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Showing papers on "Filter design published in 2000"


Proceedings Article
01 Jan 2000
TL;DR: This paper proposes a new particle filter based on sequential importance sampling that outperforms standard particle filtering and other nonlinear filtering methods very substantially and is in agreement with the theoretical convergence proof for the algorithm.
Abstract: In this paper, we propose a new particle filter based on sequential importance sampling. The algorithm uses a bank of unscented filters to obtain the importance proposal distribution. This proposal has two very "nice" properties. Firstly, it makes efficient use of the latest available information and, secondly, it can have heavy tails. As a result, we find that the algorithm outperforms standard particle filtering and other nonlinear filtering methods very substantially. This experimental finding is in agreement with the theoretical convergence proof for the algorithm. The algorithm also includes resampling and (possibly) Markov chain Monte Carlo (MCMC) steps.

1,681 citations


Book
12 Dec 2000
TL;DR: In this paper, the authors proposed a direct-coupled Resonator BPF (BPF) filter design by experimental method and determined the Coupling Coefficient of BPF.
Abstract: 1. Introduction.- 2. Basic Structure and Characteristics of SIR.- 3. Quarter-Wavelength-Type SIR.- 4. Half-Wavelength-Type SIR.- 5. One-Wavelength-Type SIR.- 6. Expanded Concept and Technological Trends in SIR.- Appendix. Analysis of Resonator Properties Using General-Purpose Microwave Simulator.- A.1 Design Parameters of Direct-Coupled Resonator BPF.- A.2 Filter Design by Experimental Method.- A.3.2 Determination of Coupling Coefficient.- References.

275 citations


Journal ArticleDOI
TL;DR: In this article, a new resonator-embedded cross-coupled filter, constructed by stepped-impedance hairpin resonators and miniaturized hairpin resonance, is presented.
Abstract: Stepped-impedance resonators have been thoroughly studied in this paper. Two equations for odd- and even-mode resonance are derived from a new network model. The size and resonant frequencies of the resonator could then be designed based on these two equations. A new resonator-embedded cross-coupled filter, constructed by stepped-impedance hairpin resonators and miniaturized hairpin resonators is presented. This new filter is very compact and has lower spurious response. A 0/spl deg/ feed structure, which adds two transmission zeros to the filter response, is also studied. The two zeros are so close to the passband that the selectivity and out-of-band rejection of the filter are significantly increased. The design has been verified by experiment results.

272 citations


Patent
13 Jul 2000
TL;DR: In this article, a digital compensation signal processing component (DCSP) is used to compensate for the frequency and time dependent AM-AM and AM-PM distortion characteristics of a nonlinear amplifier.
Abstract: A predistortion system comprises a digital compensation signal processing component (DCSP) (52) which predistorts a wideband input transmission signal to compensate for the frequency and time dependent AM-AM and AM-PM distortion characteristics of a non-linear amplifier (64) The DCSP (52) comprises a data structure (52H) in which each element stores a set of compensation parameters (preferably including FIR filter coefficients) for predistorting the input transmission signal The parameter sets are preferably indexed within the data structure (52H) according to multiple signal characteristics, such as instantaneous amplitude and integrated signal envelope, each of which corresponds to a respective dimension of the data structure (52H) The sets of compensation parameters are generated periodically and written to the data structure (52H) by an adaptive control processing and compensation estimator (ACPCE) (70) that performs a non-real-time analysis of amplifier input and output signals The ACPCE (70) also implements various system identification processes for measuring the characteristics of the power amplifier (64) and generating initial sets of filter coefficients

209 citations


Patent
13 Jul 2000
TL;DR: In this article, a digital compensation signal processing component (DCSP) is used to pre-dictate a wideband input transmission signal to compensate for the frequency and time dependent AM-AM and AM-PM distortion characteristics of a non-linear amplifier.
Abstract: A predistortion system comprises a digital compensation signal processing component (DCSP) (52) which predistorts a wideband input transmission signal to compensate for the frequency and time dependent AM-AM and AM-PM distortion characteristics of a non-linear amplifier (64). The DCSP (52) comprises a data structure (52H) in which each element stores a set of compensation parameters (preferably including FIR filter coefficients) for predistorting the input transmission signal. The parameter sets are preferably indexed within the data structure (52H) according to multiple signal characteristics, such as instantaneous amplitude and integrated signal envelope, each of which corresponds to a respective dimension of the data structure (52H). To predistort the input transmission signal, an addressing circuit (52C-52G) digitally generates a set of data structure indices from the input transmission signal, and the indexed set of compensation parameters is loaded into a compensation circuit (52A, 52B) which digitally predistorts the input transmission signal. This process of loading new compensation parameters into the compensation circuit (52A, 52B) is preferably repeated every sample instant, so that the predistortion function varies from sample-to-sample. The sets of compensation parameters are generated periodically and written to the data structure (52H) by an adaptive control processing and compensation estimator (ACPCE) (70) taht performs a non-real-time analysis of amplifier input and output signals. The ACPCE (70) also implements various system identification processes for measuring the characteristics of the power amplifier (64) and generating initial sets of filter coefficients. In an antenna array embodiment (Figures 33 and 34), a single ACPCE (70) generates the compensation parameters sets for each of multiple amplification chains (64) on a time-shared basis. In an embodiment (Figure 32) in which the amplification chain (64) includes multiple nonlinear amplifiers (60A) that can be individually controlled (e.g., turned ON and OFF) to conserve power, the data structure (52H) separately stores compensation parameter sets for each operating state of the amplification chain (64).

205 citations


Journal ArticleDOI
TL;DR: This paper discusses use of the more general coefficient of determination in nonlinear filtering, and addresses the VC dimension of increasing operators in terms of their morphological kernel/basis representations.

190 citations


Journal ArticleDOI
01 Jun 2000
TL;DR: The problem of robust energy-to-peak filtering for linear systems with convex bounded uncertainties is investigated and a full order stable linear filter is designed that minimizes the worst-case peak value of the filtering error output signal with respect to all bounded energy inputs.
Abstract: The problem of robust energy-to-peak filtering for linear systems with convex bounded uncertainties is investigated in this paper. The main purpose is to design a full order stable linear filter that minimizes the worst-case peak value of the filtering error output signal with respect to all bounded energy inputs, in such a way that the filtering error system remains quadratically stable. Necessary and sufficient conditions are formulated in terms of linear Matrix Inequalities - LMIs, for both continuous- and discrete-time cases.

174 citations


Book
01 Sep 2000
TL;DR: This book shows readers how to design many types of filters that cannot be designed using conventional techniques in basic analog and digital IIR filter design--regardless of the technology.
Abstract: From the Publisher: A complete up-to-date reference for advanced analog and digital IIR filter design rooted in elliptic functions. Revolutionary in approach, this book opens up completely new vistas in basic analog and digital IIR filter design--regardless of the technology. By introducing exceptionally elegant and creative mathematical stratagems (e.g., accurate replacement of Jacobi elliptic functions by functions comprising polynomials, square roots, and logarithms), optimization routines carried out with symbolic analysis by Mathematica, and the advance filter design software of MATLAB, it shows readers how to design many types of filters that cannot be designed using conventional techniques. The filter design algorithms can be directly programed in any language or environment such as Visual BASIC, Visual C, Maple, DERIVE, or MathCAD. Signals; Systems; Transforms; Classical Analog Filter Design; Advanced Analog Filter Design Case Studies; Advanced Analog Filter Design Algorithms; Multi-criteria Optimization of Analog Filter Designs; Classical Digital Filter Design; Advanced Digital Filter Design Case Studies; Advanced Digital Filter Design Algorithms; Multi-criteria Optimization of Digital Filter Designs; Elliptic Functions; Elliptic Rational Function.

166 citations


Journal ArticleDOI
TL;DR: The results of this multicenter registry support the need for innovative filter design, as well as a randomized, prospective study, to evaluate the current practice of temporary vena cava filter placement and its complications.

163 citations


Journal ArticleDOI
TL;DR: In this article, a method to design and implement a very efficient multistage decimation filter for a sigma-delta A/D converter is proposed, which can be easily extended to a multiple-stage implementation.
Abstract: In this paper, a method to design and implement a very efficient multistage decimation filter for a sigma-delta (/spl Sigma//spl Delta/) A/D converter is proposed The scheme is composed of two stages, but the proposed method can be easily extended to a multiple-stage implementation The first-stage filter is obtained by properly rotating the zero-pole distribution of a comb filter in the z-plane The obtained structure exhibits linear phase and can be implemented by using a recursive structure with only two multipliers The design phase is easy and very flexible As most of the quantization noise is eliminated at the first stage, the second-stage filter can be designed with relaxed specifications Any classical design algorithm can be used for it An alternative scheme for the second stage can be obtained by splitting the stage into two substages, and the method proposed in this paper can be iterated

148 citations


Patent
19 Jun 2000
TL;DR: In this paper, a digital compensation signal processing component (DCSP) is used to pre-dictate a wideband input transmission signal to compensate for the frequency and time dependent AM-AM and AM-PM distortion characteristics of a nonlinear amplifier.
Abstract: A predistortion system comprises a digital compensation signal processing component (DCSP) (52) which predistorts a wideband input transmission signal to compensate for the frequency and time dependent AM-AM and AM-PM distortion characteristics of a non-linear amplifier (64). The DCSP (52) comprises a data structure (52H) in which each element stores a set of compensation parameters (preferably including FIR filter coefficients) for predistorting the input transmission signal. The parameter sets are preferably indexed within the data structure (52H) according to multiple signal characteristics, such as instantaneous amplitude and integrated signal envelope, each of which corresponds to a respective dimension of the data structure (52H). To predistort the input transmission signal, an addressing circuit (52C-52G) digitally generates a set of data structure indices from the input transmission signal, and the indexed set of compensation parameters is loaded into a compensation circuit (52A, 52B) which digitally predistorts the input transmission signal. This process of loading new compensation parameters into the compensation circuit (52A, 52B) is preferably repeated every sample instant, so that the predistortion function varies from sample-to-sample. The sets of compensation parameters are generated periodically and written to the data structure (52H) by an adaptive control processing and compensation estimator (ACPCE) (70) that performs a non-real-time analysis of amplifier input and output signals. The ACPCE (70) also implements various system identification processes for measuring the characteristics of the power amplifier (64) and generating initial sets of filter coefficients. In an antenna array embodiment (Figures 33 and 34), a single ACPCE (70) generates the compensation parameters sets for each of multiple amplification chains (64) on a time-shared basis. In an embodiment (Figure 32) in which the amplification chain (64) includes multiple nonlinear amplifiers (60A) that can be individually controlled (e.g., turned ON and OFF) to conserve power, the data structure (52H) separately stores compensation parameter sets for each operating state of the amplification chain (64).

Journal ArticleDOI
TL;DR: In this paper, the authors present design considerations for programmable high-frequency continuous-time filters implemented in standard digital CMOS processes, where accumulation MOS capacitors are used as integrating elements to reduce area, and a constant-capacitance scaling technique is employed to ensure that even parasitic capacitances remain invariant when transconductors are switched in and out of the filter.
Abstract: We present design considerations for programmable high-frequency continuous-time filters implemented in standard digital CMOS processes. To reduce area, accumulation MOS capacitors are used as integrating elements. The filter design problem is examined from the viewpoint of programmability. To allow frequency scalability without deterioration of noise performance and of the frequency response shape, we employ a technique called "constant-capacitance scaling," which assures that even parasitic capacitances remain invariant when transconductors are switched in and out of the filter. This technique is applied to the design of a programmable fourth order Butterworth continuous-time filter with a bandwidth programmable from 60 to 350 MHz implemented in a 0.25-/spl mu/m digital CMOS process. The filter has a dynamic range of 54 dB, dissipates 70 mW from a 3.3-V supply, and occupies an area of 0.15 mm/sup 2/.

Journal ArticleDOI
John Platt1
TL;DR: In this article, an error metric inspired by psychophysical experiments is used to reduce the number of pixels to be set in a high-resolution input image, and a linear system of equations can be expressed as a set of filters.
Abstract: Displays with repeating patterns of colored subpixels gain spatial resolution by setting individual subpixels rather than by setting entire pixels. This paper describes optimal filtering that produces subpixel values from a high-resolution input image. The optimal filtering is based on an error metric inspired by psychophysical experiments. Minimizing the error metric yields a linear system of equations, which can be expressed as a set of filters. These filters provide the same quality of font display as standard anti-aliasing at a point size 25% smaller. This optimization forms the filter design framework for Microsoft's ClearType.

Journal ArticleDOI
01 Dec 2000
TL;DR: A new approach to circuit synthesis based on genetic algorithms is presented that produces design solutions that are more efficient than those resulting from formal design methods or created manually by an experienced analogue circuit designer.
Abstract: Most analogue systems are designed manually because automatic circuit synthesis tools are available for only a limited range of design problems. A new approach to circuit synthesis based on genetic algorithms is presented. Using this method it is possible in principle to synthesise circuits to meet any linear or nonlinear, frequency-domain or time-domain, specification. When applied to existing filter design problems this circuit synthesis method produces design solutions that are more efficient than those resulting from formal design methods or created manually by an experienced analogue circuit designer.

Patent
30 Nov 2000
TL;DR: In this paper, a disk drive using an accelerometer to sense linear vibration and cancel its effects with an adaptive algorithm during track following is described, where the accelerometer is oriented to detect acceleration associated with torque that tends to cause the actuator to move off-track notwithstanding the efforts of the servo control system.
Abstract: Disclosed is a disk drive using an accelerometer to sense linear vibration and cancel its effects with an adaptive algorithm during track following. The accelerometer is oriented to detect acceleration associated with torque that tends to cause the actuator to move off-track notwithstanding the efforts of the servo control system. The accelerometer's filtered output is used to modify the control effort. The disk drive uses the position error signal to adaptively filter the accelerometer's output in an effort to mathematically converge on a set of optimal filter coefficients and thereby reduce the effect of vibration that may otherwise impose a torque on the actuator.

Journal ArticleDOI
Christi K. Madsen1
TL;DR: In this article, a general design algorithm for infinite impulse response (IIR) bandpass and arbitrary magnitude response filters that use optical all-pass filters as building blocks is presented, and a reduced set of unique operating states is discussed for implementing a reconfigurable multichannel selection filter.
Abstract: A general design algorithm is presented for infinite impulse response (IIR) bandpass and arbitrary magnitude response filters that use optical all-pass filters as building blocks. Examples are given for an IIR multichannel frequency selector, an amplifier gain equalizer, a linear square-magnitude response, and a multi-level response. Major advantages are the efficiency of the IIR filter compared to finite impulse response (FIR) filters, the simplicity of the optical architecture, and its tolerance for loss. A reduced set of unique operating states is discussed for implementing a reconfigurable multichannel selection filter.

Proceedings ArticleDOI
11 Oct 2000
TL;DR: An example filter design is presented that shows the error involved in limiting the number of allowable non-zero CSD coefficients for a real FIR bandpass filter.
Abstract: It is shown that the use of a canonical signed digit (CSD) representation of the filter coefficients can significantly reduce the complexity of the hardware implementation of digital FIR filters. This paper presents an example filter design that shows the error involved in limiting the number of allowable non-zero CSD coefficients for a real FIR bandpass filter. If not done carefully, brute force limiting can lead to large errors in the frequency response. The error is evaluated for varying numbers of non-zero CSD coefficients. Lastly, a system level architecture with a multiplier utilizing the properties of the CSD number representation system is proposed.

Journal ArticleDOI
TL;DR: A Gaussian interpolation Filter and cubic interpolation filter are presented as more accurate interpolation filters compared to the conventional linear interpolationfilter and OFDM and in particular coded OFDM DAB systems are discussed.
Abstract: In this paper, a Gaussian interpolation filter and cubic interpolation filter are presented as more accurate interpolation filters compared to the conventional linear interpolation filter. In addition to an interpolation filter, a low pass filter using FFT and IFFT is also presented to reduce the noisy components of a channel estimate obtained by an interpolation filter. Channel estimates after low-pass filtering combined with interpolation filters can lower the error floor compared to the use of only interpolation filters. Computer simulation demonstrates that the presented channel estimation methods exhibit improved performance compared to the conventional linear interpolation filter. OFDM and in particular coded OFDM DAB systems are discussed.

Journal ArticleDOI
TL;DR: A new method of superresolving pupil-plane filter design in confocal microscopy is presented in which the properties of the desired point-spread function are specified and an optimization procedure is used to determine a suitable pupil-planes filter.
Abstract: We present a new method of superresolving pupil-plane filter design in confocal microscopy in which we specify the properties of the desired point-spread function and use an optimization procedure to determine a suitable pupil-plane filter. A new, flexible method of filter implementation using reconfigurable binary optical elements is described, and experimental results are presented.

Proceedings ArticleDOI
13 Jul 2000
TL;DR: In this paper, the Particle Filter is used for recursive estimation of the required density of the state vector as a set of random samples with associated weights for a single-sensor angle-only tracking problem with own ship maneuver.
Abstract: The tracking performance of the Particle Filter is compared with that of the Range-Parameterised EKF (RPEKF) and Modified Polar coordinate EKF (MPEKF) for a single-sensor angle-only tracking problem with ownship maneuver. The Particle Filter is based on representing the required density of the state vector as a set of random samples with associated weights. This filter is implemented for recursive estimation, and works by propagating the set of samples, and then updating the associated weights according to the new received measurement. The RPEKF, which is essentially a weighted sum of multiple EKF outputs, and the MPEKF are known for their robust angle-only tracking performance. This comparative study shows that the Particle Filter performance is the best, although the RPEKF is only marginally worse. The superior performance of the Particle Filter is particularly evident for high noise conditions where the EKF type trackers generally diverge. Also, the Particle Filter and the RPEKF are found to be robust to the level of a priori knowledge of initial target range. On the contrary, the MPEKF exhibits degraded performance for poor initialisation.

Proceedings ArticleDOI
12 Dec 2000
TL;DR: In this paper, a linear matrix inequality (LMI) based filter design approach for fixed-order robust fault detection and isolation (FDI) is examined, which provides necessary and sufficient conditions for the existence of a solution to detect and isolation of faults using an H/sub /spl infin// formulation.
Abstract: A linear matrix inequality (LMI) based filter design approach for fixed-order robust fault detection and isolation (FDI) is examined. The proposed filter design provides necessary and sufficient conditions for the existence of a solution to the detection and isolation of faults using an H/sub /spl infin// formulation. These conditions are expressed in terms of LMIs with matrix rank constraints, and a parameterization of all admissible filters is provided, which corresponds to a feasible solution. A convex LMI problem is obtained for the full-order FDI filter design. Finally, the proposed methods are demonstrated using a structural system simulation example, which include faulty actuators, sensors and external disturbances.

PatentDOI
TL;DR: In this paper, a cascade of two narrow-band filters Ai(Z) and Bi(Zng) with a fixed delay is proposed to represent the feedback path in each subband.
Abstract: A new subband feedback cancellation scheme is proposed, capable of providing additional stable gain without introducing audible artifacts. The subband feedback cancellation scheme employs a cascade of two narrow-band filters Ai(Z)and Bi(Z)ng with a fixed delay, instead of a single filter Wi(Z)and a delay to represent the feedback path in each subband. The first filter, Ai(Z), is called the training filter, and models the static portion of the feedback path in ith subband, including microphone, receiver, ear canal resonance, and other relatively static parameters. The training filter can be implemented as a FIR filter or as an IIR filter. The second filter, B?I?(Z), is called a tracking filter and is typically implemented as a FIR filter with fewer taps than the training filter. This second filter tracks the variations of the feedback path in the i?th? subband caused by jaw movement or objects close to the ears of the user.

Journal ArticleDOI
TL;DR: A significant advantage resulting from the application of the proposed SVD filter lies in its ability to perform noise suppression independently on a single lead ECG record with only a limited number of data samples.
Abstract: The proposed filter assumes the noisy electrocardiography (ECG) to be modeled as a signal of deterministic nature, corrupted by additive muscle noise artefact. The muscle noise component is treated to be stationary with known second-order characteristics. Since noise-free ECG is shown to possess a narrow-band structure in discrete cosine transform (DCT) domain and the second-order statistical properties of the additive noise component is preserved due to the orthogonality property of DCT, noise abatement is easily accomplished via subspace decomposition in the transform domain. The subspace decomposition is performed using singular value decomposition (SVD), The order of the transform domain SVD filter required to achieve the desired degree of noise abatement is compared to that of a suboptimal Wiener filter using DCT. Since the Wiener filter assumes both the signal and noise structures to be statistical, with a priori known second-order characteristics, it yields a biased estimate of the ECG beat as compared to the SVD filter for a given value of mean-square error (mse). The filter order required for performing the subspace smoothing is shown to exceed a certain minimal value for which the mse profile of the SVD filter follows the minimum-mean-square error (mmse) performance warranted by the suboptimal Wiener filter. The effective filter order required for reproducing clinically significant features in the noisy ECG is then set by an upper bound derived by means of a finite precision linear perturbation model. A significant advantage resulting from the application of the proposed SVD filter lies in its ability to perform noise suppression independently on a single lead ECG record with only a limited number of data samples.

Journal ArticleDOI
TL;DR: An approximate time-frequency design of both optimal filters is proposed, and bounds are presented that show that for underspread processes, the time- frequencies designed filters are nearly optimal.
Abstract: This paper presents a time-frequency framework for optimal linear filters (signal estimators) in nonstationary environments. We develop time-frequency formulations for the optimal linear filter (time-varying Wiener filter) and the optimal linear time-varying filter under a projection side constraint. These time-frequency formulations extend the simple and intuitive spectral representations that are valid in the stationary case to the practically important case of underspread nonstationary processes. Furthermore, we propose an approximate time-frequency design of both optimal filters, and we present bounds that show that for underspread processes, the time-frequency designed filters are nearly optimal. We also introduce extended filter design schemes using a weighted error criterion, and we discuss an efficient time-frequency implementation of optimal filters using multiwindow short-time Fourier transforms. Our theoretical results are illustrated by numerical simulations.

Patent
25 Oct 2000
TL;DR: In this paper, a sensor array (10 sub.1-N) is defined by a multiple of coefficients and the coefficients are set so as to maximize the signal to noise ratio of the receiving array's output.
Abstract: A sensor array (10 sub.1-N) receiving system which incorporates one or more filters (16 sub.1-N) that are capable of adaptive and/or fixed operation. The filters are defined by a multiple of coefficients and the coefficients are set so as to maximize the signal to noise ratio of the receiving array's output. In one preferred embodiment, the filter coefficients are adaptively determined and are faded into a predetermined group of fixed values upon the occurrence of a specified event. Thereby, allowing the sensor array (10 sub.1-N) to operate in both the adaptive and fixed modes, and providing the array with the ability to employ the mode most favorable for a given operating environment. In another preferred embodiment, the filter coefficients are set to a fixed group of values which are determined to be optimal for a predefined noise environment.

Patent
Liza G. Boland1, Johan Janssen1
18 Dec 2000
TL;DR: In this paper, a filter structure is provided that facilitates the use of the filter as either a continuous delay Farrow filter or a selectable delay polyphase filter, and an inversion of Farrow filters is presented.
Abstract: A filter structure is provided that facilitates the use of the filter as either a continuous delay Farrow filter or a selectable delay polyphase filter. The less complex polyphase filter is used when the desired scale substantially corresponds to a defined phase of the polyphase filter, or when time or power is not available to achieve the desired scale exactly; otherwise, the continuous delay Farrow filter is used. By providing an ability to switch to a continuous delay, the number of stages of the polyphase filter can be reduced. Additionally, an inversion of a Farrow filter is presented that provides for a continuous delay decimation filter with substantially reduced computational complexity compared to a direct embodiment of a Farrow filter. This inverted filter is also configurable as a polyphase filter, to provide selectable scale-resolution capabilities.

Proceedings ArticleDOI
05 Jun 2000
TL;DR: With this new filter and using multiple tacho references, waveforms, as well as amplitude and phase may be extracted without the beating interactions that are associated with conventional methods.
Abstract: The filter characteristics of the Vold-Kalman (1993, 1960, 1961) order tracking filter are presented. Both the frequency response as well as the time response and their time-frequency relationship have been investigated for different filter types and guidelines for optimum choice of filter parameters are presented. The Vold-Kalman filter allows for the high performance simultaneous tracking of orders in systems with multiple independent shafts. With this new filter and using multiple tacho references, waveforms, as well as amplitude and phase may be extracted without the beating interactions that are associated with conventional methods. Orders extracted as waveforms have no phase bias, and may hence be used for playback, synthesis and tailoring.

Book
29 Feb 2000
TL;DR: This paper presents a meta-modelling framework for tuning in Continuous-time Filters using MOS Capacitor Modeling and a review of Integrator Architectures to explore further applications of Scaling.
Abstract: List of Figures. List of Tables. Preface. 1. Introduction. 2. MOS Capacitor Modeling. 3. A Review of Integrator Architectures. 4. Time Scaling in Electrical Networks. 5. Filter Design. 6. Filter Testing and Measurement Results. 7. Further Applications of Scaling. 8. Tuning in Continuous-time Filters. Appendices. Index.

Journal ArticleDOI
TL;DR: A digital filter is a basic building block in any digital signal processing (DSP) system and simulation results presented show how finite bit precisions can affect the performance of a digital filter.
Abstract: A digital filter is a basic building block in any digital signal processing (DSP) system. The simulation results presented show how finite bit precisions can affect the performance of a digital filter. IIR filters are shown to be even more susceptible to finite bit precision effects than FIR filters. However, these effects can be reduced using the IIR filter with a cascaded structure.

Journal ArticleDOI
TL;DR: The proposed algorithm for adaptive beamforming in the OFDM system is derived by calculating the pilot error signals in the frequency domain, transforming the frequency-domain error signals into time- domain error signals, and updating the filter coefficients of the adaptive beamformer in the direction of minimizing the MSE.
Abstract: This paper presents an adaptive beamforming algorithm for an OFDM system with an adaptive array antenna. The proposed algorithm for adaptive beamforming in the OFDM system is derived by (1) calculating the pilot error signals in the frequency domain, (2) transforming the frequency-domain error signals into time-domain error signals, (3) updating the filter coefficients of the adaptive beamformer in the direction of minimizing the MSE. The convergence behavior and performance improvement of the proposed approach are investigated through computer simulation by applying it to the conventional OFDM system.