Showing papers on "Infinite impulse response published in 1984"
TL;DR: In this article, a cascade of two sections is proposed for finite impulse response (FIR) digital filters, where the first section generates a sparse set of impulse response samples and the other section generates the remaining samples by using interpolation.
Abstract: A new approach to implement computationally efficient finite impulse response (FIR) digital filters is presented. The filter structure is a cascade of two sections. The first section generates a sparse set of impulse response samples and the other section generates the remaining samples by using interpolation. The method can be used to implement most practical FIR filters with significant savings in the number of arithmetic operations. Typically 1/2 to 1/8 of the number of multipliers and adders of conventional FIR filters are required in the implementation. The saving is achieved both in the linear phase and the non-linear phase cases. In addition, the new implementation gives smaller coefficient sensitivities and better roundoff noise properties than conventional implementations.
440 citations
TL;DR: A tutorial-style framework is presented for understanding the current status of adaptive infinite-impulse-response (IIR) filters and the structures of provable convergent adaptive algorithms are derived.
Abstract: A tutorial-style framework is presented for understanding the current status of adaptive infinite-impulse-response (IIR) filters The paper begins with a detailed discussion of the difference equation models that are useful as adaptive IIR filters The particular form of the resulting prediction error generic to adaptive IIR filters is highlighted and the structures of provable convergent adaptive algorithms are derived A brief summary of particular, currently known performance properties, drawn principally from the system identification literature, is followed by the formulation of three illustrative adaptive signal processing problems, to which these adaptive IIR filters are applicable The concluding section discusses various open issues raised by the formulation of this framework
236 citations
TL;DR: In this article, the authors present filtering methods for interfacing time-discrete systems with different sampling frequencies, which are applicable for sampling rate conversion between any two sampling frequencies; the conversion ratio may even be irrational or slowly time varying.
Abstract: The paper presents filtering methods for interfacing time-discrete systems with different sampling frequencies. The methods are applicable for sampling rate conversion between any two sampling frequencies; the conversion ratio may even be irrational or slowly time varying. Interpolation by irrational factors requires digital filters with nonperiodically varying coefficients. This is dealt with in two ways. 1) All possible coefficient values are precalculated. This is, in a sense, possible because of the finite resolution needed. Or 2), the coefficients can be updated in real time using either FIR or IIR filters. The first solution requires a huge coefficient memory; the second scheme, on the other hand, is computationally intensive. While discussing both of these solutions, more practical intermediate schemes incorporating both FIR-and IIR-type filters are suggested. The suggested practical implementations are either based on analog reconstruction filters where the derived digital filter coefficients are functions of the distances between current input and output samples or digital interpolators combined with simple analog interpolation schemes for finding the desired values in between the uniform output samples from the digital interpolator.
169 citations
TL;DR: In this article, the simultaneous design in both magnitude and group-delay of digital transfer functions on the basis of multiple criterion optimization is considered, and both causal IIR filters and nonlinear phase FIR filters are studied.
Abstract: This work considers the simultaneous design in both magnitude and group-delay of digital transfer functions on the basis of multiple criterion optimization. Both causal IIR filters and nonlinear phase FIR filters are studied. Examples of the optimal tradeoff filters, both FIR filters and IIR filters, are presented and their characteristics are analyzed.
86 citations
TL;DR: The proposed infinite impulse response filter has a special structure that guarantees the desired transfer characteristics and is derived using a general prediction error framework.
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.
82 citations
DOI•
01 Oct 1984
TL;DR: Efficient systolic arrays for matrix and vector multiplication, at both word and bit levels, are described and three applications relevant for signal processing are outlined: a convolver, an IIR filter and a linear classifier.
Abstract: Matrix and vector multiplications are widely used in signal processing in operations such as FIR and IIR filtering, feature extraction and classification. Frequently, signal processing must be done in real time requiring the use of special purpose VLSI hardware. Regular structures such as systolic arrays are well suited for matrix and vector operations and are also amenable to VLSI implementation. This paper describes efficient systolic arrays for matrix and vector multiplication, at both word and bit levels, and outlines three applications relevant for signal processing: a convolver, an IIR filter and a linear classifier.
62 citations
TL;DR: An overview of adaptive prediction in differential pulse code modulation systems used for speech encoding at 16 to 32 kilobits/sec.(kbps) is presented, and comparative performances results of several backward adaptive algorithms and predictor structures are discussed.
Abstract: An overview of adaptive prediction in differential pulse code modulation (DPCM) systems used for speech encoding at 16 to 32 kilobits/sec(kbps) is presented Features of the paper include a discussion of both infinite impulse response and finite impulse response predictors and a development of the various predictor implementation structures, such as the direct or transversal form, the lattice form, and the cascade of second order sections form Differences between forward and backward adaptation are described, and comparative performances results of several backward adaptive algorithms and predictor structures are discussed
34 citations
TL;DR: In this paper, a new structure for the block realization of IIR, 2-D digital filters is proposed, which is based on a matrix representation of 2D convolutions and results in a 2D state variable description with block feedback.
31 citations
01 Mar 1984
TL;DR: A model based on research in the psychophysics of vision is developed for use in the design of image processing filters in order to quantify the results of imageprocessing as perceived by a human observer.
Abstract: A model based on research in the psychophysics of vision is developed for use in the design of image processing filters in order to quantify the results of image processing as perceived by a human observer. An alternate formulation of the classical frequency-domain filter design problem with both space and frequency-domain specifications is developed. New experimental results concerning the masking effect in the vicinity of edges are reported, and a structure of a model for detection of distortion in complex images is proposed. The ability of the model to predict the visibility of one-dimensional patterns in the vicinity of edges is tested. A distortion measure based on the model is introduced, and practical filter-design criteria are developed. Both FIR linear phase filters and IIR filters are used to demonstrate the applicability of the methods.
25 citations
TL;DR: It is shown that an aperiodic discrete-continuous sampling system may be modelled by a difference equation matrix which may be used in the same way as in the case of periodic discrete systems, which allows the computation of the outputs as finite combinations of previous inputs and outputs.
Abstract: It is 8hown that an aperiodic discrete-continuous sampling system may be modelled by a difference equation matrix which may be used in the same way as in the case of periodic discrete systems. This fact allows the computation of the outputs as finite combinations of previous inputs and outputs. This is accomplished using a finite set of time-varying parameters, which are dependent on the parameters of the continuous transfer matrix and on a finite set of sampling periods equal to the order of the system state, and thus avoids impulse response methods and state variable measurements.
23 citations
TL;DR: In this article, an implementation for maximally-flat FIR filters is proposed that requires a much smaller number of multiplications than a direct form structure, and the values of the multiplier coefficients in the implementation are conveniently small, and do not span a huge dynamic range.
Abstract: An implementation for maximally-flat FIR filters is proposed that requires a much smaller number of multiplications than a direct form structure. The values of the multiplier coefficients in the implementation are conveniently small, and do not span a huge dynamic range, unlike in a direct form implementation.
TL;DR: In this article, the authors considered free-field, baffled, and pressure release boundary conditions, in the case of an arbitrary spatial distribution of combined, amplitude-time delay modulation in an aperture.
Abstract: Acoustic diffraction of plane impulsive waves is considered for free‐field, baffled, and pressure release boundary conditions, in the case of an arbitrary spatial distribution of combined, amplitude‐time delay modulation in an aperture. The method is based on the impulse response analysis of parallel aperture lines, the line impulse responses being then integrated to give an aperture impulse response. A closed‐form, analytical expression is derived for lines having an arbitrary amplitude modulation. In the case of nonmodulated apertures of an arbitrary shape, the aperture impulse response is also of an analytical form, directly involving the aperture contour line. The analysis is performed with emphasis on maintaining clear physical sense of successive evaluation steps.
01 Jan 1984
TL;DR: In this article, an improved SAW-filters with a multstrlp coupler, two weighted transducers and minimum length is presented, where the desired frequency response is approximated by a FIR-filter.
Abstract: An improved design of SAW-filters with a multlstrlp coupler, two weighted transducers and minimum length is presented. The desired frequency response is approximated by a FIR-filter. Using llnear programming techniques a smooth passband is obtained. This frequency response is shared out to both transducers by the z ero separation method. The difference between their responses is minimized allowing asymmetric time functions. Finally a procedure to compensate for circuit and diffraction effects is proposed.
TL;DR: Through this technique, pipelined IIR filters based on RNS Read-Only-Memory table look-up techniques can be designed which offer throughput rates equal to the tableLook-up time of the ROM's, and the increase in speed will often justify the additional hardware.
Abstract: The well-known advantages of pipelining as applied to Finite Impulse Response (FIR) Residue Number System (RNS) arithmetic digital filters is extended to the important area of Infinite Impulse Response (IIR) digital filters through a new technique based on augmentation of the IIR transfer function. Through this technique, pipelined IIR filters based on RNS Read-Only-Memory (ROM) table look-up techniques can be designed which offer throughput rates equal to the table look-up time of the ROM's. This high-speed realization can be achieved even though the recursive filter algorithm requires multiple delays in realizing the output of the filter. For the example of a typical second-order IIR filter, the pipelined structure represents a five-fold increase in speed over standard techniques. Higher order realizations will yield proportionately higher speed improvements. Although the new technique does increase somewhat the hardware complexity of the filter, the increase in speed will often justify the additional hardware. The paper discusses the basic technique, stability considerations, and hardware realizations.
01 Jan 1984
TL;DR: It is shown that for a large class of laws, unequalized impulse responses with few nonzero coefficients are points of convergence, and that there exist undesired local minima for a subset of functions that includes those previously proposed.
Abstract: We consider the adaptive equalization of an unknown linear time-invariant channel without observations of the input sequence, by updating the impulse response coefficients of the equalizer with the output of the channel times a memoryless nonlinear function of the equalizer output. To date, no such function is known to result in global convergence to the inverse of the channel when the input consists of binary data. The effect of the selection of the memoryless nonlinearity in the convergence properties of the adaptive scheme is studied, and it is shown that for a large class of laws (including the continuous functions), unequalized impulse responses with few nonzero coefficients are points of convergence, and that there exist undesired local minima for a subset of functions that includes those previously proposed.
TL;DR: In this article, the authors proposed a digital line frequency interference elimination algorithm using finite impulse response (FIR) and infinite impulse response(IIR) discretetime filters, which is shown to be easier to implement in wideband multiple notch applications than conventional and adaptive digital filters.
Abstract: This paper presents a practical approach for, cancelling a periodic interference from signal measurements. In the method the waveform of the interference is continuously estimated and subtracted from the measurement. The rejection properties are made independent of small variations in the fundamental frequency of the interference by fixing the sampling rate to be an exact multiple of it. Special attention is paid to the line frequency interference rejection problem where the notch at each harmonic is required to be very narrow although the sampling frequency is high if compared with the lowest notch frequency. Implementations using finite impulse response (FIR) and infinite impulse response (IIR) discretetime filters are discussed. The proposed filters are shown to be easier to implement in wideband multiple notch applications than conventional and adaptive digital filters. A digital line frequency interference eliminator requiring only two multipliers has been constructed using an IIR filter. All of its 128 notches are 1-Hz wide and over 70-dB deep at 50 Hz and at its harmonics. The sampling frequency is 25.6 kHz and the available signal bandwidth is 6.4 kHz. The compensation of the low-frequency behavior of the digital IIR filter has been accomplished by combining appropriately analog filters with the digital part. The resulting frequency response with essentially linear phase characteristics starts from zero frequency. Through multiplexing the filter can also be used as a two- or four-channel system.
TL;DR: In this article, a class of recursive digital filters having equiripple behavior in both the magnitude and group delay responses is introduced, which consist of an all-pole IIR component cascaded with a linear phase FIR component.
Abstract: This paper introduces a class of recursive digital filters having equiripple behavior in both the magnitude and group delay responses. The filters consist of an all-pole IIR component cascaded with a linear phase FIR component. The IIR component is synthesized so that its group delay exhibits an equiripple variation in the passband, whereas the FIR component is used to obtain the desired equiripple nature for the amplitude response in both the passband and stopband(s). An iterative procedure is presented for optimizing the ripple in the group delay and the order of the IIR part in such a way that the overall multiplication rate attains its minimum. Examples show that standard simultaneous amplitude and phase optimization methods only slightly change the results obtained using this method. The new method is straightforward to implement and avoids the problems of guessing good initial values and of premature termination. Several examples show that the new filters require significantly fewer multipliers in narrow-band applications than equivalent FIR and delay equalized elliptic designs at the expense of a negligible variation in the group delay.
01 Mar 1984
TL;DR: It is shown that a general two-dimensional transfer function can be exactly realized in terms of polynomials of order one, each one of which is a function of one of the two space variables only.
Abstract: A new approach to the realization of two-dimensional (2-d) FIR and IIR digital filters is introduced based on the so-called "Lower-Upper triangular (LU) decomposition" of matrix coefficients of their two-dimensional polynomials. It is shown that a general two-dimensional transfer function can be exactly realized in terms of polynomials of order one, each one of which is a function of one of the two space variables only. It is demonstrated that the LU realization scheme exhibits high inherent parallelism as well as great modularity, regularity and flexibility. This paper also shows that the computational requirements of the LU realization are much less than these of the Jordan (J), Singular Value (SV) and canonical realization schemes.
01 Mar 1984
TL;DR: The interactions between digital filter design and VLSI design methodology and the constraints imposed by the MOS-architecture are discussed, and an appropriate filter structure is proposed.
Abstract: The interactions between digital filter design and VLSI design methodology are discussed. The constraints imposed by the MOS-architecture are discussed, an appropriate filter structure is proposed. The mapping of this structure to a VLSI-realization is shown for intermediate and high sample rates. Design examples for FIR and IIR filters for video applications and lowpass filters for decimation and interpolation are given.
TL;DR: In this article, the problem of computationally efficient design of finite impulse response Wiener filters with linear phase is considered, and a fast order recursive algorithm for solving these normal equations is developed.
Abstract: In this paper, the problem of computationally efficient design of finite impulse response Wiener filters with linear phase is considered. For stationary signals these filters are determined by the solution of normal systems of equations whose associated matrix has a Toeplitz-plus (or minus)-Hankel structure. A fast order recursive algorithm for solving these normal equations is developed. The componentparts of this scheme are the Levinson algoritun and an elegant relationship connecting Wiener filters and Wiener filters with linear phase. Finally, it is shown that the well known two multiplier lattice structure introduced by Itakura and Saito can be used in a natural way to realize a ladder Wiener filter with linear phase.
TL;DR: In this article, a stray-insensitive switched-capacitor (SC) implementation of a finite impulse response (FIR) filter is described. But the implementation is limited to a single SC.
Abstract: Finite impulse response (FIR) filters can be realized without multipliers and delays by delta modulating the input signal [5]. In this letter, we describe a stray-insensitive switched-capacitor (SC) implementation of such delta modulation FIR filters.
TL;DR: In this paper, it was shown that the approximation problem for minimum phase FIR filters with arbitrary specifications on the attenuation can be solved by a method published in [3] although that method is based on the same idea of that developed in the above \footnote[1]{paper, it is simpler and also easier to apply.
Abstract: It is shown that the approximation problem for minimumphase FIR filters with arbitrary specifications on the attenuation can be solved by a method published in [3] although that method is based on the same idea of that developed in the above \footnote[1]{paper}, it is simpler and also easier to apply.
TL;DR: Simulations of numerous impulse voltage test and neasuring circuits show that subject to restrictions cn the time to steady state a set of three step response pararneters is necessary and sufficient to Limit response errors.
Abstract: Simulations of numerous impulse voltage test and neasuring circuits show that subject to restrictions cn the time to steady state a set of three step response pararneters is necessary and sufficient to Limit response errors. Fewer paraneters can only be used for special cases.
TL;DR: In this paper, a general approach to the design of single-rate fractional-step delay (FSD) filters for array beamforming, based on a state-space formulation, is presented.
Abstract: A general approach to the design of single-rate fractional-step delay (FSD) filters for array beamforming, based on a state-space formulation, is presented. This approach is applicable to both FIR and IIR implementations. In this approach, state-space realizations of FSD filters can be derived from state-space realizations of parent interpolating filters. The realizations so produced have the property that the fractional-step delay is determined solely by the B (input coupling) matrix. The A (system) and C (output coupling) matrix operations are independent of the fractional-step delay, and thus can be shared by all array element channels. All FSD beamformers are subject to spurious spatial response lobes generated by aliasing in the interpolating filter. However, these spurious lobes can be effectively suppressed by appropriate design of the magnitude response of the interpolating filter. If the magnitude response of the interpolating filter is chosen so that the spurious sidelobes are effectively suppressed, then although the phase response of the interpolating filter will affect the temporal response of the system, it will have no effect on the spatial response of the array; i.e., it is not necessary to have linear phase response in the interpolating filter to obtain ideal beam patterns. Comparisons of computation rates for FIR and IIR implementation of FSD beamformers are presented.
TL;DR: In this paper, a procedure for the minimal realisation of an impulse response matrix is proposed, based on the expansion of moments of the impulse response matrices about an arbitrary point a. The procedure is illustrated with the help of an example.
Abstract: A procedure for the minimal realisation of an impulse response matrix is proposed. The procedure is based on the expansion of moments of the impulse response matrix about an arbitrary point a. The procedure is illustrated with the help of an example.
TL;DR: In this article, a new technique is described to synthesize a finite impulse response of linear time-variant (LTV) digital filter, which is decomposed into a sum of products of two orthogonal sequences.
Abstract: The paper describes a new technique to synthesize a finite impulse response of linear time-variant (LTV) digital filter. First, a finite impulse response is decomposed into a sum of products of two orthogonal sequences. The direct implementation of the decomposed impulse response leads to the parallel connection of linear time-invariant (LTI) digital filters, followed by time-varying multipliers. A simple filter structure is obtained by properly modifying the sequences to realize the parallel form structure as a cascade connection of first- or second-order recursive LTI filters. The structure is easy to implement on a computer and saves computation time. Numerical examples illustrating the technique are included.
TL;DR: A simple theorem giving the necessary and sufficient conditions for the recursibility of N-D IIR digital filters is proven and a method for checking the compatibility of a desired recursion direction with respect to the specified filter output mask structure is developed.
Abstract: A simple theorem giving the necessary and sufficient conditions for the recursibility of N-D IIR digital filters is proven. This theorem is used to develop a linear programming technique to test the recursibility of a given filter. The concept of N-D sector is introduced and used to model the structure of N-D recursive filters. These results lead to a method for checking the compatibility of a desired recursion direction with respect to the specified filter output mask structure.
TL;DR: In this article, a near-optimum method for the design of one-dimensional IIR digital filters with coefficients of finite precision is proposed, which decreases the number of iterations required by branch and bound algorithms without appreciably degrading the accuracy of the resulting design.
Abstract: A near-optimum method for the design of one-dimensional IIR digital filters with coefficients of finite precision is proposed. The application of this method decreases the number of iterations required by ‘ branch and bound ’ algorithms, without appreciably degrading the accuracy of the resulting design.
01 Mar 1984
TL;DR: This paper presents a technique which can be used to eliminate the edge transients associated with data truncation for spatial domain filter applications.
Abstract: An image may be considered to be a finite extent two dimensional sequence with truncation occuring at the edges of the image. The corresponding data set is equivalent to the product of a two dimensional rectangular window function and an infinite duration two dimensional function. The edge transients associated with data truncation can propagate throughout the output image when infinite impulse response spatial domain filters are used. These transient effects can completely obscure the output image with certain types of filters. This paper presents a technique which can be used to eliminate the edge transients associated with data truncation for spatial domain filter applications.
01 Mar 1984
TL;DR: A new version of an optimal algorithm used to design finite wordlength linear phase FIR (Finite Impulse Response) digital filters is considered, which allows the designer to introduce constraints in the discrete optimization problem.
Abstract: We propose some improvements to an optimal algorithm used to design finite wordlength linear phase FIR (Finite Impulse Response) digital filters. We consider also a new version, which allows the designer to introduce constraints in the discrete optimization problem. Examples are given concerning the optimal design of : a filter without multipliers, a third-band filter, a low-pass filter with a constraint on step response ripple.