Showing papers on "Filter design published in 1999"

Tri-state median filter for image denoising

Abstract: A novel nonlinear filter, called tri-state median (TSM) filter, is proposed for preserving image details while effectively suppressing impulse noise. We incorporate the standard median (SM) filter and the center weighted median (CWM) filter into a noise detection framework to determine whether a pixel is corrupted, before applying filtering unconditionally. Extensive simulation results demonstrate that the proposed filter consistently outperforms other median filters by balancing the tradeoff between noise reduction and detail preservation.

Optical Filter Design and Analysis: A Signal Processing Approach

Abstract: From the Publisher: A Unique, Cutting-Edge Approach to Optical Filter Design With more and more information being transmitted over fiber-optic lines, optical filtering has become crucial to the advanced functionality of today’s communications networks. Helping researchers and engineers keep pace with this rapidly evolving technology, this book presents digital processing techniques for optical filter design. This higher-level approach focuses on filter characteristics and enables readers to quickly calculate the filter response as well as tackle larger and more complex filters. The authors incorporate numerous theoretical and experimental results from the literature and discuss applications to a variety of systems—including the new wavelength division multiplexing (WDM) technology, which is fast becoming the preferred method for system upgrade and expansion. Special features of this book include: *The theory underlying various architectures that can approximate any filter function *Filter design techniques applicable to a broad range of materials systems—from silica to fiber to microelectromechanical (MEM) systems *Design examples relevant to filters for WDM systems and planar waveguide devices *250 figures as well as problem sets for use in graduate-level studies

Disturbance observer and feedforward design for a high-speed direct-drive positioning table

Abstract: Design and implementation of a discrete-time tracking controller for a precision positioning table actuated by direct-drive motors is considered. The table has acceleration capabilities in excess of 5 G, positioning accuracy at the micron level, and is used in applications such as semiconductor packaging. The controller proposed uses a disturbance observer and proportional derivative (PD) compensation in the feedback path and a zero phase error tracking controller and zero phase low-pass filter in the feedforward path. The existing disturbance observer design techniques are extended to account for time delay in the plant. Practical difficulties with excessive feedforward gains are examined and a low-order filter design method is proposed. Experimental results for quantized low-order position reference trajectories, which are commonly used in industrial systems, demonstrate the effectiveness of the approach.

Bark and ERB bilinear transforms

Abstract: Use of a bilinear conformal map to achieve a frequency warping nearly identical to that of the Bark frequency scale is described Because the map takes the unit circle to itself, its form is that of the transfer function of a first-order allpass filter Since it is a first-order map, it preserves the model order of rational systems, making it a valuable frequency warping technique for use in audio filter design A closed-form weighted-equation-error method is derived that computes the optimal mapping coefficient as a function of sampling rate, and the solution is shown to be generally indistinguishable from the optimal least-squares solution The optimal Chebyshev mapping is also found to be essentially identical to the optimal least-squares solution The expression 08517[arctan(006583fs)]/sup 1/2/-0916 is shown to accurately approximate the optimal allpass coefficient as a function of sampling rate f/sub s/ in kHz for sampling rates greater than 1 kHz A filter design example is included that illustrates improvements due to carrying out the design over a Bark scale Corresponding results are also given and compared for approximating the related "equivalent rectangular bandwidth (ERB) scale" of Moore and Glasberg (ACTA Acustica, vo82, p335-45, 1996) using a first-order allpass transformation Due to the higher frequency resolution called for by the ERB scale, particularly at low frequencies, the first-order conformal map is less able to follow the desired mapping, and the error is two to three times greater than the Bark-scale case, depending on the sampling rate

An improved particle filter for non-linear problems

Abstract: The Kalman filter provides an effective solution to the linear Gaussian filtering problem However where there is nonlinearity, either in the model specification or the observation process, other methods are required Methods known generically as `particle filters' are considered These include the condensation algorithm and the Bayesian bootstrap or sampling importance resampling (SIR) filter These filters represent the posterior distribution of the state variables by a system of particles which evolves and adapts recursively as new information becomes available In practice, large numbers of particles may be required to provide adequate approximations and for certain applications, after a sequence of updates, the particle system will often collapse to a single point A method of monitoring the efficiency of these filters is introduced which provides a simple quantitative assessment of sample impoverishment and the authors show how to construct improved particle filters that are both structurally efficient in terms of preventing the collapse of the particle system and computationally efficient in their implementation This is illustrated with the classic bearings-only tracking problem

A new algorithm for elimination of common subexpressions

Abstract: The problem of an efficient hardware implementation of multiplications with one or more constants is encountered in many different digital signal-processing areas, such as image processing or digital filter optimization. In a more general form, this is a problem of common subexpression elimination, and as such it also occurs in compiler optimization and many high-level synthesis tasks. An efficient solution of this problem can yield significant improvements in important design parameters like implementation area or power consumption. In this paper, a new solution of the multiple constant multiplication problem based on the common subexpression elimination technique is presented. The performance of our method is demonstrated primarily on a finite-duration impulse response filter design. The idea is to implement a set of constant multiplications as a set of add-shift operations and to optimize these with respect to the common subexpressions afterwards. We show that the number of add/subtract operations can be reduced significantly this way. The applicability of the presented algorithm to the different high-level synthesis tasks is also indicated. Benchmarks demonstrating the algorithm's efficiency are included as well.

Modified DFT filter banks with perfect reconstruction

Abstract: In this paper, essential features of the recently introduced modified discrete Fourier transform DFT (MDFT) filter bank are presented. First, it is shown that all analysis and synthesis filters-obtained by appropriate complex modulation of a low-pass prototype filter-are linear phase. This is important for subband image coding applications. Another important property is the structure-inherent alias cancellation: all odd alias spectra are automatically compensated in the synthesis filter bank. Further, the MDFT filter bank provides perfect reconstruction for the same prototypes as for cosine-modulated filter banks. Thus, the same design methods can be used. Finally, different mappings of the input signal into the subbands are discussed and a comparison to the well-known cosine-modulated filter banks is given.

A receding horizon Kalman FIR filter for discrete time-invariant systems

Abstract: A receding horizon Kalman FIR filter is presented that combines the Kalman filter and the receding horizon strategy when the horizon initial state is assumed to be unknown. The suggested filter is a FIR filter form which has many good inherent properties. It can always be defined irrespective of singularity problems caused by unknown information about the horizon initial state. The suggested filter can be represented in either an iterative form or a standard FIR form. It is also shown that the suggested filter possesses the unbiasedness property and the remarkable deadbeat property irrespective of any horizon initial condition. The validity of the suggested filter is illustrated by numerical examples.

A new approach to subband adaptive filtering

Abstract: Subband adaptive filtering has attracted much attention lately. In this paper, we propose a new structure and a new formulation for adapting the filter coefficients. This structure is based on polyphase decomposition of the filter to be adapted and is independent of the type of filter banks used in the subband decomposition. The new formulation yields improved convergence rate when the LMS algorithm is used for coefficient adaptation. As we increase the number of bands in the filter, the convergence rate increases and approaches the rate that can be obtained with a flat input spectrum. The computational complexity of the proposed scheme is nearly the same as that of the fullband approach. Simulation results are included to demonstrate the efficacy of the new approach.

Feedback cancellation improvements

Abstract: Feedback cancellation apparatus uses a cascade of two filters along with a short bulk delay. The first filter is adapted when the hearing aid is turned on in the ear. This filter adapts quickly using a white noise probe signal, 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 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 shifts the filter response to make the most effective use of the limited number of filter coefficients.

Removing Noise and Preserving Details with Relaxed Median Filters

Abstract: In this paper, a median based filter called relaxed median filter is proposed. The filter is obtained by relaxing the order statistic for pixel substitution. Noise attenuation properties as well as edge and line preservation are analyzed statistically. The trade-off between noise elimination and detail preservation is widely analyzed. It is shown that relaxed median filters preserve details better than the standard median filter, and remove noise better than other median type filters.

FIR Filter Design via Spectral Factorization and Convex Optimization

Abstract: We consider the design of finite impulse response (FIR) filters subject to upper and lower bounds on the frequency response magnitude The associated optimization problems with the filter coefficients as the variables and the frequency response bounds as constraints are in general nonconvex Using a change of variables and spectral factorization we can pose such problems as linear or nonlinear convex optimization problems As a result we can solve them efficiently (and globally) by recently developed interior-point methods We describe applications to filter and equalizer design and the related problem of antenna array weight design

A new derivation of the APES filter

Abstract: We introduce a novel design criterion for data-dependent narrowband filters that are of interest in temporal or spatial spectral analysis applications. The solution to the design problem considered is shown to coincide with the previously introduced amplitude and phase estimation (APES) filter. The new derivation of APES in this article sheds more light on the properties of APES and provides some intuitive explanation of the performance superiority of the APES filter over the Capon filter.

Design of near perfect reconstruction oversampled filter banks for subband adaptive filters

Abstract: In this brief, a design algorithm for real-valued and complex-valued oversampled filter banks which yield a low level of inband alias and enable simple subband adaptive structures is presented. The filter banks are either based on complex modulation of a real-valued low-pass prototype or on the direct or modulated setups of real-valued filter banks. If real-valued filter banks are required, then the different channels will have different subsampling ratios so that the bandpass sampling theorem is not violated. This brief also presents design examples of real-valued and complex-valued filter banks.

Digital filter design for inversion problems in sound reproduction

Abstract: When it is attempted to invert, or eliminate the influence of, a single-channel system such as a loudspeaker response or a room response, some frequencies are usually boosted by an excessive amount by the inverse filter. In particular, inversion of experimentally measured data tends to cause an excessive boost of frequencies just below the Nyquist frequency. A similar problem is encountered when designing digital filters for a multichannel virtual source imaging system such as the stereo dipole. A digital filter design method is presented that allows selective attenuation of unwanted peaks in the spectrum of the reproduced sound by using frequency-dependent regularization.

Oscillation-test methodology for low-cost testing of active analog filters

Abstract: The oscillation-test strategy is a low cost and robust test method for mixed-signal integrated circuits. Being a vectorless test method, it allows one to eliminate the analog test vector generator. Furthermore, as the oscillation frequency is considered to be digital, it can be precisely analyzed using pure digital circuitry and can be easily interfaced to test techniques dedicated to the digital part of the circuit under test (CUT). This paper describes the design for testability (DFT) of active analog filters based on oscillation-test methodology. Active filters are transformed to oscillators using very simple techniques. The tolerance band of the oscillation frequency is determined by a Monte Carlo analysis taking into account the nominal tolerance of all circuit under test components. Discrete practical realizations and extensive simulations based on CMOS 1.2 /spl mu/m technology parameters affirm that the test technique presented for active analog filters ensures high fault coverage and requires a negligible area overhead. Finally, the DFT techniques investigated are very suitable for automatic testable filter synthesis and can be easily integrated in the tools dedicated to automatic filter design.

A trellis search algorithm for the design of FIR filters with signed-powers-of-two coefficients

Abstract: An efficient two-stage algorithm is presented for designing finite-impulse response (FIR) filters that employ sums of signed-powers-of-two (SPT) coefficients. In the first stage, a prototype filter is designed using a fast time-domain approximation. This is followed, in the second stage, where the design problem is formulated as a dynamic-programming-like recursive optimization problem, by a trellis search that optimizes the filter's frequency response. The proposed search algorithm, which iteratively designs filters that employ an increasing number of SPT terms, provides a means to control the filter's implementation complexity. Design examples demonstrate that our algorithm is capable of producing filters having a better frequency response than existing methods while using fewer SPT terms. We also show that the proposed algorithm can be used to design special FIR filters such as matched transmit and receive filters employing sums of signed-powers-of-two coefficients. Also presented is a modified algorithm that further reduces the required number of adders in a filter by exploiting redundancies within the coefficients.

Method and apparatus for enhancing noise-corrupted speech

Abstract: A noise suppression device receives data representative of a noise-corrupted signal which contains a speech signal and a noise signal, divides the received data into data frames, and then passes the data frames through a pre-filter to remove a dc-component and the minimum phase aspect of the noise-corrupted signal. The noise suppression device appends adjacent data frames to eliminate boundary discontinuities, and applies fast Fourier transform to the appended data frames. A voice activity detector of the noise suppression device determines if the noise-corrupted signal contains the speech signal based on components in the time domain and the frequency domain. A smoothed Wiener filter of the noise suppression device filters the data frames in the frequency domain using different sizes of a window based on the existence of the speech signal. Filter coefficients used for Wiener filter are smoothed before filtering. The noise suppression device modifies magnitude of the time domain data based on the voicing information outputted from the voice activity detector.

New finite-dimensional filters for parameter estimation of discrete-time linear Gaussian models

Abstract: The authors derive a new class of finite-dimensional recursive filters for linear dynamical systems. The Kalman filter is a special case of their general filter. Apart from being of mathematical interest, these new finite-dimensional filters can be used with the expectation maximization (EM) algorithm to yield maximum likelihood estimates of the parameters of a linear dynamical system. Important advantages of their filter-based EM algorithm compared with the standard smoother-based EM algorithm include: 1) substantially reduced memory requirements, and 2) ease of parallel implementation on a multiprocessor system. The algorithm has applications in multisensor signal enhancement of speech signals and also econometric modeling.

Speckle filtering of SAR images based on adaptive windowing

Abstract: Speckle noise usually occurs in synthetic aperture radar (SAR) images owing to coherent processing of SAR data. The most well-known image domain speckle filters are the adaptive filters using local statistics such as the mean and standard deviation. The local statistics filters adapt the filter coefficients based on data within a fixed running window. In these schemes, depending on the window size, there exists trade-off between the extent of speckle noise suppression and the capability of preserving fine details. The authors propose a new adaptive windowing algorithm for speckle noise suppression which solves the problem of window size associated with the local statistics adaptive filters. In the algorithm, the window size is automatically adjusted depending on regional characteristics to suppress speckle noise as much as possible while preserving fine details. Speckle noise suppression gets stronger in homogeneous regions as the window size increases succeedingly. In fine detail regions, by reducing the window size successively, edges and textures are preserved. The fixed-window filtering schemes and the proposed one are applied to both a simulated SAR image and an ERS-1 SAR image to demonstrate the excellent performance of the proposed adaptive windowing algorithm for speckle noise.

A broad-band tunable CMOS channel-select filter for a low-IF wireless receiver

Abstract: Researchers agree that the active filter for channel-selection limits the dynamic range in a fully integrated wireless receiver, which uses no external components. This paper describes a filter designed for a wideband wireless LAN receiver operating in the 2.4-2.48 GHz ISM band. The receiver converts the desired channel to a low IF to enable on-chip rejection of the image. The analog filter must pass up to a 10 MHz wide single-channel centered after downconversion to IF ranging from 5 to 10 MHz. The classic requirements on all RF IC's apply to this filter, namely how to achieve the desired frequency response with the highest linearity and lowest noise at a given current consumption. For a channel-select filter, linearity is specified as out-of-band 3rd order intercept (IP3), to limit the in-band intermodulation from large interferers lying in the filter stop band. This work addresses the problem in filter architecture and circuit implementation. The achieved dynamic range surpasses all other published filter designs, except those cases where gain is favorably interleaved with filtering.

A new approach for single-phase harmonic current detecting and its application in a hybrid active power filter

Abstract: In this paper, a new approach for single-phase harmonic current detection is presented. The approach was obtained through extending ideas of three-phase instantaneous reactive power theory and constructing a two-phase system from the existing single-phase circuit. By theoretical and simulation analysis, it is shown to be a precise approach, which can be easily realized and has the merits of better steady state and dynamic performance than conventional approaches that could be used in single-phase circuits. The approach was applied into a hybrid active power filter, which combines a series active filter and a shunt passive filter together and aims at solving the harmonics problem originated by high-power single-phase nonlinear load. Following the system configuration and basic principles of the hybrid active power filter, the overall detection and control algorithm, the PWM generating technique and the DC voltage stabilizing method are introduced in detail. The detection and control algorithm were then realized by a digital control circuit with DSP processor, and successfully employed in a prototype hybrid active power filter. Experimental results on the prototype verified the effectiveness of the new detecting approach, the performance of the control circuit and the filtering characteristics of the hybrid active power filter.

Noise control system

Abstract: A noise control system includes: a control sound generator for generating a control sound; an error detector for detecting an error signal between the control sound and noise; a noise detector for detecting a noise source signal; an adaptive filter for outputting a control signal; and a coefficient updator for updating a coefficient of the adaptive filter. The coefficient updator includes at least a first digital filter, a first coefficient update calculator, a second digital filter, a phase inverter, a third digital filter, and a second coefficient update calculator. Alternatively, the coefficient updator includes at least a first digital filter, a second digital filter, a third digital filter, a coefficient update calculator, a phase inverter, a first adder, and a second adder. In either case, the coefficient updator has a function of suppressing an increase in a coefficient gain of the adaptive filter in a predetermined frequency band.

Fast implementations of the filtered-X LMS and LMS algorithms for multichannel active noise control

Abstract: In some situations where active noise control could be used, the well-known multichannel version of the filtered-X least mean square (LMS) adaptive filter is too computationally complex to implement. We develop a fast, exact implementation of this adaptive filter for which the system's complexity scales according to the number of filter coefficients within the system. In addition, we extend computationally efficient methods for effectively removing the delays of the secondary paths within the coefficient updates to the multichannel case, thus yielding fast implementations of the LMS adaptive algorithm for multichannel active noise control. Examples illustrate both the equivalence of the algorithms to their original counterparts and the computational gains provided by the new algorithms.

Procedure for reading the data stored in a transponder and a transponder system for the execution of the procedure

Abstract: For reading the data stored in a transponder by means of an interrogation device, the interrogation device at first receives the background noise for the purpose of detecting interference frequencies present in this background noise On the basis of the interference frequencies acquired, coefficients for an adaptive filter are computed by means of which this filter may be tuned in such a way as to suppress the interference frequencies The response signal from the transponder with the superimposed background noise is received by the interrogation device and routed through the adaptive filter which acts to suppress the interference frequencies The signal available at the output of the filter can then be demodulated for the purpose of reading the data stored The transponder system for the execution of the procedure comprises a digital signal processor which computes coefficients for an adaptive filter on the basis of the interference frequencies acquired, and tunes the filter in such a way that the interference frequencies within the RF response signal received from the transponder, carrying the superimposed background noise, are suppressed The output signal from the adaptive filter may then be used for further processing