Showing papers on "Filter design published in 1997"

Hybrid solutions for improving passive filter performance in high power applications

Abstract: This paper presents a new control scheme for a parallel hybrid active filter system intended for harmonic compensation of large nonlinear loads up to 50 MVA, to meet IEEE 519 recommended harmonic standards. The active filter is small rated, 2%-3% of load kilovoltampere rating. The control scheme is based on the concept of synthesizing a dynamically variable inductance, and its usefulness is demonstrated for an active filtering application. A synchronous reference frame (SRF) controller implements the dynamically varying negative or positive inductance by generating active filter inverter voltage commands. This variable inductance controller parallel hybrid active filter system can selectively synthesize multiple "active inductances" at dominant harmonic frequencies without affecting passive filter impedances at all other frequencies. This controller also provides a "current limiting" function to prevent passive filter overloading under ambient harmonic loads and/or supply voltage distortions. Three implementation variations of a parallel hybrid active filter system are presented. This paper also proposes the use of power factor correction capacitors as low cost passive filters for a parallel hybrid active filter system, which are controlled to provide either single or multiple tuned harmonic sinks and to increase cost effectiveness for high power applications. Simulation results validate the proposed variable inductance controller operation for mistuned passive filters.

Multimode digital modem

Abstract: A modem operating selectively in the voice frequency and higher frequency bands which supports multiple line codes. A DSP is used to implement different existing ADSL line codes on the same hardware platform. The modem negotiates in real time for a desired line transmission rate to accommodate line condition and service cost requirements which may be implemented at the beginning of each communication session by exchange of tones between modems. A four step MDSL modem initialization process provides line code and rate compatibility. The handshake protocol and receiver algorithm for CAP based MDSL modems allows reliable modem synchronization over severely amplitude distorted channels and makes use of a short length sequence to train a synchronizing equalizer at the receiver. The algorithm and corresponding training sequence to train the transmitter filter are provided. After training to this sequence, a matched filter or correlator detects the inverted sync sequence. Detection of the inverted sequence signals commencement of normal reference training of the demodulation equalizers. An internal state machine in an MDSL modem records and monitors line status and notifies state change to other MDSL and host processor. The protocol for exchanging line connection management messages is a simplified LCP for MDSL. In a DMT system, a transmitter filter reduces the length of effective channel impulse response. Iimplementation of the filter combines time domain convolution and frequency domain multiplication to reduce needed computation power. The filter coefficients update may occur through a feedback channel.

Application of filter sharpening to cascaded integrator-comb decimation filters

Abstract: A new architecture for the implementation of high-order decimation filters is described. It combines the cascaded integrator-comb (CIC) multirate filter structure with filter sharpening techniques to improve the filter's passband response. This allows the first-stage CIC decimation filter to be followed by a fixed-coefficient second-stage filter, rather than a programmable filter, thereby achieving a significant hardware reduction over existing approaches. Furthermore, the use of fixed-coefficient filters in place of programmable-coefficient filters improves the overall throughput rate. The resulting architecture is well suited for single-chip VLSI implementation with very high data-sample rates. We discuss an example with specifications suitable for use in a wideband satellite communication subband tuner system and for signal analysis.

Review of high-performance three-phase power-factor correction circuits

Abstract: This paper reviews progress in topology, control, and design aspects in three-phase power-factor correction (PFC) techniques. Different switching rectifier topologies are presented for various applications. Representative soft-switching schemes, including zero-voltage and zero-current switched pulsewidth modulated (PWM) techniques, are investigated. Merits and limitations of these techniques are discussed and illustrated by experimental results obtained on prototype converters. Control and input filter design issues are also discussed.

Game theory approach to discrete H/sub /spl infin// filter design

Abstract: A finite-horizon discrete H/sub /spl infin// filter design with a linear quadratic (LQ) game approach is presented. The exogenous inputs composed of the "hostile" noise signals and system initial condition are assumed to be finite energy signals with unknown statistics. The design criterion is to minimize the worst possible amplification of the estimation error signals in terms of the exogenous inputs, which is different from the classical minimum variance estimation error criterion for the modified Wiener or Kalman filter design. The approach can show how far the estimation error can be reduced under an existence condition on the solution to a corresponding Riccati equation. A numerical example is given to compare the performance of the H/sub /spl infin// filter with that of the conventional Kalman filter.

Evaluation and design of filters using a Taylor series expansion

Abstract: We describe a new method for analyzing, classifying, and evaluating filters that can be applied to interpolation filters as well as to arbitrary derivative filters of any order. Our analysis is based on the Taylor series expansion of the convolution sum. Our analysis shows the need and derives the method for the normalization of derivative filter weights. Under certain minimal restrictions of the underlying function, we are able to compute tight absolute error bounds of the reconstruction process. We demonstrate the utilization of our methods to the analysis of the class of cubic BC-spline filters. As our technique is not restricted to interpolation filters, we are able to show that the Catmull-Rom spline filter and its derivative are the most accurate reconstruction and derivative filters, respectively, among the class of BC-spline filters. We also present a new derivative filter which features better spatial accuracy than any derivative BC-spline filter, and is optimal within our framework. We conclude by demonstrating the use of these optimal filters for accurate interpolation and gradient estimation in volume rendering.

Optimised input filter design and low-loss switching techniques for a practical matrix converter

Abstract: The matrix converter permits frequency conversion in a single-stage process. The perceived disadvantage of the matrix converter is that conduction losses are high. However, semisoft current commutation and optimal sequence switching can be used to minimise commutation losses so that at high switching frequencies the total losses in the matrix converter can be less than those in a conventional rectifier–inverter combination. The viability of the matrix converter depends to a large extent on the size and cost of the input filter components required to meet international power quality standards. In the paper filter designs are examined and guidelines established. Practical tests have been carried out on a 3.5 kW converter to validate computer models. It is concluded that the matrix converter is viable if the right combination of semiconductor switching techniques and input filter design are employed.

Frequency estimation by demodulation of two complex signals

Abstract: This paper presents a method for frequency estimation in a power system by demodulation of two complex signals. In power system analysis, the /spl alpha//spl beta/-transform is used to convert three phase quantities to a complex quantity where the real part is the in-phase component and the imaginary part is the quadrature component. This complex signal is demodulated with a known complex phasor rotating in opposite direction to the input. The advantage of this method is that the demodulation does not introduce a double frequency component. For signals with high signal to noise ratio, the filtering demand for the double frequency component can often limit the speed of the frequency estimator. Hence, the method can improve fast frequency estimation of signals with good noise properties. The method loses its benefits for noisy signals, where the filter design is governed by the demand to filter harmonics and white noise. The method has been previously published, but not explored to its potential. The paper presents four examples to illustrate the strengths and weaknesses of the method.

Clutter rejection filters in color flow imaging: a theoretical approach

Abstract: A general class of linear clutter rejection filters is described, covering the commonly used filter types including FIR/IIR filters with linear initialization, as well as regression filters, where the clutter component is estimated by least square curve fitting. The filter can be described by a complex valued matrix, and a frequency response is defined. However, in contrast to a time invariant filter, the general linear filter may create frequency components which are not present in the input signal. This produces bias in the velocity and velocity spread estimates. It is shown that the clutter filter effect on the autocorrelation estimates can be described by a frequency domain transfer function, but unlike time invariant filters, the transfer function is different for each temporal lag of the autocorrelation function. Using a two dimensional (axial and temporal dimension) model of the received signal, the bias in velocity and velocity spread is quantified, both for the autocorrelation algorithm and the time shift cross-correlation estimator. Theoretical expressions, as well as numerical examples are given.

A nonlinear filter for estimating a sinusoidal signal and its parameters in white noise: on the case of a single sinusoid

Abstract: A nonlinear filter is proposed for estimating a complex sinusoidal signal and its parameters (frequency, amplitude, and phase) from measurements corrupted by white noise. This filter is derived by applying an extended complex Kalman filter (ECKF) to a nonlinear stochastic system whose state variables are a function of its frequency and a sample of an original signal, and then, proof of the stability is given in the case of a single complex sinusoid. Simulations demonstrate that the proposed nonlinear filter is effective as a method for estimating a single complex sinusoid and its frequency under a low signal-to-noise ratio (SNR). In addition, the effect of the initial condition in the filter on frequency estimation is also discussed.

The Dolph-Chebyshev Window: A Simple Optimal Filter

Abstract: Analyzed data for numerical prediction can be effectively initialized by means of a digital filter. Computation time is reduced by using an optimal filter. The construction of optimal filters involves the solution of a nonlinear minimization problem using an iterative procedure. In this paper a simple filter based on the Dolph‐Chebyshev window, which has properties similar to those of an optimal filter, is described. It is shown to be optimal for an appropriate choice of parameters. It has an explicit analytical expression and is easily implemented. Its effectiveness is demonstrated by application to Richardson’s forecast: the initial pressure tendency is reduced from 145 hPa pe r6ht o 20.9 hPa per 6 h. Use of the filter is not restricted to initialization; it may also be applied as a weak constraint in four-dimensional data assimilation.

Results of a multicenter study of the retrievable tulip vena cava filter : Early clinical experience

Abstract: To evaluate clinically a new, retrievable vena caval filter in a multicenter study. The Tulip filter is a stainless steel half-basket that is suitable for antegrade or retrograde insertion via an 8.5 Fr introducer sheath. The filter can be retrieved via the jugular approach using an 11 Fr coaxial retrieval system. Forty-eight filters were implanted via the femoral approach and 38 via the jugular approach in 83 patients. Follow-up examinations (plain films, colorcoded duplex sonography) were performed up to 3 years after filter insertion (mean 136 days) in 75 patients. Twenty-seven patients were screened by colorcoded duplex sonography for insertion site thrombosis. An appropriate filter position was achieved in all cases. Insertion problems occurred in 3 cases; these were not due to the filter design but to an imperfect prototype insertion mechanism that has now been modified (n=2) or a manipulation error (n=1). In 2 of these cases the filters were replaced percutaneously; 1 patient required venotomy for filter removal. No further complications due to filter insertion occurred. Two filters were used as temporary devices and were successfully removed after 6 and 11 days, respectively. There was 1 fatal recurrent pulmonary embolism (PE) and 2 non-fatal PE, 5 complete and 3 partial caval occlusions, and 3 caudal migrations of the filter. Insertion site venous thrombosis was not seen in the 27 patients monitored for this complication. Precise placement of the Tulip filter is feasible by either access route and the device appears mechanically stable. Further observations are needed to confirm that safe filter removal is practical up to 10 days after its insertion.

Equiripple FIR filter design by the FFT algorithm

Abstract: The fast Fourier transform (FFT) algorithm has been used in a variety of applications in signal and image processing. In this article, a simple procedure for designing finite-extent impulse response (FIR) discrete-time filters using the FFT algorithm is described. The zero-phase (or linear phase) FIR filter design problem is formulated to alternately satisfy the frequency domain constraints on the magnitude response bounds and time domain constraints on the impulse response support. The design scheme is iterative in which each iteration requires two FFT computations. The resultant filter is an equiripple approximation to the desired frequency response. The main advantage of the FFT-based design method is its implementational simplicity and versatility. Furthermore, the way the algorithm works is intuitive and any additional constraint can be incorporated in the iterations, as long as the convexity property of the overall operations is preserved. In one-dimensional cases, the most widely used equiripple FIR filter design algorithm is the Parks-McClellan algorithm (1972). This algorithm is based on linear programming, and it is computationally efficient. However, it cannot be generalized to higher dimensions. Extension of our design method to higher dimensions is straightforward. In this case two multidimensional FFT computations are needed in each iteration.

A transfer function approach to harmonic filter design

Abstract: This article details a new transfer function approach in passive harmonic filter design for industrial and commercial power system applications Filter placement along with six common filter configurations are presented Harmonic impedance, voltage division and current division transfer functions are derived and used in a practical filter design procedure that incorporates IEEE-519 distortion limits directly into the design and component specification process A simple four-step filter design procedure is outlined and used in a variable speed motor drive pumping plant application

Design of recursive 1-D variable filters with guaranteed stability

Abstract: The digital filters with adjustable frequency-domain characteristics are called variable filters. Variable filters are used in many signal processing fields, but the recursive variable filters are extremely difficult to design due to the stability problem. This paper proposes a new method for designing recursive one-dimensional (1-D) variable filters whose stability is guaranteed. The method finds the coefficients of the transfer function of a variable digital filter as the multidimensional (M-D) polynomials of a few variables. The variables specify different frequency-domain characteristics, thus, we call the variables the spectral parameters. In applying the resulting variable filters, substituting different values of the spectral parameters into the M-D polynomials will obtain different filter coefficients and, thus, obtain different frequency-domain characteristics. To guarantee the stability, we first perform coefficient substitutions on the denominator coefficients such that they satisfy the stability conditions. Then both denominator and numerator coefficients are determined as M-D polynomials. In determining the M-D polynomials, we also propose an efficient least-squares approximation method that requires only solving simultaneous linear equations. Two examples are given to show the effectiveness of the proposed variable filter design technique.

Data-driven design of RASTA-like filters.

Abstract: We describe use of Linear Discriminant Analysis (LDA) for data-driven automatic design of RASTA-like lters. The LDA applied to rather long segments of time trajectories of critical-band energies yields FIR lters to be applied to these time trajectories in the feature extraction module. Frequency responses of the rst three discriminant vectors are in principle consistent with the ad hoc designed RASTA, delta and double-delta lters. On a connected digit task the new features outperform the original RASTA processing.

IIR multiple notch filter design based on allpass filter

Abstract: A new technique for IIR multiple notch filter design is proposed. The specification of notch filter is first transformed into that of allpass filter. Then, we develop an effective approach to design this desired allpass filter. The realization of proposed notch filter is equivalent to the realization of an allpass filter. Due to the mirror-image symmetry relation between the numerator and denominator polynomials of allpass filter, the notch filter can be realized by a computationally efficient lattice structure with very low sensitivity. Moreover, some examples are presented to examine the effectiveness of proposed method.

Subband acoustic noise suppression

Abstract: A conditioning circuit for use with a microphone 224 placed near a loudspeaker 206 has microphone-in 225 and speaker-line 208 input terminals, an echo canceller circuit (212, 216, 220, 222), a noise-suppresser circuit 230, and a synthesis filter 234 coupling noise-suppresser circuit 230 to a microphone-out output terminal 236. Microphone-in 225 and speaker-line 108 input terminals respectively receive microphone and speaker signals. The echo canceller circuit is coupled between microphone-in 225 and speaker-line 108 input terminals and produces a subband reduced-echo microphone signal by (i) transforming the microphone signal into a subband microphone signal and the speaker signal into a filtered subband speaker signal, and (ii) subband subtracting the filtered subband speaker signal from the subband microphone signal. Noise-suppresser circuit 230 is coupled to the echo canceller circuit to produce a subband reduced-noise, reduced-echo microphone signal by subband noise suppression of the subband reduced-echo microphone signal. Synthesis filter 234 transforms the subband reduced-noise, reduced-echo microphone signal into a fullband reduced-noise, reduced-echo microphone signal. Because the subband signal is not restored to a fullband signal until after undergoing both echo cancellation and noise reduction, my system requires less processing power than systems that apply a synthesis filter between subband echo cancellation and noise reduction. This system is useful in hands-free telephones, especially hands-free cellular telephones used in automobiles. A circuit 538 for subbahd detection of near-end speech in the microphone signal can be provided so echo cancellation filter coefficients can be automatically frozen when near-end speech is detected.

A simple design method for near-perfect-reconstruction nonuniform filter banks

Abstract: In this correspondence, we propose a simple design method for nonuniform integer-decimated filter banks based on a uniform cosine-modulated filter bank. The resulting distortion and aliasing are comparable to the stopband attenuation of the prototype filter. Examples are given to demonstrate the proposed method.

Designing harmonic filters for adjustable speed drives to comply with IEEE-519 harmonic limits

Abstract: This paper discusses the application of the revised IEEE-519 Harmonics standards to typical industrial facilities employing adjustable speed drives (ASDs). The harmonic generation characteristics of ASDs are described, Requirements for control of the harmonic currents are developed as a function of the ASD characteristics, overall plant loading level, power system characteristics, and power factor correction requirements. Filter design procedures are presented for controlling the harmonic currents injected onto the power system.

A programmable 36-mhz digital filter bank for radio science

Abstract: We have designed and constructed an agile 8-channel digital filter bank on a 9U VME board that operates at a maximum clock rate of 36 MHz. A set of these boards have been employed to create 100-200 MHz, 50-100 channel spectrometers. Our applications involve detection of weak signals in a background of noise--pulsar radio astronomy and the search for narrow-band radio emissions from extraterrestrial civilizations. The agility factors include total bandwidth, spacing of filter channels, selection of filter response and choice of output data format (voltage or detected power). The input is a complex analog voltage centered on zero frequency which is passed to the board via coaxial ports on the front panel. The computational kernels on this board are Harris HSP 43168 Finite Impulse Response (FIR) filter devices. We can also operate the board in a 16-channel output mode. The 2-4 bit format digital output is presented to a custom backplane. Along with our development we have written a suite of simulation software tasks both to determine the sensitivity loss and non-linear gain response that result from quantization and to optimize filter responses.

Coupled-mode analysis of fiber-optic add drop filters for dense wavelength-division multiplexing.

Abstract: We present a coupled-mode theory of fiber-optic add drop filters, which involve directional coupling between two fibers combined with fiber Bragg gratings defined inside the coupling region. The analysis self-consistently accounts for both the directional and the reflection coupling, and the propagation constants and structure of the supermodes of the combined structure are derived. We present a detailed analysis of a filter design based on identical fibers. The calculated device parameters satisfy the requirements for dense wavelength-division multiplexing applications.

A mixture-of-experts framework for adaptive Kalman filtering

Abstract: This paper proposes a modular and flexible approach to adaptive Kalman filtering using the framework of a mixture-of-experts regulated by a gating network. Each expert is a Kalman filter modeled with a different realization of the unknown system parameters such as process and measurement noise. The gating network performs on-line adaptation of the weights given to individual filter estimates based on performance. This scheme compares very favorably with the classical Magill filter bank, which is based on a Bayesian technique, in terms of: estimation accuracy; quicker response to changing environments; and numerical stability and computational demands. The proposed filter bank is further enhanced by periodically using a search algorithm in a feedback loop. Two search algorithms are considered. The first algorithm uses a recursive quadratic programming approach which extremizes a modified maximum likelihood function to update the parameters of the best performing filter in the bank. This particular approach to parameter adaptation allows a real-time implementation. The second algorithm uses a genetic algorithm to search for the parameter vector and is suited for post-processed data type applications. The workings and power of the overall filter bank and the suggested adaptation schemes are illustrated by a number of examples.

Efficient polyphase quadrature digital tuner

Abstract: A polyphase quadrature digital tuner system which converts input signals to baseband inphase and quadrature signal components. The system includes a signal receiver which receives the input signals having a frequency centered around a predetermined carrier frequency. A signal processor continuously samples the input signals and multiplies selected portions of the input signals by a value of 1 or -1 to produce discrete sequences of N input samples, where N is an integer. An inphase signal channel includes a first set of N filters in a first filter stage each having respective filter coefficients, the first set of filters arranged to receive the discrete sequences, and a first signal summer which sums the outputs of the first set of N filters to produce the inphase signal component. A quadrature signal channel includes a second set of N filters in the first filter stage each having respective filter coefficients, the second set of filters arranged to receive the discrete sequences, and a second signal summer which sums the outputs of the second set of N filters to produce the quadrature signal component. The input samples are provided to the inphase and quadrature signal channels so that each filter of both channels receives one input sample of each sequence.

System for dynamically adapting the length of a filter

Abstract: A novel process by which the utilization of a central processing unit (CPU) in performing filtering operations can be reduced by shortening the filter's length thus degrading the performance of the system down to a predetermined level or threshold. If the filter is adaptive (150), the process waits for the filter to converge (152). Then, the quality criteria is measured (154). If the measured quality criteria is above the quality threshold (158), then M taps are removed from the filter (156) and the process returns to step 150. If the measured quality criteria is below the threshold, M taps are added back to the filter (160) and the process terminates.

Optimality of the maximum average correlation height filter for detection of targets in noise

Abstract: A statistical analysis is provided for the properties of the re- cently developed maximum average correlation height (MACH) filter (Mahalanobis et al. 1994). It is shown that the MACH filter can be inter- preted as an optimum filter for the detection of targets in additive noise. A rationale is given for using a popular peak-to-sidelobe ratio metric to characterize the output of the MACH filter. Finally, the performance of the MACH filter is compared to that of the matched spatial filter (MSF) in terms of the relation between the probabilities of correct detection and false alarm, which is represented as a receiver operating characteristic (ROC) curve. © 1997 Society of Photo-Optical Instrumentation Engineers. (S0091-3286(97)00910-0)

Universal current-mode filter with reduced number of active and passive components

Abstract: A new current-mode (CM) universal active filter with single-input and three-outputs (SITO) employing only four CCIIs and a minimum number of passive components is presented. The proposed filter based on an RLC shunt circuit, has a good sensitivity performance and achieves the desired filter characteristics without any component matching.

Elliptical cavity resonators for dual-mode narrow-band filters

Abstract: A novel cavity resonator with elliptical cross section is proposed in order to realize dual-mode narrow-band filters without tuning and coupling elements. The absence of any discontinuity inside the cavities significantly enhances the unloaded Q, the ability to operate with higher power levels, and the ease of manufacturing. Proper choice of the ellipticity and of the inclination angle controls the desired coupling and tuning actions. Dual-mode coupling is generated by the step discontinuity between the input rectangular waveguide and an inclined elliptical waveguide. A rigorous full-wave electromagnetic model for this discontinuity has been developed and validated using a specialized hardware configuration. Experimental data compare very favorably with theoretical results. Representative prototypes of elliptical cavities exhibiting various degrees of coupling have been carefully measured proving the accuracy of the model and its applicability for narrow-band X- and Ku-band filter design. The full-wave analysis of a complete four-pole narrow-band elliptic filter at 12 GHz and the measured response of a corresponding prototype demonstrate the capability of achieving reliable results using the proposed approach.

A comparison of normal estimation schemes

Abstract: The task of reconstructing the derivative of a discrete function is essential for its shading and rendering as well as being widely used in image processing and analysis. We survey the possible methods for normal estimation in volume rendering and divide them into two classes based on the delivered numerical accuracy. The three members of the first class determine the normal in two steps by employing both interpolation and derivative filters. Among these is a new method which has never been realized. The members of the first class are all equally accurate. The second class has only one member and employs a continuous derivative filter obtained through the analytic derivation of an interpolation filter. We use the new method to analytically compare the accuracy of the first class with that of the second. As a result of our analysis we show that even inexpensive schemes can in fact be more accurate than high order methods. We describe the theoretical computational cost of applying the schemes in a volume rendering application and provide guidelines for helping one choose a scheme for estimating derivatives. In particular we find that the new method can be very inexpensive and can compete with the normal estimations which pre-shade and pre-classify the volume (M. Levoy, 1988).