Showing papers on "Infinite impulse response published in 1992"
23 Mar 1992
TL;DR: The authors apply the criterion used in the unbiased estimation of log spectrum to the spectral model represented by the mel-cepstral coefficients to solve the nonlinear minimization problem involved in the method and derive an adaptive algorithm whose convergence is guaranteed.
Abstract: The authors describe a mel-cepstral analysis method and its adaptive algorithm. In the proposed method, the authors apply the criterion used in the unbiased estimation of log spectrum to the spectral model represented by the mel-cepstral coefficients. To solve the nonlinear minimization problem involved in the method, they give an iterative algorithm whose convergence is guaranteed. Furthermore, they derive an adaptive algorithm for the mel-cepstral analysis by introducing an instantaneous estimate for gradient of the criterion. The adaptive mel-cepstral analysis system is implemented with an IIR adaptive filter which has an exponential transfer function, and whose stability is guaranteed. The authors also present examples of speech analysis and results of an isolated word recognition experiment. >
374 citations
TL;DR: A novel iterative algorithm for deriving the least squares frequency response weighting function which will produce a quasi-equiripple design is presented and typically produces a design which is only about 1 dB away from the minimax optimum solution in two iterations and converges to within 0.1 dB in six iterations.
Abstract: It has been demonstrated by several authors that if a suitable frequency response weighting function is used in the design of a finite impulse response (FIR) filter, the weighted least squares solution is equiripple. The crux of the problem lies in the determination of the necessary least squares frequency response weighting function. A novel iterative algorithm for deriving the least squares frequency response weighting function which will produce a quasi-equiripple design is presented. The algorithm converges very rapidly. It typically produces a design which is only about 1 dB away from the minimax optimum solution in two iterations and converges to within 0.1 dB in six iterations. Convergence speed is independent of the order of the filter. It can be used to design filters with arbitrarily prescribed phase and amplitude response. >
266 citations
TL;DR: An algorithm for the approximation of finite impulse response filters by infinite impulse response (IIR) filters is presented, based on a concept of balanced model reduction, which formulating a reduced state-space system description is input/output equivalent to the system that would more conventionally be obtained following the explicit step of constructing an (interim) balanced realization.
Abstract: An algorithm for the approximation of finite impulse response (FIR) filters by infinite impulse response (IIR) filters is presented. The algorithm is based on a concept of balanced model reduction. The matrix inversions normally associated with this procedure are notoriously error prone due to ill conditioning of the special matrix forms required. This difficulty is circumvented here by directly formulating a reduced state-space system description which is input/output equivalent to the system that would more conventionally be obtained following the explicit step of constructing an (interim) balanced realization. Examples of FIR by IIR filter approximations are included. >
168 citations
TL;DR: It is concluded that error feedback is a very powerful and versatile method for cutting down the quantization noise in any classical infinite impulse response (IIR) filter implemented as a cascade of second-order direct form sections.
Abstract: The problem of solving the optimal (minimum-noise) error feedback coefficients for recursive digital filters is addressed in the general high-order case. It is shown that when minimum noise variance at the filter output is required, the optimization problem leads to set of familiar Wiener-Hopf or Yule-Walker equations, demonstrating that the optimal error feedback can be interpreted as a special case of Wiener filtering. As an alternative to the optimal solution, the formulas for suboptimal error feedback with symmetric or antisymmetric coefficients are derived. In addition, the design of error feedback using power-of-two coefficients is discussed. The efficiency of high order error feedback is examined by test implementations of the set of standard filters. It is concluded that error feedback is a very powerful and versatile method for cutting down the quantization noise in any classical infinite impulse response (IIR) filter implemented as a cascade of second-order direct form sections. The high-order schemes are attractive for use with high-order direct form sections. >
102 citations
TL;DR: Two new lattice-based algorithms for adaptive IIR filtering and system identification are proposed, one a reinterpretation of the Steiglitz-McBride method, and the other a variation on the output error method.
Abstract: Previous attempts at applying lattice structures to adaptive infinite-impulse-response (IIR) filtering have met with gradient computations of O(N/sup 2/) complexity. To overcome this computational burden, two new lattice-based algorithms are proposed for adaptive IIR filtering and system identification, with both algorithms of O(N) complexity. The first algorithm is a reinterpretation of the Steiglitz-McBride method (1965), while the second is a variation on the output error method. State space models are employed to make the derivations transparent, and the methods can be extended to other parameterizations if desired. The set of possible stationary points of the algorithms is shown to be consistent with the convergent points obtained from the direct-form versions of the Steiglitz-McBride and output error methods, whose properties are well studied. The derived algorithms are as computationally efficient as existing direct-form based algorithms, while overcoming the stability problems associated with time-varying direct-form filters. >
99 citations
TL;DR: In this article, a finite impulse response (FIR) filter that can synthesize any fractional sample delay by a nonlinear interpolation technique is presented, and analytically closed-form solutions for the tap weights of such an FIR filter and their frequency responses are also presented.
Abstract: A finite impulse response (FIR) filter that can synthesize any fractional sample delay by a nonlinear interpolation technique is presented. Analytically closed-form solutions for the tap weights of such an FIR filter and their frequency responses are also presented. >
98 citations
TL;DR: In this paper, a closed-form expression for the impulse velocity potential of rectangular piston-like transducers, without any far field or paraxial approximation, is presented, and the complexity introduced by the geometrical discontinuities of rectangular apertures is analyzed.
Abstract: A closed‐form expression for the impulse velocity potential of rectangular pistonlike transducers, without any far field or paraxial approximation, is presented. The classical time‐domain impulse response approach is used considering free‐field, rigid baffle, and pressure release boundary conditions. Previous approaches to the rigid baffled rectangular piston require the use of superposition methods in order to find a general solution numerically. These must add or subtract, according to the field point location, the analytical expressions that were derived only for specific field points or geometrical regions. In this paper the complexity introduced by the geometrical discontinuities of rectangular apertures is analyzed. A new compacting methodology is proposed and applied to obtain a general solution for the impulse response. This new solution provides the value of the impulse response directly in the time domain, without requiring superposition methods. In addition, a closed‐form solution for the pressure impulse response is also presented. This can be useful for physical insight and a qualitative analysis of transient and continuous wave pressure fields. Also included is a description of the temporal behavior of the impulse velocity potential and the pressure impulse response for field points in different regions. The proposed solution allows an efficient and accurate computation of pressure fields under realistic excitations. Several examples illustrate the use of this new solution in the computation of transient pressure waveforms, when wideband and relatively narrow‐band excitation pulses are used. Three‐dimensional plots of the peak amplitude of the transient pressure near field are presented, and certain characteristics are analyzed using the pressure impulse response.
98 citations
TL;DR: In this article, the statistical properties of the impulse response function of double-carrier multiplication avalanche photodiodes (APDs) are determined, including the effect of dead space, i.e., the minimum distance that a newly generated carrier must travel in order to acquire sufficient energy to become capable of causing an impact ionization.
Abstract: The statistical properties of the impulse response function of double-carrier multiplication avalanche photodiodes (APDs) are determined, including the effect of dead space, i.e., the minimum distance that a newly generated carrier must travel in order to acquire sufficient energy to become capable of causing an impact ionization. Recurrence equations are derived for the first and second moments and the probability distribution function of a set of random variables that are related, in a deterministic way, to the random impulse response function of the APD. The equations are solved numerically to produce the mean impulse response, the standard deviation, and the signal-to-noise ratio (SNR), all as functions of time. >
90 citations
TL;DR: The method is best suited for signal processing applications in which 'batch' processing of the data is used, however, sequential processing, can be accommodated when delays at the beginning of a processing segment can be tolerated.
Abstract: A new method for initializing the memory registers of infinite impulse response (IIR) filters is introduced. In addition to providing improved performance as compared to other methods of initialization, this method is unique in that it makes no prior assumptions regarding the input-signal content. Therefore, the method applies equally well to a variety of IIR filter designs and applications. The method is best suited for signal processing applications in which 'batch' processing of the data is used. However, sequential processing, can be accommodated when delays at the beginning of a processing segment can be tolerated. >
73 citations
Book•
01 Jan 1992
TL;DR: In this article, the Discrete Fourier Transform (DFT) and Fast Fourier transform (FFT) are used to estimate the probability of a given signal in a continuous linear filter.
Abstract: Signals and Systems. Brief Review of Continuous Linear Filters. Sampling and the z -Transform. Recursive-Filter Design. Finite Impulse Response (FIR) and Nonrecursive Filters. The Discrete Fourier Transform and the Fast Fourier Transform. Basic Concepts of Probability Theory and Random Processes. Quantization Effects in Digital Filters. Digital Filtering Applied to Estimation: The Discrete Kalman Filter. Index.
70 citations
TL;DR: A new method involves designing a finite-impulse-response (FIR) filter satisfying the given frequency response specifications and subsequently obtaining a significantly lower order IIR filter using model reduction based on impulse-response gramians.
Abstract: A new method for the design of a linear-phase infinite-impulse-response (IIR) filter is presented. It involves designing a finite-impulse-response (FIR) filter satisfying the given frequency response specifications and subsequently obtaining a significantly lower order IIR filter using model reduction based on impulse-response gramians. The general outline of the method and a brief overview of the existing linear-phase FIR filter design and model-reduction techniques are presented. The impulse-response gramian and the model-reduction algorithm used are presented. The method is illustrated by design examples and is compared with other methods for the design of linear-phase IIR filters using equalizers. >
23 Mar 1992
TL;DR: The authors derive infinite impulse response (IIR) biorthogonal solutions based on a pair of zero-phase halfband filters derived from Butterworth half band filters.
Abstract: A class of biorthogonal systems leading to linear-phase wavelets is presented. A notable feature of this structure is that the wavelets are derived from a filter bank where the lowpass analysis filter is constrained to be a halfband filter. The authors derive finite impulse response (FIR) biorthogonal solutions from a pair of Lagrange halfband filters. They also consider infinite impulse response (IIR) biorthogonal solutions based on a pair of zero-phase halfband filters derived from Butterworth halfband filters. >
TL;DR: Fundamental constraints on the form of infinite impulse response (IIR) periodically time-varying (PTV) filters are identified, and a design technique with well-defined error and stability characteristics based on those constraints is presented.
Abstract: Fundamental constraints on the form of infinite impulse response (IIR) periodically time-varying (PTV) filters are identified, and a design technique with well-defined error and stability characteristics based on those constraints is presented. The design technique is based on the selection of poles and zeros within the time-invariant filter banks of equivalent PTV filter analysis structures. A simple example is presented to illustrate the design method, which implements the IIR PTV as a time-invariant all-feedback IIR filter of the form 1/D(z/sup P/) cascaded with an finite impulse response (FIR) PTV filter. An application of IIR PTV filters to telecommunications transmultiplexing is presented to illustrate the design method and for comparison to an existing PTV design method. The computational complexity of the resulting system compares favorably with that of existing transmultiplexers. >
10 May 1992
TL;DR: In this article, a novel scheme for implementing a perfect reconstruction filter bank using infinite-impulse-response (IIR) all-pass filters for the processing of infinite length input signals is described.
Abstract: A novel scheme for implementing a perfect reconstruction filter bank using infinite-impulse-response (IIR) all-pass filters for the processing of infinite length input signals is described. The scheme uses double buffers to implement noncausal, stable synthesis filters operating on infinite length sequences. Finite word-length effects on the performance of the overall system are discussed, and various applications for this structure are proposed. >
TL;DR: A new algorithm is proposed on the basis of the least mean square equation error (LMSEE) algorithm, which manages to remedy the bias while retaining the parameter stability in the IIR system identification and adaptive filtering.
Abstract: In the area of infinite impulse response (IIR) system identification and adaptive filtering the equation error algorithms used for recursive estimation of the plant parameters are well known for their good convergence properties. However, these algorithms give biased parameter estimates in the presence of measurement noise. A new algorithm is proposed on the basis of the least mean square equation error (LMSEE) algorithm, which manages to remedy the bias while retaining the parameter stability. The so-called bias-remedy least mean square equation error (BRLE) algorithm has a simple form. The compatibility of the concept of bias remedy with the stability requirement for the convergence procedure is supported by a practically meaningful theorem. The behavior of the BRLE has been examined extensively in a series of computer simulations. >
10 May 1992
TL;DR: In this paper, the stability of sigma-delta modulators with second-order FIR (finite impulse response) noise transfer functions was studied and the results of a simple computer algorithm proved the zero-input stability of 94% of the modulators.
Abstract: The authors study the stability of sigma-delta modulators with second-order FIR (finite impulse response) noise transfer functions. At present, there is no analytical criterion which is both necessary and sufficient for the stability of such systems, even when the input is known. The authors present a variety of tests, culminating with the results of a simple computer algorithm which proves the zero-input stability of 94% of the modulators of the foregoing type which are, according to brute-force simulations, believed to be stable. The algorithm may be extended to higher-order FIR, or even IIR (infinite impulse response) transfer functions and may also be modified to incorporate bounded inputs. >
TL;DR: In this article, an effective eigenfilter approach is proposed for designing FIR filters with complex-valued frequency responses by minimizing a quadratic measure of the error in the frequency band, an eigenvector of an appropriate matrix is computed to get the filter coefficients.
Abstract: An effective eigenfilter approach is proposed for designing FIR filters with complex-valued frequency responses. By minimizing a quadratic measure of the error in the frequency band, an eigenvector of an appropriate matrix is computed to get the filter coefficients. The algorithm is easy and optimal in the least squares sense; it is used to design to major classes of FIR filters, including multiband filters, differentiators, Hilbert transformers, and all-pass phase. Several examples and comparisons to the existing linear programming methods are presented to demonstrate the flexibility and effectiveness of this approach. >
TL;DR: In this article, the authors describe the design and VLSI implementation of a single-chip 85-MHz fourth-order infinite impulse response (IIR) digital filter chip fabricated in 0.9-mu m CMOS technology.
Abstract: The authors describe the design and VLSI implementation of a single-chip 85-MHz fourth-order infinite impulse response (IIR) digital filter chip fabricated in 0.9- mu m CMOS technology. The coefficient and input data word lengths of the filter are 10 b each, and the output data word length is 15 b. The coefficients are fully programmable. The chip can be programmed to implement any IIR filter from first to fourth order or an FIR filter up to 16th order at sample rates up to 85 MHz. A total of seventeen 10*10 multiply-add modules are used in this chip. The chip contains 80000 devices in an active area of 14 mm/sup 2/. It dissipates 2.2 W at 85-MHz clock rate and performs over 1.5*10/sup 9/ multiply-add operations per second. The underlying filtering algorithm, chip architecture, circuit and layout design, speed issues, and test results are described. The results of an E-beam probing experiment on packaged chips at 100-MHz clock rates are also presented and discussed. >
10 May 1992
TL;DR: In this paper, the authors examined various iterative reweighted least squared error algorithms to obtain an L/sub p/ approximation for designing a linear phase FIR (finite impulse response) filter.
Abstract: The authors examine various iterative reweighted least squared error algorithms to obtain an L/sub p/ approximation for designing a linear phase FIR (finite impulse response) filter. These methods consider 1 >
26 Oct 1992
TL;DR: Simulation results are presented to show how novel approaches to adaptive digital filtering based on genetic algorithms and simulated annealing are able to tackle the problems of global optimality and dimensionality when adapting high-order IIR filters.
Abstract: Novel approaches to adaptive digital filtering based on genetic algorithms (GAs) and simulated annealing (SA) are proposed. Algorithms based on using the gradient of the mean square error or on least-square principles have been found to have inadequacies when adapting IIR filters. The process of adaptation can be cast as an optimization problem. GAs are used in the context of multiparameter optimization. Simulation results are presented to show how these approaches are able to tackle the problems of global optimality and dimensionality when adapting high-order IIR filters. Hybrid schemes where concepts of SA are incorporated into GAs are proposed. >
TL;DR: The linear smoothed Newton (LSN) predictor is extended in this work by including a recursive term in the basic transfer function and cascading the rest of the successive difference paths with appropriately delayed extrapolation filters of corresponding polynomial orders, which leads to computationally efficient IIR predictors with significantly lowered gain at the higher frequencies.
Abstract: Newton predictors have considerable gain at the higher frequencies, which reduces their applicability to practical signal processing where the narrowband primary signal is often corrupted by additive wideband noise. Two modifications that can be used to extrapolate low-order polynomials have been proposed. In both approaches, the highest order difference of successive input samples, approximating the constant nonzero derivative, is smoothed before it is added to the lower order differences, reducing the undesired noise gain. The linear smoothed Newton (LSN) predictor is extended in this work by including a recursive term in the basic transfer function and cascading the rest of the successive difference paths with appropriately delayed extrapolation filters of corresponding polynomial orders. This leads to computationally efficient IIR predictors with significantly lowered gain at the higher frequencies. The recursive predictor is analyzed in the time and frequency domains and compared to the other predictors. >
Abstract: The roundoff noise properties of floating point digital filters are examined. To make the analysis tractable, a high level model to deal with the errors in the inner product operation is developed. This model establishes a broad connection between coefficient sensitivity and roundoff noise. Along with the model, an efficient procedure to keep track of the addition scheme used in the inner product, and to compute the statistics of the errors is introduced. A systematic procedure based on the model is then developed to derive general expressions for the roundoff noise of FIR, direct form IIR, and state-space filters. The expressions in the context of state space filters are explored in some detail. Optimality issues are considered, and it is shown that when double precision accumulation is used, the optimal filters are similar in nature to those derived in the context of fixed point arithmetic with the essential difference that they also do depend on the spectrum of the input signal. Optimality with respect to addition schemes, and second-order filters are also examined in some detail. >
TL;DR: A room simulator has been developed as part of a project involving virtual acoustic environments, similar to auditorium simulators for home use, and its implementation, design tradeoffs, and results are discussed.
Abstract: A room simulator has been developed as part of a project involving virtual acoustic environments. The system is similar to auditorium simulators for home use. The simulated reverberant field is rendered using four to six loudspeakers evenly spaced around the perimeter of a listening area. Listeners are not constrained to any particular orientation, although best results are obtained near the center of the space. The simulation is driven from a simple description of the desired room and the location of the sound source. The system accepts monophonic input sound and renders the simulated reverberant field in real time. Early echo generation is based on the source image model, which determines a finite impulse response filter per output channel. Diffuse reverberant field generation is accomplished using infinite impulse response reverberators based on nested and cascaded allpass filters. The system is implemented using Motorola 56001 digital signal processors, one per output channel. The talk will discuss implementation, design tradeoffs, and results.
TL;DR: A high-speed algorithm for calculating the square root which has higher execution speed and smaller calculation error and is implemented on the commercially available DSP (TM320C25) and is compared with the Newton-Raphson method.
Abstract: A high-speed algorithm for calculating the square root is proposed. This algorithm, which can be regarded as calculation of the step response of a kind of nonlinear IIR filter, requires no divisions. Therefore, it is suitable for a VLSI digital signal processor (DSP) which has a high-speed hardware multiplier but does not usually have a high-speed hardware divider. The convergence properties of the algorithm are analyzed and used to develop a practical implementation of the procedure. It is implemented on the commercially available DSP (TM320C25) and is compared with the Newton-Raphson method. The proposed algorithm has two advantages over the Newton-Raphson method: higher execution speed and smaller calculation error. >
01 Jan 1992
TL;DR: An algorithm-independent lower bound on the achievable approximation error is derived, and an approximation method that involves the solution of a fixed number of all-pass (Nehari) extension problems (and is therefore called the Nehari shuffle) is presented.
Abstract: This paper presents a new approach to the problem of designing a finite impulse response filter of specified length, q, which approximates in uniform frequency (L,) norm a given desired (possibly infinite impulse response) causal, stable filter transfer function. We derive an algorithm-independent lower bound on the achievable approximation error and then present an approximation method which involves the solution of a fixed number of all-pass (Nehari) extension problems and so is called the Nehari shuffle. Upper and lower bounds on the approxi- mation error are derived for the algorithm. These bounds are calculable a priori so the length of filter required to satisfy a given maximum error can be found before designing the filter. Examples indicate that the method closely approaches the de- rived global lower bound. We compare the new method with the Preuss (complex Remez exchange) algorithm in some ex- amples.
TL;DR: The use of resistor networks to implement 2D spatial IIR and FIR filters operating in real time is described and a correspondence is drawn between such networks and digital signal processors.
Abstract: The use of resistor networks to implement 2D spatial IIR and FIR filters operating in real time is described. A correspondence is drawn between such networks and digital signal processors. A broad array of filtering functions can be realized using networks of positive resistors, voltage-controlled resistors, and negative resistors. The transfer functions effected by networks of this type are precisely defined, and the applications, limitations, and design of two-dimensional resistive mesh computers are discussed. >
TL;DR: In this article, an approach to the problem of designing a finite impulse response filter of specified length q which approximates in uniform frequency (L/sub infinity /) norm a given desired (possibly infinite impulse response) causal, stable filter transfer function is presented.
Abstract: An approach to the problem of designing a finite impulse response filter of specified length q which approximates in uniform frequency (L/sub infinity /) norm a given desired (possibly infinite impulse response) causal, stable filter transfer function is presented. An algorithm-independent lower bound on the achievable approximation error is derived, and an approximation method that involves the solution of a fixed number of all-pass (Nehari) extension problems (and is therefore called the Nehari shuffle) is presented. Upper and lower bounds on the approximation error are derived for the algorithm. Examples indicate that the method closely approaches the derived global lower bound. The method is compared with the Preuss (complex Remez exchange) algorithm in some examples. >
TL;DR: The identification of non-minimum-phase finite-impulse-response (FIR) systems driven by third-order stationary colored signals that are not linear processes is addressed and the linear part of the bispectrum of a signal is discussed.
Abstract: The identification of non-minimum-phase finite-impulse-response (FIR) systems driven by third-order stationary colored signals that are not linear processes is addressed Modeling the linear part of the bispectrum of a signal is discussed The bispectrum of a signal is decomposed into two multiplicative factors The linear bispectrum is defined as the factor of the bispectrum that can exactly be modeled using a third-order white-noise-driven linear shift-invariant (LSI) system The linear bispectrum of the output of the unknown LSI system is represented using an ARMA (autoregressive moving average) model, where the MA parameters correspond to the unknown FIR system impulse response coefficients, and the AR parameters model the linear bispectrum of the input signal An algorithm for identifying the MA and AR parameters is given How the proposed method is different from fitting an ARMA model directly to the bicumulants or the bispectrum of the system output is discussed The method is applied to blur identification >
TL;DR: In this article, a second order IIR adaptive notch filter was developed to control howling in a speakerphone system, which achieved an 8-14 dB howling margin enhancement with this new technique.
Abstract: A new second order IIR adaptive notch filter which has a faster convergence rate was developed to control howling in a speakerphone system. Simulation results obtained using real hybrid and acoustic transfer functions show an 8-14 dB howling margin enhancement with this new technique.
TL;DR: It is shown that in all the applications, the goal of adaptation is met whenever a matching condition and a positive real condition are satisfied, and a special case of the results therefore resolves the long-standing problem of the convergence and the unbiasedness of the output error identification scheme in the presence of colored noise.
Abstract: General stochastic parallel model adaptation problems that consist of an unknown linear time-invariant system and a partially or wholly tunable system connected in parallel, with a common input, are considered. The goal of adaptation is to tune the partially tunable system so that its output matches that of the unknown system, despite the presence of any disturbance which is stochastically uncorrelated with the input. The general formulation allows applications to adaptive feedforward control and adaptive active noise canceling with input contamination, in addition to output error identification and adaptive IIR filtering. It is shown that in all the applications, the goal of adaptation is met whenever a matching condition and a positive real condition are satisfied. A special case of the results therefore resolves the long-standing problem of the convergence and the unbiasedness of the output error identification scheme in the presence of colored noise. A simple general technique for analyzing the strong consistency of parameter estimation with projection is also developed. >