image with its filtered version obtained using one of the two nonlinear filters. Very high quality enhancement has been achieved for natural images. >

FIRGEN: a computer-aided design system for high performance FIR filter integrated circuits

Abstract: The authors describe automatic architecture and floorplan generation techniques for integrated circuit fixed-coefficient FIR (finite impulse response) filters that can achieve high sample rates with compact layouts. These techniques have been implemented in a filter design system called FIRGEN that can automate the entire design from filter specifications to final chip layout. It can be retargeted to new cell libraries and place and route tools. Result on four chips designed with FIRGEN are presented. These achieve sample rates ranging from 25 MHz to 112 MHz. >

A modified parallel-coupled filter structure that improves the upper stopband rejection and response symmetry

Abstract: A modified parallel-coupled microstrip line filter structure is presented. Using this new structure improves the filter upper stopband rejection by at least 15 dB, and the filter response symmetry is also improved. Compared with the traditional parallel-coupled filter, the modified filter used less space and is easy to lay out owing to its inline structure. Several examples show the performance improvement of the filters fabricated in both low-dielectric-constant (2.55) and high-dielectric-constant (10.2) substrates. >

Feedback cancellation in hearing aids: results from a computer simulation

Abstract: Feedback cancellation in hearing aids involves estimating the feedback signal and subtracting it from the microphone input signal. The feedback-cancellation system described updates the estimated feedback path whenever changes are detected in the feedback behavior. When a change is detected, the normal hearing-aid processing is interrupted, a pseudorandom probe signal is injected into the system, and a set of filter coefficients is adjusted to give an estimate of the feedback path. The hearing aid is then returned to normal operation with the feedback-cancellation filter as part of the system. Two approaches are investigated for computing the filter coefficients: a least-mean square (LMS) adaptive filter and a Wiener filter. Test results are presented for a computer simulation of an in-the-ear (ITE) hearing aid. The simulation results indicate that more than 10 dB of cancellation can be obtained and that the Wiener filter is more effective in the presence of strong interference. >

Genetic algorithm in the design of FIR filters

Abstract: The author's aim is to describe a method of designing finite impulse response (FIR) filters that is automatic, rapid, and gives filter realisations of near minimal computational complexity. Existing methods of filter design are reviewed to show that none possesses all these features. These methods include recent work using a sequential algorithm that produces realisations of guaranteed minimum complexity and thus provides a reference for the results described in the paper. Genetic algorithms are described, and a method of representing the problem of filter synthesis for solution by a genetic algorithm is given. Results are presented, demonstrating the suitability of the genetic algorithm design method. >

Cosine-modulated analysis-synthesis filterbank with critical sampling and perfect reconstruction

Abstract: The authors present a simple derivation of a parallel filterbank based on cosine-modulated versions of a model low-pass filter. With a nonuniform channel separation an efficient implementation consisting of a DFT (discrete Fourier transform) related transform and subfilters is possible. Using critical sampling of each channel and FIR (finite impulse response) filters, the conditions for perfect reconstruction are given. The computational complexity of the derived FIR filterbank is much lower than for a tree-structured FIR filterbank but cannot compete with the most efficient IIR filterbanks. >

Active type noise controller

Abstract: PURPOSE:To suppress the increment of arithmetic load on an arithmetic unit and to attain noise reduction precisely by predicting the divergence of a control sound source based on the update quantity of the filter coefficient of an adaptive filter. CONSTITUTION:The filter coefficient is computed so as to minimize the square e of sound pressure based on a reference signal r1m to which filter processing is applied corresponding to the number of combination of transfer relation between microphones 8a-8h and speakers 7a-7d by a microprocessor 16, and the sum computation of the square e of the sound pressure, and the filter processing is applied to a reference signal from a frequency-voltage conversion circuit 11 by the filter coefficient as updating that of the adaptive filter 13 adaptively, and the speakers 7a-7d can be driven. Therefore, since it is possible to predict the divergence of the control sound source and to operate a divergence regulation means 22, divergence phenomenon can be suppressed even when the mechanical characteristics of the speakers 7a-7d and the microphones 8a-8h are changed due to the lapse of time, or the temperature change of control space occurs, etc. Also, it is possible to suppress the increment of arithmetic load on the arithmetic unit.

Variable decimation architecture for a delta-sigma analog-to-digital converter

Abstract: An analog-to-digital converter includes a delta-sigma modulator (10), having the output thereof filtered by a digital filter section. The digital filter section includes a first fixed decimation filter (12) followed by a variable decimation filter section (14) and an output low-pass filter section (16), having a fixed decimation ratio. The fixed variable decimation filter section (14) includes a single FIR filter (24) that has data processed therethrough with different sampling rates. A recursive controller (26) receives an external configuration input to determine the number of passes through the filter (24) that are required to provide the desired decimation ratio.

Performance evaluation of subband coding and optimization of its filter coefficients

Abstract: In this paper, two analytical methods to evaluate coding performance of subband coding are proposed,and optimization of its filter coefficients from the viewpoint of energy compaction property is considered.The first method is based on matrix representation of subband coding in time domain, where the codinggain given by Jayant and Noll is introduced as a performance measure for filter banks with orthogonalproperty. The second method is based on optimum bit allocation problem for subband coding (multiratefilter bank), where the unified coding gain is derived as a new performance measure which can be appliedto arbitrary transform techniques. We then try to find filter coefficients which maximize the unified codinggain according to input characteristics. This approach leads to optimization of filter coefficients from theviewpoint of energy compaction property.

Subband echo canceller with adjustable coefficients using a series of step sizes

Abstract: A received input signal and an echo signal resulting from the passage of the received input signal through an echo path are both analyzed or divided into a plurality of common subbands. The received input signal in each subband is supplied to an estimated echo path provided in the subband, by which it is rendered into an echo replica signal. The echo replica signal is subtracted, by a subtractor provided in each subband, from the echo signal in the same subband as the echo replica signal to obtain a residual echo signal. The residual echo signals in the respective subbands are synthesized into a full-band residual echo signal. The estimated echo path in each subband is formed by a digital FIR filter and its filter coefficients are calculated by a coefficient calculation part in the subband, based on the received input signal, the residual echo signal and a step size matrix. The filter coefficients are iteratively updated so that the residual echo signal in each subband may be minimized. The step size matrix is used to define the step size of the filter coefficients and is determined by an acoustic field characteristics calculation part, based on the variation characteristics of an impulse response of the echo path in each subband.

On-signal decomposition techniques

Abstract: Well-known block transforms and perfect reconstruction orthonormal filter banks are evaluated based on their frequency behavior and energy compaction. The filter banks outperform the block transforms for the signal sources considered. Although the latter are simpler to implement and already the choice of the existing video coding standards, filter banks with simple algorithms may well become the signal decomposition technique for the next generation video codecs, which require a multiresolution signal representation.

Digital virtual earth active cancellation system

Abstract: A digital virtual earth active cancellation system (14) which receives a phenomena input signal (r) representing residual phenomena to be cancelled and has an adaptive filter (44) which generates a cancellation signal (y). A system impulse response (c) is convolved with the cancellation signal (y) and is subtracted from the input signal (r) to produce an estimate of noise (x). The adaptive filter (44) produces the cancellation signal (y) by filtering the estimated noise (x) with a filter weight (a) that is adapted by the system impulse response (c). By convolving the estimate of noise (x) with the system impulse (c) to adapt the filter (44), the values sent to an adapter (50) for the adaptive filter (44) are kept within 90° phase of the residual signal (r). This substantially eliminates the problems associated with destructive feedback due to phase shifts without the need for external reference signals.

Statistically optimized PR-QMF design

Abstract: A multivariable optimization problem is set to design 2-band PR-QMFs in this paper. The energy compaction, aliasing energy, step response, zero-mean high-pass filter, uncorrelated subband signals, constrained nonlinearity of the phase-response, and the given input statistics are simultaneously considered in the proposed optimal filter design technique. A set of optimal PR-QMF solutions and their optimization criteria along with their energy compaction performance are given for comparison. This approach of PR-QMF design leads to an input driven adaptive subband filter bank structure. It is expected that these optimal filters outperform the well-known fixed PR-QMFs in the literature for image and video coding applications.

Variable length stochastic gradient algorithm

Abstract: The transversal variable-length stochastic gradient algorithm is described. It is a modification of the stochastic gradient algorithm that allows dynamic allocation of coefficients of an adaptive filter. The order of the filter and the adaptation step size are changed automatically when an appropriate level of performance is reached during the course of the adaptation process. In this way, the algorithm results in fast convergence, typical of low-order filters, and good steady-state performance, typical of high-order filters. >

Closed-loop feedback control system having an adaptive filter

Abstract: A closed-loop feedback control system is disclosed for controlling an operation of a load (4). An actual operation signal is subtracted from a reference operation signal to produce a resultant signal which is used to control the operation of the load (4). A filter unit, such as a notch filter, operates to suppress any machine resonance frequencies found in the operation signal due to load fluctuations, machine variations, operating environment changes, deterioration with age, etc. A filter coefficient of the filter unit is adjusted so that any fluctuation in the resonance frequency can be suppressed effectively.

Image filtering using multiresolution representations

Abstract: It is shown how multiresolution representations can be used for filter design and implementation. These representations provide a coarse frequency decomposition of the image, which forms the basis for two filtering techniques. The first method, based on image pyramids, is used for approximating the convolution of an image with a given mask. In this technique, a filter is designed using a least-squares procedure based on filters synthesized from the basic pyramid equivalent filters. The second method is an adaptive noise reduction algorithm. An optimally filtered image is synthesized from the multiresolution levels, which in this case are maintained at the original sampling density. Individual pixels of the image representation are linearly combined under a minimum mean square error criterion. This uses a local signal-to-noise ratio estimate to provide the best compromise between noise removal and resolution loss. >

Echo canceller with improved doubletalk detection

Abstract: An echo canceller for use in send/receive apparatus includes a frequency-domain adaptive filter which provides filter parameters to a time-domain programmable filter. The programmable filter produces a signal y(k) which is an estimate of the echo signal produced by acoustic coupling between a loudspeaker in the received path and a microphone in the send path. By subtracting this estimate from the signal produced in the send path, an output signal r(k) is obtained from the send path which is essentially free of echoes. An accurate estimate of the echo signal cannot be obtained if a send signal is present at the input of the send path. Therefore, detecting means in the adaptive filter detects whether such a send signal is present and in that case blocks the transfer of filter parameters from the adaptive filter to the programmable filter. Erroneous modification of the estimated echo signal is thereby avoided. Detection of the presence of a send signal is based on whether the ratio of the power of the signal z(k) in the send path to the power of an output signal of the adaptive filter exceeds a specific threshold.

Low precision finite impulse response filter for digital interpolation

Abstract: A low precision Finite Impulse Response filter (FIR) is provided for filtering in a digital interpolation operation. The interpolation operation includes two steps, a sampling rate conversion operation for interspersing zeroes between samples in an input sequence and a filtering step of filtering out images that result from this operation. The filtering operation utilizes a FIR filter that utilizes low precision filter coefficients that are selected to tune the frequency response such that the low end frequency response including the pass band, the transition band, and the portion of the stop band immediately after the transition band provides a response equivalent to that commensurate with substantially higher precision FIR filter coefficients, with the high frequency end of the stop band gradually increasing. A second, low pass filter section is provided for filtering out the image energy that exists at the output of the FIR filter in the high frequency end of the stop band to provide an overall filter response that is commensurate to that utilizing substantially higher precision FIR coefficients in the filter section. The FIR filter coefficients utilized are restricted to the set of [-1, 0, +1] such that a multiplierless FIR filter can be realized. The FIR filter coefficients are obtained by processing the infinite FIR filter coefficients through a software delta-sigma quantizer which quantizes the output to the desired low precision FIR filter coefficients.

Adaptive filters based on order statistics

Abstract: An adaptive filter structure which is based on linear combinations of order statistics is proposed. An efficient method to update the filter coefficients is presented, which is based on the minimal mean-square error criterion and which is similar to the Widrow algorithm for the linear adaptive filters. Another method for coefficient update is presented, which is similar to the recursive least squares (RLS) algorithm and which has faster convergence properties. The proposed-filter can adapt well to a variety of noise probability distributions, including impulsive noise. It also performs well in the case of nonstationary signals and, therefore, it is suitable for image-processing applications. >

Doppler optimised mismatched filters

Abstract: The problem of designing radar detection filters that minimise sidelobe levels in both Doppler and range co-ordinates is considered. The new algorithm for a Doppler optimised mismatch filter is constructed. The generalisation enables sidelobe suppression for the polyphase and complex multilevel sequences. For minimax filter coefficients calculation, the iterative reweighted least-square procedure is successfully applied. Favourable experimental results are obtained, especially for polyphase codes.

The design of perfect reconstruction nonuniform band filter banks

Abstract: The authors describe a general procedure for the design of analysis-synthesis systems based on nonuniform filter banks. The procedure is based on a time-domain analysis which results in a set of time-domain conditions for the exact reconstruction of the input signal at the output. These conditions are used as part of a powerful tool for designing FIR (finite impulse response) filter banks with an arbitrary nonuniform frequency resolution. This framework allows for the design of systems with arbitrary rational decimation rates in different bands. Systems based on maximally or non-maximally decimated filter banks, low and minimum delay systems, and systems based on block decimators are also among the systems that can be designed using this method. >

Programmable canonic signed digit filter chip

Abstract: A programmable canonic signed digit (CSD) filter is provided which employs programmable CSD multipliers. The filter receives a digital input signal and includes a tapped delay line for providing a delay between the input samples. The filter advantageously employs a plurality of programmable CSD multipliers which receive programmable input filter coefficients and perform multiplication of the coefficient with the delayed input signal. Summation hardware is further included for summing the outputs of the plurality of multipliers to provide the filter output therefrom.

Performance evaluation of minimum average correlation energy filters

Abstract: A new SDF type correlation filter referred to as the minimum average correlation energy (MACE) filter has been recently described in the literature. In this paper, we experimentally address the distortion tolerance and noise properties of this filter. The MACE filter has attractive properties that include: easily detectable peaks, distortion invariance, simplified training set selection, solutions to input bias effects, performance in noise and real background clutter, and less clutter with its reduced number of training set images. Each of these properties is investigated in detail in this paper.