Showing papers on "Filter design published in 2002"

On the origin of the bilateral filter and ways to improve it

Abstract: Additive noise removal from a given signal is an important problem in signal processing. Among the most appealing aspects of this field are the ability to refer it to a well-established theory, and the fact that the proposed algorithms in this field are efficient and practical. Adaptive methods based on anisotropic diffusion (AD), weighted least squares (WLS), and robust estimation (RE) were proposed as iterative locally adaptive machines for noise removal. Tomasi and Manduchi (see Proc. 6th Int. Conf. Computer Vision, New Delhi, India, p.839-46, 1998) proposed an alternative noniterative bilateral filter for removing noise from images. This filter was shown to give similar and possibly better results to the ones obtained by iterative approaches. However, the bilateral filter was proposed as an intuitive tool without theoretical connection to the classical approaches. We propose such a bridge, and show that the bilateral filter also emerges from the Bayesian approach, as a single iteration of some well-known iterative algorithm. Based on this observation, we also show how the bilateral filter can be improved and extended to treat more general reconstruction problems.

A novel Kalman filter for frequency estimation of distorted signals in power systems

Abstract: A simple and novel approach in the design of an extended Kalman filter (EKF) for the measurement of power system frequency has been presented in this paper. The design principles and the validity of the model have been outlined. The performance of this filter has been compared with some of the existing methods for estimating the frequency of a signal under noisy conditions. The feasibility of the proposed filter has been tested in the laboratory under worst-case measurement and network conditions, which might occur in a typical power system. Also, the proof of the stability for the proposed filter has been discussed for a single sinusoid. It has been found that the proposed algorithm is suitable for real-time applications especially when the frequency changes are abrupt and the signal is corrupted with noise and other disturbances due to harmonics.

Block iterative DFE for single carrier modulation

Abstract: An iterative block decision feedback equaliser (IB-DFE) for single carrier modulation is proposed. Filtering operations are implemented by discrete Fourier transforms (DFTs) which yield a reduced computational complexity, for both filter design and signal processing, when compared to existing DFEs. Moreover, the new IB-DFE operates on blocks of the receive signal, thus allowing the use of error correction codes on the feedback data signal.

Clutter filter design for ultrasound color flow imaging

Abstract: For ultrasound color flow images with high quality, it is important to suppress the clutter signals originating from stationary and slowly moving tissue sufficiently. Without sufficient clutter rejection, low velocity blood flow cannot be measured, and estimates of higher velocities will have a large bias. The small number of samples available (8 to 16) makes clutter filtering in color flow imaging a challenging problem. In this paper, we review and analyze three classes of filters: finite impulse response (FIR), infinite impulse response (IIR), and regression filters. The quality of the filters was assessed based on the frequency response, as well as on the bias and variance of a mean blood velocity estimator using an autocorrelation technique. For FIR filters, the frequency response was improved by allowing a non-linear phase response. By estimating the mean blood flow velocity from two vectors filtered in the forward and backward direction, respectively, the standard deviation was significantly lower with a minimum phase filter than with a linear phase filter. For IIR filters applied to short signals, the transient part of the output signal is important. We analyzed zero, step, and projection initialization, and found that projection initialization gave the best filters. For regression filters, polynomial basis functions provide effective clutter suppression. The best filters from each of the three classes gave comparable bias and variance of the mean blood velocity estimates. However, polynomial regression filters and projection-initialized IIR filters had a slightly better frequency response than could be obtained with FIR filters.

Transmit Wiener filter for the downlink of TDDDS-CDMA systems

Abstract: We derive the transmit Wiener filter for DS-CDMA systems which depends upon the noise power at the receivers. We show that the transmit Wiener filter converges to the transmit matched filter and the transmit zero-forcing filter for low and high signal-to-noise-ratio, respectively. Simulation results show the superiority of the transmit Wiener filter compared to the other two transmit filters. Moreover, we observe that the application of the three transmit filters and the respective receive filters lead to similar results.

Reconstruction of nonuniformly sampled bandlimited signals by means of digital fractional delay filters

Abstract: This paper considers the problem of reconstructing a class of nonuniformly sampled bandlimited signals of which a special case occurs in, e.g., time-interleaved analog-to-digital converter (ADC) systems due to time-skew errors. To this end, we propose a synthesis system composed or digital fractional delay filters. The overall system (i.e., nonuniform sampling and the proposed synthesis system) can be viewed as a generalization of time-interleaved ADC systems to which the former reduces as a special case. Compared with existing reconstruction techniques, our method has major advantages from an implementation point of view. To be precise, (1) we can perform the reconstruction as well as desired (in a certain sense) by properly designing the digital fractional delay filters, and (2) if properly implemented, the fractional delay filters need not be redesigned in case the time skews are changed. The price to pay for these attractive features is that we need to use a slight oversampling. It should be stressed, however, that the oversampling factor is less than two as compared with the Nyquist rate. The paper includes error and quantization noise analysis. The former is useful in the analysis of the quantization noise and when designing practical fractional delay filters approximating the ideal filters.

Performance of a planar filter using a 0/spl deg/ feed structure

Abstract: The advantage of using a 0/spl deg/ feed structure in filter design is that two extra transmission zeros are created in the stopband while the passband response remains unchanged. This feed structure is analyzed by using transmission matrices. A new lumped-circuit model for a coupled resonator filter is then proposed to take into account the effects of this feed structure. Finally, the feed structure is applied to the design of a cross-coupled filter. All the theoretical analysis and design procedures have been successfully verified by experiment results.

Analog and digital filter design

Abstract: Introduction Time And Frequency Response Poles and Zeroes Analog Lowpass Filters Highpass Filters Bandpass Filters Bandstop Filters Impedance Matching Networks Phase Shift Networks (All-Pass Filters) Selecting Components for Analog Filters Filter Design Software Transmission Lines and Printed Circuit Boards as Filters Filters For Phase Locked Loops Filter Integrated Circuits Introduction to Digital Filters Digital 'Fir' Filter Design IIR Filter Design Design Equations

Algorithms for Statistical Signal Processing

Abstract: 1. Introduction. Characterization of Signals. Characterization of Linear Time-Invariant Systems. Sampling of Signals. Linear Filtering Methods Based on the DFT. The Cepstrum. Summary and References. Problems. 2. Algorithms for Convolution and DFT. Modulo Polynomials. Circular Convolution as Polynomial Multiplication mod un- 1. A Continued Fraction of Polynomials. Chinese Remainder Theorem for Polynomials. Algorithms for Short Circular Convolutions. How We Count Multiplications. Cyclotomic Polynomials. Elementary Number Theory. Convolution Length and Dimension. The DFT as a Circular Convolution. Winograd's DFT Algorithm. Number-Theoretic Analogy of DFT. Number-Theoretic Transform. Split-Radix FFT. Autogen Technique. Summary and References. Problems. 3. Linear Prediction and Optimum Linear Filters. Innovations Representation of a Stationary Random Process. Forward and Backward Linear Prediction. Solution of the Normal Equations. Properties of the Linear Prediction-Error Filters. AR Lattice and ARMA Lattice-Ladder Filters. Wiener Filters for Filtering and Prediction. Summary and References. Problems. 4. Least-Squares Methods for System Modeling and Filter Design. System Modeling and Identification. Lease-Squares Filter Design for Prediction and Deconvolution. Solution of Least-Squares Estimation Problems. Summary and References. Problems. 5. Adaptive Filters. Applications of Adaptive Filters. Adaptive Direct-Form FIR Filters. Adaptive Lattice-Ladder Filters. Summary and References. Problems. 6. Recursive Least-Squares Algorithms for Array Signal Processing. QR Decomposition for Least-Squares Estimation. Gram-Schmidt Orthogonalization for Least-Squares Estimation. Givens Algorithm for Time-Recursive Least-Squares Estimation. Recursive Least-Squares Estimation Based on the Householder Transformation. Order-Recursive Least-Squares Estimation Algorithms. Summary and References. Problems. 7. QRD-Based Fast Adaptive Filter Algorithms. Background. QRD Lattice. Multichannel Lattice. Fast QR Algorithm. Multichannel Fast QR Algorithm. Summary and References. Problems. 8. Power Spectrum Estimation. Estimation of Spectra from Finite-Duration Observations of Signals. Nonparametric Methods for Power Spectrum Estimation. Parametric Methods for Power Spectrum Estimation. Minimum-Variance Spectral Estimation. Eigenanalysis Algorithms for Spectrum Estimation. Summary and References. Problems. 9. Signal Analysis with Higher-Order Spectra. Use of Higher-Order Spectra in Signal Processing. Definition and Properties of Higher-Order Spectra. Conventional Estimators for Higher-Order Spectra. Parametric Methods for Higher-Order Spectrum Estimation. Cepstra of Higher-Order Spectra. Phase and Magnitude Retrieval from the Bispectrum. Summary and References. Problems. References. Index.

Broadband and compact multi-pole microstrip bandpass filters using ground plane aperture technique

Abstract: A ground plane aperture technique is developed for effective enhancement of the capacitive coupling factor in a parallel-coupled microstrip line (PCML). By applying a so-called 'short-open calibration' (SOC) scheme in the fullwave method of moments (MoM) algorithm, this PCML with two external lines is characterised by an equivalent J-inverter network with its susceptance and two electrical line lengths. Extracted parameters indicate that the coupling factor appears to be frequency-dependent and its maximum value rises rapidly as the aperture is widened. With the introduction of a single microstrip line section between two identical PCMLs, a broadband and compact multi-pole microstrip bandpass filter is proposed for the first time, and its electrical behaviour is studied and optimised on the basis of its equivalent circuit network. The network-based optimised results are confirmed by an EM simulation of the entire filter layout, featuring ultra-broadband and four-pole bandpass behaviour. Further, a single capacitively loaded line section is utilised to formulate a multi-pole bandpass filter, and its electrical effects are also discussed for filter design. The predicted and measured results confirm attractive properties of the proposed multi-pole filter with BW=60%. |S/sub 11/|<-16 dB and 220% wide upper stop-band.

A CMOS highly linear channel-select filter for 3G multistandard integrated wireless receivers

Abstract: A new approach for designing digitally programmable CMOS integrated baseband filters is presented. The proposed technique provides a systematic method for designing filters exhibiting high linearity and low power. A sixth-order Butterworth low-pass filter with 14-bit bandwidth tuning range is designed for implementing the baseband channel-select filter in an integrated multistandard wireless receiver. The filter consumes a current of 2.25 mA from a 2.7-V supply and occupies an area of 1.25 mm/sup 2/ in a 0.5-/spl mu/m chip. The proposed filter design achieves high spurious free dynamic ranges (SFDRs) of 92 dB for PDC (IS-54), 89 dB for GSM, 84 dB for IS-95, and 80 dB for WCDMA.

Adaptive interpolation filter system for motion compensated predictive video coding

Abstract: An adaptive interpolation filter system for searching to obtain an optimized interpolation filter that minimizes prediction error in a video codec includes an interpolation module and a discrete search space. A plurality of discrete interpolation filters are positioned in a three dimensional grid within the search space. The interpolation module may select a current minimum filter. Based on the current minimum filter, a search region within the search space that includes a plurality of candidate filters located adjacent to the current minimum filter may be identified. The interpolation module may interpolate a reference image signal with each of the candidate filters. The candidate filter resulting in the smallest prediction error may be identified as the current minimum filter and the search repeated until the prediction error is minimized.

A unified approach for the design of 2-D digital filters via semidefinite programming

Abstract: This paper attempts to demonstrate that a modem optimization methodology known as semidefinite programming (SDP) can serve as the algorithmic core of a unified design tool for a variety of two-dimensional (2-D) digital filters Representative SDP-based designs presented in the paper include minimax and weighted least-squares designs of FIR filters with continuous and discrete coefficients, and minimax design of stable separable-denominator IIR filters Our studies are motivated by the fact that SDP as a subclass of convex programming can be solved efficiently using recently developed interior-point methods and, more importantly, constraints on amplitude/phase responses in certain frequency regions and on stability (for IIR filters), that are often encountered in many filter design problems, can be formulated in a natural way as linear matrix inequalities (LMI) which allow SDP to apply Design examples for each class of filters are included to demonstrate that SDP-based methods can in many cases be useful in producing optimal or near-optimal 2-D filters with reduced computational complexity

Road Slope and Vehicle Mass Estimation Using Kalman Filtering

Abstract: SUMMARYKalman filtering is used as a powerful method to obtain accurate estimation of vehicle mass and road slope. First the problem of estimating the slope when the vehicle mass is known is studied using two different sensor configurations. One where speed is measured and one where both speed and specific-force is measured. A filter design principle is derived guaranteeing the estimation error under a worst case situation (when assuming first order dynamics). The simultaneous estimation problem required an Extended Kalman Filter (EKF) design when measuring speed only whereas the additional specific force ease yielded a simple filter structure with a time-variant measurement equation. Additionally the filter needs present propulsion force which in our case is calculated form the engine speed and amount of fuel injected. When the vehicle uses the foundation brakes the estimates are frozen since varying friction properties makes the braking force unknown. Both sensor configurations are concluded to be robus...

Novel coupling schemes for microwave resonator filters

Abstract: This paper introduces novel coupling schemes for microwave resonator filters. It is shown that higher order filter characteristics can be obtained from lower order sections, which are connected in parallel between the source and load, by proper superposition of the individual lower order responses. This property can be used in modular filter design by focusing on separate sections of the filter one at a time. In addition, some of these coupling schemes exhibit zero-shifting properties, whereby transmission zeros can be shifted from one side of the passband to the other by simply changing the resonant frequencies of the resonators while keeping all the coupling coefficients unchanged. Several novel filter designs of different kinds (microstrip, planar waveguide cavity, and dual-mode types) are introduced to prove the new method and to give an idea of the extended design possibilities. Good agreement between measured, computed, and synthesized results is demonstrated.

Ultra-selective 22-pole 10-transmission zero superconducting bandpass filter surpasses 50-pole Chebyshev filter

Abstract: An ultra-selective filter for third-generation (3G) and fourth-generation wireless application is presented. The demonstrated filter consists of 22 resonators and five cross couplings that produce ten transmission zeros. The filter was designed at 1950-MHz center frequency with a 20-MHz bandwidth to meet existing 3G wireless applications. The measured filter data shows excellent selectivity, better than 30-dB/100-kHz skirt slopes, and 90 dB of rejection at 350 kHz from the band edge. This filter performance surpasses the performance of a 50-pole Chebyshev filter. In order to fit a large number of resonators into a limited wafer area, a new compact resonator was developed. The filter was fabricated using a YBCO thin film on a 2-in MgO wafer.

An efficient and simple method for designing prototype filters for cosine-modulated pseudo-QMF banks

Abstract: We present a new method to design prototype filters for conventional cosine-modulated pseudo-quadrature mirror filter (QMF) banks. This method is based on windowing, and sets the 3-dB cutoff frequency of the filter obtained at /spl pi//2M. In this way, the filter bank performance can be significantly improved compared to other existing design methods.

Filtering method for nonlinear systems with constraints

Abstract: A constrained filtering method is proposed to deal with the filtering problems for nonlinear systems with constraints. The problem is converted to a sequence of recursive estimation problems in which the system equations and constraint conditions are treated as pseudo-measurements. To resolve the singularity problem arising from the constraints, a modified maximum-likelihood method for nonlinear systems is developed. The simulation results from the application of the proposed scheme to the target tracking problem shows that the constrained filtering method can enhance the performance of filter design significantly.

Monitoring and control of an adaptive filter in a communication system

Abstract: A communication system having an echo canceller is disclosed. One embodiment of the echo canceller includes an adaptive filter used to provide an estimate of reflected echo which is removed from the send signal. The echo canceller may also include a near-end talker signal detector which may be used to prevent the adaptive filter from adapting when a near-end talker signal is present. The echo canceller may also include a nonlinear processor used to further reduce any residual echo and to preserve background noise. The echo canceller may also include a monitor and control unit which may be used to monitor the filter coefficients and gain of the adaptive filter to maintain stability of the echo canceller, estimate pure delay, detect a tone, and inject a training signal. The echo canceller may also include a nonadaptive filter used to reduce the length of the adaptive filter.

Design and analysis of the inverter-side filter used in the dynamic voltage restorer

Abstract: The role of the inverter-side filter used in the dynamic voltage restorer (DVR) is examined. Using the circuit analysis method, it is shown that the selection of the filter parameters can affect the DVR inverter rating. Furthermore, with the DVR filter-series injection transformer combination, the loading effect of the filter on the restorer and the primary supply system has been quantified under normal as well as voltage sag conditions. A systematic filter design method is then proposed, the primary objective of which is to achieve specific harmonic performance while ensuring that the DVR inverter rating and the loading effect are minimized. Illustrative examples are also included.

Automated Tuning of an Extended Kalman Filter Using the Downhill Simplex Algorithm

Abstract: A method of tuning a Kalman filter by means of the downhill simplex numerical optimization algorithm is presented. The problem is defined by a brief description of the Kalman filter and the extended Kalman filter and the sensitivity of filter performance to process noise and measurement noise covariance matrices Q and R. The filter tuning problem for a system processing simulated data is then formulated as a numerical optimization problem by defining a performance index based on state estimate errors. The resulting performance index is then minimized using the downhill simplex algorithm. The technique is then applied to three numerical examples of increasing complexity to demonstrate its practical utility.

Feedback cancellation in a hearing aid with reduced sensitivity to low-frequency tonal inputs

Abstract: A feedback cancellation system with reduced sensitivity to low-frequency tonal inputs is provided. Such a system can be used, for example, in a hearing aid to prevent cancellation of the desired tonal inputs to the hearing aid, thus improving the gain at high frequencies of the hearing aid while simultaneously preserving the desired tonal inputs at low frequencies. The feedback cancellation system comprises a first adaptive filter block for adaptively filtering an error signal to remove the low-frequency tonal components from the error signal. The first adaptive filter block is constrained so that only low-frequency tones in the error signal are cancelled, thus enabling the feedback cancellation system to still cancel “whistling” at high frequencies due to the temporary instability of the hearing aid. A second adaptive filter block adaptively filters a feedback path signal to produce an adaptively filtered feedback path signal. The first and second adaptive filter blocks are identical and filter coefficients of the first adaptive filter block are copied to those of the second adaptive filter block. Using an LMS adaptation algorithm, filter coefficients of an adaptive filer of the feedback cancellation system are controlled by the adaptively filtered error signal and the adaptively filtered feedback path signal respectively inputted from the first and second adaptive filter blocks. The adaptive filter then produces an adaptively filtered modeled feedback signal to be subtracted from an electrical audio signal input for updating the error signal of the hearing aid. The hearing aid processes the updated error signal with a digital signal processor to generate an audio output.

Impulse response shortening and symbol synchronization in OFDM communication systems

Abstract: A method for receiving at a receiver having a variable filter a transmitted signal that includes a periodic training signal The method includes (a) receiving and sampling the transmitted signal at the receiver to produce a digital complex baseband signal; (b) filtering the digital complex baseband signal with the variable filter; (c) detecting the periodic training signal in the filtered digital complex baseband signal; (d) determining a desired channel impulse response based on the detected periodic training signal; (e) calculating filter coefficients required by the variable filter to achieve the desired channel impulse response; and (f) adjusting the variable filter according to the calculated filter coefficients A receiver for implementing the method is also provided

Method and apparatus for mitigating adjacent channel interference in a wireless communication system

Abstract: Techniques for detecting and mitigating adjacent channel interference (ACI) in a wireless (eg, CDMA) communication system In one aspect, ACI may be determined by signaling or detected by filtering a pre-processed signal in each frequency range where ACI may be present (eg, with a respective bandpass filter), estimating the energy of the filtered signal for each frequency range, comparing the estimated energy against an ACI threshold, and indicating the presence or absence of ACI at each frequency range based on the result of the comparison In another aspect, a selectable filter (eg, a FIR filter) having a number of possible filter responses (eg, provided by a number of sets of filter coefficients) may be used to provide filtering for the pre-processed signal and to reject any detected ACI One of the possible filter responses is selected for use depending on whether and where ACI has been detected

A high-speed, programmable, CSD coefficient FIR filter

Abstract: A new high-speed, programmable FIR filter is presented, which is a multiplierless filter with CSD encoding coefficients. We propose a new programmable CSD encoding structure to make CSD coefficients programmable. Compared with the conventional FIR structure with Booth multipliers, this coding structure improves the speed of filter and decreases the area. We design a 10-bits, 18-taps video luminance filter with the presented filter structure. The completed filter core occupies 6.8 /spl times/ 6.8 mm of silicon area in 0.6 /spl mu/m 2P2M CMOS technology, and its maximum work frequency is 100 MHz.

Mixed-mode EMI noise and its implications to filter design in offline switching power supplies

Abstract: Analysis of the newly discovered mixed-mode EMI noises provides a better understanding of the filter attenuation mechanism. Many practical design issues are investigated from this point of view. Hopefully, the results given in the paper reduce the mystery and the cut-and-try process in filter design in offline power supplies.

Image Filter Design with Evolvable Hardware

Abstract: The paper introduces a new approach to automatic design of image filters for a given type of noise. The approach employs evolvable hardware at simplified functional level and produces circuits that outperform conventional designs. If an image is available both with and without noise, the whole process of filter design can be done automatically, without influence of a designer.