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Showing papers on "Filter design published in 1998"


Book
16 Dec 1998
TL;DR: In this paper, the authors present an approach for the estimation of 1st and 2nd-order functions of LC ladder filters using Opamps, which is based on the Bessel-Thomson Delay Approximation Delay Equalization Frequency Transformations.
Abstract: Fundamentals Introduction Filter Characterization Types of Filters Steps in Filter Design Analysis Continuous-Time Filter Functions Stability Passivity for One- and Two-Port Networks Reciprocity The Approximation Problem Introduction Filter Specifications and Permitted Functions Formulation of the Approximation Problem Approximation to the Ideal Lowpass Filter Filters with Linear Phase: Delays Bessel-Thomson Delay Approximation Delay Equalization Frequency Transformations Design Tables of Passive LC Ladder Filters Impedance Scaling Predistortion Active Elements Introduction Ideal Controlled Sources Impedance Transformation (Generalized Impedance Converters and Inverters) Negative Resistance Ideal Operational Amplifier The Ideal Operational Transconductance Amplifier (OTA) Realization of 1st- and 2nd-Order Functions Using Opamps Introduction Realization of 1st-Order Functions The General 2nd-Order Filter Function Sensitivity of 2nd-Order Filters Realization of Biquadratic Functions Using SABs Realization of a Quadratic with a Positive Real Zero Biquads Obtained Using the Twin-Tee RC Network Two Opamp Biquads Three Opamp Biquads Realization of High-Order Functions Using Opamps Introduction Selection Criteria for High-Order Function Realizations Mutliparameter Sensitivity High-Order Function Realization Methods Cascade Connection of 2nd-Order Sections Mutli-Loop Feedback Filters Cascade of Biquartics Simulation of LC Ladder Filters Using Opamps Introduction Resistively Terminated Lossless LC ladder Filters Methods of LC Ladder Filter Simulation The Gyrator Generalized Impedance Converter FDNRs Complex Impedance Scaling Functional Simulation Wave Active Filters Introduction Wave Active Filters Wave Active Equivalents (WAE) Economical Wave Active Filters Sensitivity of WAFs Operation of WAFs at Higher Frequencies Complementary Transfer Functions Wave Simulation of Inductance Linear Transformation Active Filters (LTA Filters)

377 citations


Journal ArticleDOI
TL;DR: The relations of non-subsampled filter banks to continuous-time filtering are investigated and the design flexibility is illustrated by giving a procedure for designing maximally flat two-channel filter banks that yield highly regular wavelets with a given number of vanishing moments.
Abstract: Perfect reconstruction oversampled filter banks are equivalent to a particular class of frames in l/sup 2/(Z). These frames are the subject of this paper. First, the necessary and sufficient conditions of a filter bank for implementing a frame or a tight frame expansion are established, as well as a necessary and sufficient condition for perfect reconstruction using FIR filters after an FIR analysis. Complete parameterizations of oversampled filter banks satisfying these conditions are given. Further, we study the condition under which the frame dual to the frame associated with an FIR filter bank is also FIR and give a parameterization of a class of filter banks satisfying this property. Then, we focus on non-subsampled filter banks. Non-subsampled filter banks implement transforms similar to continuous-time transforms and allow for very flexible design. We investigate the relations of these filter banks to continuous-time filtering and illustrate the design flexibility by giving a procedure for designing maximally flat two-channel filter banks that yield highly regular wavelets with a given number of vanishing moments.

369 citations


Journal ArticleDOI
TL;DR: Design algorithms for hybrid filter banks (HFBs) for high-speed, high-resolution conversion between analog and digital signals are presented and a gain normalization technique is developed to maximize the dynamic range in the finite-precision implementation.
Abstract: This paper presents design algorithms for hybrid filter banks (HFBs) for high-speed, high-resolution conversion between analog and digital signals. The HFB is an unconventional class of filter bank that employs both analog and digital filters. When used in conjunction with an array of slower speed converters, the HFB improves the speed and resolution of the conversion compared with the standard time-interleaved array conversion technique. The analog and digital filters in the HFB must be designed so that they adequately isolate the channels and do not introduce reconstruction errors that limit the resolution of the system. To design continuous-time analog filters for HFBs, a discrete-time-to-continuous-time ("Z-to-S") transform is developed to convert a perfect reconstruction (PR) discrete-time filter bank into a near-PR HFB; a computationally efficient algorithm based on the fast Fourier transform (FFT) is developed to design the digital filters for HFBs. A two-channel HFB is designed with sixth-order continuous-time analog filters and length 64 FIR digital filters that yield -86 dB average aliasing error. To design discrete-time analog filters (e.g., switched-capacitors or charge-coupled devices) for HFBs, a lossless factorization of a PR discrete-time filter bank is used so that the reconstruction error is not affected by filter coefficient quantization. A gain normalization technique is developed to maximize the dynamic range in the finite-precision implementation. A four-channel HFB is designed with 9-bit (integer) filter coefficients. With internal precision limited to the equivalent of 15 bits, the maximum aliasing error is -70 dB, and with the equivalent of 20 bits internal precision, maximum aliasing is -100 dB. The 9-bit filter coefficients degrade the stopband attenuation (compared with unquantized coefficients) by less than 3 dB.

241 citations


Journal ArticleDOI
TL;DR: In this article, two filters for improving the visibility of crystalline material in the presence of amorphous surface contamination layers in high-resolution electron microscope images can be constructed automatically from the information present in the Fourier transform of the recorded image.
Abstract: Two filters for improving the visibility of crystalline material in the presence of amorphous surface contamination layers in high-resolution electron microscope images can be constructed automatically from the information present in the Fourier transform of the recorded image The recorded signal is considered in the first approximation to be the sum of two signals which are uncorrelated in the frequency domain By estimating the power spectrum of the signal from the amorphous layer, an optimized estimate for the desired signal is given by the Wiener filter A second filter which uses the estimated amplitude of the amorphous signal to subtract out a background can be shown to be related to the Wiener filter The two filters are applied to an experimental image of zeolite and the effects of the two filters are compared

235 citations


PatentDOI
TL;DR: In this paper, a filterbank structure is provided which provides a flexible compromise between the conflicting goals of processing delay, filter sharpness, memory usage and band interaction, which is advantageous for hearing loss fitting, especially at low frequencies.
Abstract: A filterbank structure is provided which provides a flexible compromise between the conflicting goals of processing delay, filter sharpness, memory usage and band interaction. The filterbank has an adjustable number of bands and a stacking which provides for a selectable shift of band frequencies to one of two discrete sets of center frequencies. The width of the bands and hence the number of the bands is selected depending upon acceptable delay, memory usage, and processing speed required. The flexibility in terms of stacking of the bands provides twice the number of potential band edge placements, which is advantageous for hearing loss fitting, especially at low frequencies. The same filter coefficients can be used for analysis and synthesis, to reduce memory usage.

223 citations


Journal ArticleDOI
TL;DR: This correspondence introduces a new class of infinite impulse response (IIR) digital filters that unifies the classical digital Butterworth filter and the well-known maximally flat FIR filter.
Abstract: This correspondence introduces a new class of infinite impulse response (IIR) digital filters that unifies the classical digital Butterworth filter and the well-known maximally flat FIR filter. New closed-form expressions are provided, and a straightforward design technique is described. The new IIR digital filters have more zeros than poles (away from the origin), and their (monotonic) square magnitude frequency responses are maximally flat at /spl omega/=0 and at /spl omega/=/spl pi/. Another result of the correspondence is that for a specified cutoff frequency and a specified number of zeros, there is only one valid way in which to split the zeros between z=-1 and the passband. This technique also permits continuous variation of the cutoff frequency. IIR filters having more zeros than poles are of interest because often, to obtain a good tradeoff between performance and implementation complexity, just a few poles are best.

212 citations


Proceedings ArticleDOI
16 Aug 1998
TL;DR: A strategy to design recursive implementations of the Gaussian filter and Gaussian regularized derivative filters that yield a high accuracy and excellent isotropy in n-D space is proposed.
Abstract: We propose a strategy to design recursive implementations of the Gaussian filter and Gaussian regularized derivative filters. Each recursive filter consists of a cascade of two stable Nth-order subsystems (causal and anti-causal). The computational complexity is 2N multiplications per pixel per dimension independent of the size (/spl sigma/) of the Gaussian kernel. The filter coefficients have a closed-form solution as a function of scale (/spl sigma/) and recursion order N (N=3, 4, 5). The recursive filters yield a high accuracy and excellent isotropy in n-D space.

192 citations


Journal ArticleDOI
TL;DR: The new weighted median filter formulation leads to significantly more powerful estimators capable of effectively addressing a number of fundamental problems in signal processing that could not adequately be addressed by prior weighted median smoother structures.
Abstract: Weighted median smoothers, which were introduced by Edgemore in the context of least absolute regression over 100 years ago, have received considerable attention in signal processing during the past two decades. Although weighted median smoothers offer advantages over traditional linear finite impulse response (FIR) filters, it is shown in this paper that they lack the flexibility to adequately address a number of signal processing problems. In fact, weighted median smoothers are analogous to normalized FIR linear filters constrained to have only positive weights. It is also shown that much like the mean is generalized to the rich class of linear FIR filters, the median can be generalized to a richer class of filters admitting positive and negative weights. The generalization follows naturally and is surprisingly simple. In order to analyze and design this class of filters, a new threshold decomposition theory admitting real-valued input signals is developed. The new threshold decomposition framework is then used to develop fast adaptive algorithms to optimally design the real-valued filter coefficients. The new weighted median filter formulation leads to significantly more powerful estimators capable of effectively addressing a number of fundamental problems in signal processing that could not adequately be addressed by prior weighted median smoother structures.

183 citations


Patent
09 Jul 1998
TL;DR: In this paper, an optical filter compresses data into and/or derives data from a light signal, and the filter way weight an incident light signal by wavelength over a predetermined wavelength range according to a predetermined function so that the filter performs the dot product of the light signal and the function.
Abstract: In optical filter systems and optical transmission systems, an optical filter compresses data into and/or derives data from a light signal. The filter way weight an incident light signal by wavelength over a predetermined wavelength range according to a predetermined function so that the filter performs the dot product of the light signal and the function.

179 citations


Journal ArticleDOI
TL;DR: This paper claims that the asymptotic game filter is itself a detection filter, and demonstrates the effectiveness of the filter for time-invariant and time-varying problems in both full-order and reduced-order forms.
Abstract: The fault detection process is approximated with a disturbance attenuation problem. The solution to this problem, for both linear time-varying and time-invariant systems, leads to a game theoretic filter which bounds the transmission of all exogenous signals except the fault to be detected. In the limit, when the disturbance attenuation bound is brought to zero, a complete transmission block is achieved by embedding the nuisance inputs into an unobservable, invariant subspace. Since this is the same invariant subspace structure seen in some types of detection filters, we can claim that the asymptotic game filter is itself a detection filter. One can also make use of this subspace structure to reduce the order of the limiting game theoretic filter by factoring this invariant subspace out of the state space. The resulting lower dimensional filter will then be sensitive only to the failure to be detected. A pair of examples given at the end of the paper demonstrate the effectiveness of the filter for time-invariant and time-varying problems in both full-order and reduced-order forms.

175 citations


Journal ArticleDOI
TL;DR: In this paper, the group delay of the input reflection coefficients of sequentially tuned resonators has been shown to provide all the information necessary to design and tune filters, and that the group-delay value at the center frequency of the filter can be written quite simply in terms of the low pass prototype values, the LC elements of a bandpass structure, and the coupling coefficients of the inverter coupled filter.
Abstract: The concept of coupling coefficients has been a very useful one in the design of small-to-moderate bandwidth microwave filters. It is shown in this paper that the group delay of the input reflection coefficients of sequentially tuned resonators contains all the information necessary to design and tune filters, and that the group-delay value at the center frequency of the filter can be written quite simply in terms of the low-pass prototype values, the LC elements of a bandpass structure, and the coupling coefficients of the inverter coupled filter. This provides an easy method to measure the key elements of a filter, which is confirmed by results presented in this paper. It is also suggested that since the group delay of the reflection coefficient (i.e., the time taken for energy to get in and out of the coupled resonators) is easily measured, it is a useful conceptual alternative to coupling concepts.

Journal ArticleDOI
TL;DR: An iterative and distributed power control algorithm which iteratively updates the transmitter powers and receiver filter coefficients of the users and converges to a minimum power solution for the powers, and an MMSE multiuser detector for the filter coefficients is proposed.
Abstract: Power control algorithms assume that the receiver structure is fixed and iteratively update the transmit powers of the users to provide acceptable quality of service while minimizing the total transmitter power. Multiuser detection, on the other hand, optimizes the receiver structure with the assumption that the users have fixed transmitter powers. In this study, we combine the two approaches and propose an iterative and distributed power control algorithm which iteratively updates the transmitter powers and receiver filter coefficients of the users. We show that the algorithm converges to a minimum power solution for the powers, and an MMSE multiuser detector for the filter coefficients.

Proceedings ArticleDOI
17 May 1998
TL;DR: In this article, the authors proposed a hybrid active filter for the damping of harmonic resonance in industrial power systems, which consists of a small-rated active filter and a 5th-tuned passive filter.
Abstract: This paper proposes a hybrid active filter for the damping of harmonic resonance in industrial power systems. The hybrid filter consists of a small-rated active filter and a 5th-tuned passive filter. The active filter is characterized by detecting the 5th-harmonic current flowing into the passive filter. It is controlled in such a way as to behave as a negative or positive resistor by adjusting a feedback gain from a negative to positive value, and vice versa. The negative resistor presented by the active filter cancels a positive resistor inherent in the passive filter, so that the hybrid filter acts as an ideal passive filter with infinite quality factor. This significantly improves damping the harmonic resonance, compared with the passive filter used alone. Moreover, the active filter acts as a positive resistor to prevent an excessive harmonic current from flowing into the passive filter. Experimental results obtained from a 20-kW laboratory model verify the viability and effectiveness of the hybrid active filter proposed in this paper.

Journal ArticleDOI
TL;DR: In this paper, a self-organizing filter and smoother for the general nonlinear non-Gaussian state-space model is proposed, which is defined by augmenting the state vector with the unknown parameters of the original state space model.
Abstract: A self-organizing filter and smoother for the general nonlinear non-Gaussian state-space model is proposed. An expanded state-space model is defined by augmenting the state vector with the unknown parameters of the original state-space model. The state of the augmented state-space model, and hence the state and the parameters of the original state-space model, are estimated simultaneously by either a non-Gaussian filter/smoother or a Monte Carlo filter/smoother. In contrast to maximum likelihood estimation of model parameters in ordinary state-space modeling, for which the recursive filter computation has to be done many times, model parameter estimation in the proposed self-organizing filter/smoother is achieved with only two passes of the recursive filter and smoother operations. Examples such as automatic tuning of dispersion and the shape parameters, adaptation to changes of the amplitude of a signal in seismic data, state estimation for a nonlinear state space model with unknown parameters, ...

Patent
07 Nov 1998
TL;DR: In this paper, a cascade of two filters (114, 118) along with a short bulk delay (110) is used to model the feedback path of a hearing aid, and the second filter does not use a separate probe signal.
Abstract: Feedback cancellation apparatus uses a cascade of two filters (114, 118) along with a short bulk delay (110). The first filter (114) is adapted when the hearing aid is turned on in the ear. This filter adapts quickly using a white noise probe signal (216), and then the filter coefficients are frozen. The first filter models parts of the hearing-aid feedback path that are essentially constant over the course of the day. The second filter (118) adapts while the hearing aid is in use and does not use a separate probe signal. This filter provides a rapid correction to the feedback path model when the hearing aid goes unstable, and more slowly tracks perturbations in the feedback path that occur in daily use. The delay (110) shifts the filter response to make the most effective use of the limited number of filter coefficients.

01 Jan 1998
TL;DR: This work deals with digital waveguide modeling of acoustic tubes, such as bores of musical woodwind instruments or the human vocal tract, and a novel discrete-time signal processing technique, deinterpolation, is defined.
Abstract: This work deals with digital waveguide modeling of acoustic tubes, such as bores of musical woodwind instruments or the human vocal tract. The acoustic tube systems considered in this work are those consisting of a straight cylindrical or conical tube section or of a concatenation of several cylindrical or conical tube sections. Also, the junction of three tube sections is studied. Of special interest for our application are junctions where a side branch is connected to a cylindrical or conical tube since these are needed in the simulation of woodwind instrument bores. Basic waveguide models are generalized by employing the concept of fractional delay, which means a fraction of the unit sample interval. A fractional delay is implemented using bandlimited interpolation. A novel discrete-time signal processing technique, deinterpolation, is defined. Applying fractional delay filtering techniques, a spatially discretized waveguide model is turned into a spatially continuous one. This implies that the length of the digital waveguide can be adjusted as accurately as required, and a change of the impedance of a waveguide may occur at any desired point between sampling points. This kind of a system is called a fractional delay waveguide filter (FDWF). It is a discrete-time structure but yet a spatially continuous model for a physical system. The basic principles of digital waveguide modeling are first reviewed. Modeling techniques for cylindrical and conical acoustic tubes are described, as well as methods to simulate junctions of two or more of these sections. Different design methods for both FIR and IIR (allpass) fractional delay filters are reviewed and the theoretical foundations of FDWFs are studied. Fractional delay extensions for acoustic tube model structures are discussed and approximation errors due to fractional delay filters are analyzed. In addition, a new technique for eliminating transients due to time-varying filter coefficients in recursive filters is introduced. The models described in this work are directly applicable to physical modeling and model-based sound synthesis of speech and wind instruments.

Book ChapterDOI
23 Sep 1998
TL;DR: Initial results of the system as applied to two analog filter design problems suggest that the ability to evolve complex analog circuit representations in software is becoming more approachable on a single engineering workstation.
Abstract: We present a method of evolving analog electronic circuits using a linear representation and a simple unfolding technique. While this representation excludes a large number of circuit topologies, it is capable of constructing many of the useful topologies seen in hand-designed circuits. Our system allows circuit size, circuit topology, and device values to be evolved. Using a parallel genetic algorithm we present initial results of our system as applied to two analog filter design problems. The modest computational requirements of our system suggest that the ability to evolve complex analog circuit representations in software is becoming more approachable on a single engineering workstation.

Proceedings ArticleDOI
24 Jun 1998
TL;DR: A unified theory of tomosynthesis is derived in the context of linear system theory, and a general four-step filter design concept is presented, which is valid for any specific scan geometry.
Abstract: Tomosynthesis provides only incomplete 3D-data of the imaged object. Therefore it is important for reconstruction tasks to take all available information carefully into account. We are focusing on geometrical aspects of the scan process which can be incorporated into reconstruction algorithms by filtered backprojection methods. Our goal is a systematic approach to filter design. A unified theory of tomosynthesis is derived in the context of linear system theory, and a general four-step filter design concept is presented. Since the effects of filtering are understandable in this context, a methodical formulation of filter functions is possible in order to optimize image quality regarding the specific requirements of any application. By variation of filter parameters the slice thickness and the spatial resolution can easily be adjusted. The proposed general concept of filter design is exemplarily discussed for circular scanning but is valid for any specific scan geometry. The inherent limitations of tomosynthesis are pointed out and strategies for reducing the effects of incomplete sampling are developed. Results of a dental application show a striking improvement in image quality.

Journal ArticleDOI
TL;DR: A convenient exponential family is proposed which allows one to simplify the projection filter equation and to define an a posteriori measure of the local error of the projections filter approximation.
Abstract: This paper presents a new and systematic method of approximating exact nonlinear filters with finite dimensional filters, using the differential geometric approach to statistics. The projection filter is defined rigorously in the case of exponential families. A convenient exponential family is proposed which allows one to simplify the projection filter equation and to define an a posteriori measure of the local error of the projection filter approximation. Finally, simulation results are discussed for the cubic sensor problem.

Journal ArticleDOI
TL;DR: This algorithm involves a very simple update term that is computationally comparable to the update in the classical LMS algorithm and is demonstrated through a computer simulation example involving lowpass filtering of a one-dimensional chirp-type signal in impulsive noise.
Abstract: Stochastic gradient-based adaptive algorithms are developed for the optimization of weighted myriad filters (WMyFs). WMyFs form a class of nonlinear filters, motivated by the properties of /spl alpha/-stable distributions, that have been proposed for robust non-Gaussian signal processing in impulsive noise environments. The weighted myriad for an N-long data window is described by a set of nonnegative weights {w/sub i/}/sub i=l//sup N/ and the so-called linearity parameter K>0. In the limit, as K/spl rarr//spl infin/, the filter reduces to the familiar weighted mean filter (which is a constrained linear FIR filter). Necessary conditions are obtained for optimality of the filter weights under the mean absolute error criterion. An implicit formulation of the filter output is used to find an expression for the gradient of the cost function. Using instantaneous gradient estimates, an adaptive steepest-descent algorithm is then derived to optimize the weights. This algorithm involves a very simple update term that is computationally comparable to the update in the classical LMS algorithm. The robust performance of this adaptive algorithm is demonstrated through a computer simulation example involving lowpass filtering of a one-dimensional chirp-type signal in impulsive noise.

Journal ArticleDOI
K. Martin1
TL;DR: In this article, a weighted sum of near-adjacent IFT filters is used to realize the individual channel-bank filters, with constraints added that results in significantly improved stopband performance while still achieving small reconstruction errors.
Abstract: An approach for realizing filter banks having improved side-lobe performance compared to approaches such as those based on inverse Fourier transforms (IFTs), especially for greater frequency differences from the passband frequencies, is presented. The approach is based on using a weighted-sum of near-adjacent IFT filters to realize the individual channel-bank filters, but with constraints added that results in significantly improved stopband performance while still achieving small reconstruction errors. The proposed channel banks are suitable for realizing multitone digital data communication systems, such as Asymmetric Digital Subscriber Line (ADSL) systems, where stopband performance is critical. Under the conditions of maximal decimation, the reconstruction is not perfect, but aliasing errors are small enough to be negligible in practical communication systems. For some cases, the filter coefficients can be determined exactly without using optimization. Given the frequency-weighting coefficients reported herein, near-optimal multirate filter banks may be designed exactly without optimization for all even n.

Journal ArticleDOI
TL;DR: In this paper, a direct cascading of a wide band combine filter to a TE01 mode dielectric resonator (DR) filter is proposed to suppress the spurious response of the DR cavity filter.
Abstract: This paper presents the state of the art of high-Q TE01 mode DR cavity filters for PCS wireless base station applications. In order to have TE01 mode filter to be competitive with other high-Q cavity technologies, employment of nonadjacent coupling to implement advanced filter features and easy filter machining and integration are essential. The quadruplet and trisections are regarded as basic blocks to implement symmetric and asymmetric transmission zeros in filter stop band. The relative alignment of the magnetic mode field across the coupled adjacent cavities is analyzed to identify the sign of nonadjacent coupling. A direct cascading of a wide band combine filter to a TE01 mode dielectric resonator (DR) filter is proposed to suppress the spurious response of the DR cavity filter. This approach simplifies the integration between the DR filter and the spurious suppression device and has been proved to be very cost effective. Experimental eight- and six-pole quasi-elliptic function filters show the typical performances. To take advantage of the special property of magnetic mode field alignment across the adjacent cavities, a five-pole canonical asymmetric filter with three transmission zeros in low side is implemented. We believe this filter is a new design for high-Q cavity filter, while a three-pole elliptic function filter is new for DR filter technology.

Proceedings ArticleDOI
01 Oct 1998
TL;DR: The authors present a methodology for designing filters based on spatial smoothness and accuracy criteria and use the filters so designed for volume rendering of sampled data sets and a synthetic test function.
Abstract: The correct choice of function and derivative reconstruction filters is paramount to obtaining highly accurate renderings. Most filter choices are limited to a set of commonly used functions, and the visualization practitioner has so far no way to state his preferences in a convenient fashion. Much work has been done towards the design and specification of filters using frequency based methods. However for visualization algorithms it is more natural to specify a filter in terms of the smoothness of the resulting reconstructed function and the spatial reconstruction error. Hence, the authors present a methodology for designing filters based on spatial smoothness and accuracy criteria. They first state their design criteria and then provide an example of a filter design exercise. They also use the filters so designed for volume rendering of sampled data sets and a synthetic test function. They demonstrate that their results compare favorably with existing methods.

Journal ArticleDOI
TL;DR: The development of a filter bank structure which combines the flexibility of the short-time Fourier transform (STFT) with the implementation efficiency of the polyphase filter bank decomposition, meeting these requirements and leading to a hardware-efficient implementation, is presented.
Abstract: An approach is presented to realizing a digital channelized receiver for signal intercept applications that provides a hardware efficient implementation of a uniform filter bank in which the number of filters K is greater than the decimation factor M. The proposed architecture allows simple channel arbitration logic to be used and provides reliable instantaneous frequency measurements, even in adjacent channel crossover regions. In the proposed implementation of the filter bank, K is related to M by K=FM where F is an integer. It is shown that the optimum selection of F allows the instantaneous frequency measurement to be made in the channel crossover region and the arbitration function to be based solely on the instantaneous frequency measurement. The development of a filter bank structure which combines the flexibility of the short-time Fourier transform (STFT) with the implementation efficiency of the polyphase filter bank decomposition, meeting these requirements and leading to a hardware-efficient implementation, is presented.

Patent
09 Dec 1998
TL;DR: In this article, a scalable FIR filter architecture that requires fewer computations, less storage registers, and is capable of parallel processing, is presented, which reduces the number of computations (e.g., multiplication) by utilizing the inherent symmetry.
Abstract: A scalable FIR filter architecture that requires fewer computations, less storage registers, and is capable of parallel processing, is presented. The scalable filter architecture reduces the number of computations (e.g., multiplication) by utilizing the inherent symmetry and reduces the number of storage elements required by utilizing what is known as the transpose-form (as compared to direct-form) filter architecture. The filter architecture is scalable to accommodate different complexity levels. In accordance to the present invention, a filter can be scaled up/down by adding/subtracting a processing block to/from the existing structure. Because these processing blocks can process signals independently and simultaneously, the filter architecture in accordance to the present invention allows for parallel and distributive processing thereby meeting the required performance requirements.

Journal ArticleDOI
TL;DR: This work gives an analytical solution for the compaction filter that is characterized by its zeros on the unit circle that corresponds to the optimal two-channel FIR filter bank that maximizes the coding gain under the traditional quantization noise assumptions.
Abstract: The problem of optimum FIR energy compaction filter design for a given number of channels M and a filter order N is considered. The special cases where N

Journal ArticleDOI
TL;DR: It is shown that the reduced-rank output signal computed via truncated (Q)SVD is identical to that from an array of parallelly connected analysis-synthesis finite impulse response (FIR) filter pairs.
Abstract: We show that the reduced-rank output signal computed via truncated (Q)SVD is identical to that from an array of parallelly connected analysis-synthesis finite impulse response (FIR) filter pairs. The filter coefficients are determined by the (Q)SVD, and the filters provide an explicit description of the reduced-rank noise reduction algorithm in the frequency domain.

Proceedings ArticleDOI
06 Oct 1998
TL;DR: It is shown that hard thresholding is typically outperformed by a Wiener filter designed in an alternate wavelet domain, and a method is provided for selecting the various parameters involved in a wavelet-domain Wiener filtering scheme.
Abstract: We investigate Wiener filtering of wavelet coefficients for signal denoising Empirically designed wavelet-domain Wiener filters outperform many other denoising algorithms based on wavelet thresholding However, up to now, it has not been clear how to choose the signal model used to design the filter, because the effect of model selection on the filter performance is difficult to understand By analyzing the error involved in the Wiener filter designed with an empirically obtained signal model, we show that hard thresholding is typically outperformed by a Wiener filter designed in an alternate wavelet domain Our analysis furthermore provides a method for selecting the various parameters involved in a wavelet-domain Wiener filtering scheme

Journal ArticleDOI
TL;DR: This work studies the design of signal-adapted FIR paraunitary filter banks, using energy compaction as the adaptation criterion, and shows how regularity constraints may be incorporated into the design problem to obtain globally optimal filter banks with specified regularity.
Abstract: We study the design of signal-adapted FIR paraunitary filter banks, using energy compaction as the adaptation criterion. We present some important properties that globally optimal solutions to this optimization problem satisfy. In particular, we show that the optimal filters in the first channel of the filter bank are spectral factors of the solution to a linear semi-infinite programming (SIP) problem. The remaining filters are related to the first through a matrix eigenvector decomposition. We discuss uniqueness and sensitivity issues. The SIP problem is solved using a discretization method and a standard simplex algorithm. We also show how regularity constraints may be incorporated into the design problem to obtain globally optimal (in the energy compaction sense) filter banks with specified regularity. We also consider a problem in which the polyphase matrix implementation of the filter bank is constrained to be DCT based. Such constraints may also be incorporated into our optimization algorithm; therefore, we are able to obtain globally optimal filter banks subject to regularity and/or computational complexity constraints. Numerous experiments are presented to illustrate the main features that distinguish adapted and nonadapted filters, as well as the effects of the various constraints. The conjecture that energy compaction and coding gain optimization are equivalent design criteria is shown not to hold for FIR filter banks.

Journal ArticleDOI
TL;DR: This paper discusses the design and properties of two trajectory tracking controllers for linear time-invariant systems, and compares their implementation and experimental results on a flexible one-link robot equipped with a velocity-controlled actuator.
Abstract: This paper discusses the design and properties of two trajectory tracking controllers for linear time-invariant systems, and compares their implementation and experimental results on a flexible one-link robot equipped with a velocity-controlled actuator. High positioning accuracy and low tracking errors within a specified bandwidth are their performance specifications. Both controllers use the same state feedback controller, but have a different feedforward design approach. Both feedforward methods design stable prefilters which approximate the unstable inverse system model. The first method designs a stable prefilter using the extended bandwidth zero phase error tracking control (EBZPETC) method. The second feedforward method adds delay to the inverse model and then uses common filter design techniques to approximate this delayed frequency response.