Finite-duration Impulse Response Filter Research Articles

Design of Low-Power Structural FIR Filter Using Data-Driven Clock Gating and Multibit Flip-Flops

Optimization for power is one of the most important design objectives in modern digital signal processing (DSP) applications. The digital finite duration impulse response (FIR) filter is considered to be one of the most essential components of DSP, and consequently a number of extensive works had been carried out by researchers on the power optimization of the filters. Data-driven clock gating (DDCG) and multibit flip-flops (MBFFs) are two low-power design methods that are used and often treated separately. The combination of these methods into a single algorithm enables further power saving of the FIR filter. The experimental results show that the proposed FIR filter achieves 25% and 22% power consumption reduction compared to that using the conventional design.

Role of Mutation Strategies of Differential Evolution Algorithm in Designing Hardware Efficient Multiplier-less Low-pass FIR Filter

In recent times, system designers are becoming very much apprehensive in reducing the structural complexity of digital systems with which they deal in practice. However, the uncontrolled minimization of any digital hardware always leads to significant deterioration of system performance making it incompatible for use in any practical system. As proper trade-off is inevitably essential between achievable performance and required hardware, researchers have sought a number of artificially intelligent optimization techniques to solve it out. Since such a technique generally involves variety of constructional alternatives, appropriate use of correct option demands justified attention. Numerous evolutionary computation techniques, being a branch of biologically inspired optimization process, are being increasingly used for a number of signal processing applications of late. This paper throws enough light to select the most suitable mutation strategy of Differential Evolution (DE) algorithm for efficient design of multiplier-less low-pass finite duration impulse response (FIR) filter. Computationally efficient mutation scheme has been identified by observing convergence behavior and error histogram plot for different alternatives. Performance of the designed filter has been compared in terms of its magnitude response and the requirement of various hardware blocks for four different lengths of the filter. Consequently the name of the most favorable mutation rule has been suggested upon analyzing all the factors. Finally the supremacy of our proposed design has been established by comparing its performance with that of other state-of-the-art multiplier-less low-pass FIR filters.

A novel approach for coefficient quantization of low-pass finite impulse response filter using differential evolution algorithm

Evolutionary computational techniques have been employed judiciously in various signal processing applications of late. In this paper, such an attempt has been made to design a low-pass linear-phase multiplier-less finite duration impulse response (FIR) filter using differential evolution (DE) algorithm. This particular evolutionary optimization technique has been explored to search the impulse response coefficients of the FIR filter in the form of sum of power of two (SPT) in order to avoid the multipliers during design process. The performance of the designed low-pass filter has been studied thoroughly in terms of its frequency characteristics and primitive requirement of fundamental hardware blocks. The superiority of our design has been ascertained over a number of existing techniques by various means. Finally, the proposed filter of different lengths has been implemented on a field programmable gate array (FPGA) chip for evaluating the competency of this work. The percentage improvement in hardware complexity produced by our design has also been computed and clearly listed in this paper for convenience.

Multi-frequency periodic vibration suppressing in active magnetic bearing-rotor systems via response matching in frequency domain

A method for multi-frequency periodic vibration suppressing in active magnetic bearing (AMB)-rotor systems is proposed, which is based on an adaptive finite-duration impulse response (FIR) filter in time domain. Firstly, the theoretic feasibility of the method is proved. However, two problems would be unavoidable, if the conventional adaptive FIR filter is adopted in practical application. One is that the convergence rate of the different frequency components may be highly disparate in multi-frequency vibration control. The other is that the computational complexity is significantly increased because the long memory FIR filter is required to match the transient response time of the AMB-rotor system. To overcome the problems above, the Fast Block Least Mean Square (FBLMS) algorithm is adopted to efficiently implement the computation in frequency domain at a computational cost far less than that of the conventional FIR filter. By the FBLMS algorithm, regardless of the number of the considered frequency components in vibration disturbance, the computational complexity would be invariable. Moreover, filter’s weights in the FBLMS algorithm have the intuitional relation with signal’s frequency. As a result, the convergence rate of each frequency component can be adjusted by assigning the individual step size parameter for each weight.Experiments with the reciprocating simulating disturbance test and the rotating harmonic vibration test were carried out on an AMB-rigid rotor test rig with a vertical shaft. The experiment results indicate that the proposed method with the FBLMS algorithm can achieve the good effectiveness for suppressing the multi-frequency vibration. The convergence property of each frequency component can be adjusted conveniently. Each harmonic component of the vibration can be addressed, respectively, by reconfiguring the frequency components of the reference input signal.

Design of two-channel perfect-reconstruction linear-phase finite-duration impulse response filter banks for subband image coding by the genetic algorithm approach

In this paper, a powerful approach is proposed for the design of two-dimensional two-channel perfect-reconstruction finite-duration impulse response filter banks employing diamond-shaped linear-phase filters. When one of the analysis filter pair is properly designed, the condition of perfect reconstruction is formulated in a spatial-domain constraint equation for designing the other. The method is based on minimizing the integrated square error for the frequency response over the pass band and stop band while imposing the spatial-domain constraint. One numerical design example is given to demonstrate the perfect reconstruction for the proposed system.

Design of two-channel perfect-reconstruction linear-phase finite-duration impulse response filter banks for subband image coding by the genetic algorithm approach

In this paper, a powerful approach is proposed for the design of two-dimensional two-channel perfect-reconstruction finite-duration impulse response filter banks employing diamond-shaped linear-phase filters. When one of the analysis filter pair is properly designed, the condition of perfect reconstruction is formulated in a spatial-domain constraint equation for designing the other. The method is based on minimizing the integrated square error for the frequency response over the pass band and stop band while imposing the spatial-domain constraint. One numerical design example is given to demonstrate the perfect reconstruction for the proposed system.

Adaptive filtering of continuous GPS results

An adaptive finite-duration impulse response filter, based on a least-mean-square algorithm, has been used to mitigate multipath effects, and to derive tectonic and fault movement signals from continuous global positioning system (CGPS) data. By applying the filter on both pseudo-range and carrier-phase multipath sequences from CGPS observations on consecutive days, multipath models have been reliably derived. The standard deviations of residual time series are reduced to about one-quarter on pseudo-range and to about one-half on carrier phase. The adaptive filter is then used to process baseline solutions from a five-station array. Tectonic and fault movements have been resolved, which are in good agreement with previous studies involving many more CGPS stations.

Multipath Mitigation of Continuous GPS Measurements Using an Adaptive Filter

Though state-of-the-art dual-frequency receivers are employed in the continuous Global Positioning System (CGPS) arrays, the CGPS coordinate time series are typically very noisy due to the effects of atmospheric biases, multipath, receiver noise, and so on, with multipath generally being considered the major noise contributor. An adaptive finite-duration impulse response filter, based on a least-mean-square algorithm, has been developed to derive a relatively noise-free time series from the CGPS results. Furthermore, this algorithm is suitable for real-time applications. Numerical simulation studies indicate that the adaptive filters is a powerful signal decomposer, which can significantly mitigate multipath effects. By applying the filter to both pseudorange and carrier phase multipath sequences derived from some experimental GPS data, multipath models have been reliably derived. It is found that the best multipath mitigation strategy is forward filtering using data on two adjacent days, which reduces the standard deviations of the pseudorange multipath time series to about one fourth its magnitude before correction and to about half in the case of carrier phase. The filter has been successfully applied to the pseudorange multipath sequences derived from CGPS data. The benefit of this techniques is that the affected observable sequences can be corrected, and then these corrected observables can be used to improve the quality of the GPS coordinate results. © 2000 John Wiley & Sons, Inc.

A new algorithm for elimination of common subexpressions

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.

Predictive interpolation and decimation of narrow-band signals

A digital filter design and optimization method for sampling rate alteration of narrow-band signals is presented. The resulting predictive finite-duration impulse-response filters are used to remove the aliasing or imaging frequency components when the sampling rate is changed. The sampling rate can be changed either by an integer factor or a rational factor. A least-squares-based generalization is developed for optimal wide-band noise attenuation.

Biomechanical relationship between center of gravity and center of pressure during standing

The relationship between the position of a body's center of gravity (CG) as determined by a whole body kinematic model and that given by 3 CG position estimation techniques using the ground reaction force center of pressure (CP) has been investigated in this study. The CG and CP are related by the Newtonian mechanics equations of motion. Data required to determine directly the center of gravity position, a key variable in posture and locomotion, are usually not obtained in posture or gait trials; rather, force plate data and center of pressure data are often obtained. Consequently, previous studies have developed estimations of CG position history from CP data. The results of 3 CP-based CG estimations methods are here compared with kinematically determined CG positions in humans. The CP position varies about the CG position and has a higher frequency content than the motion of the CG. This observation, based on the authors' data and mechanics theory, provides the basis for the methods considered in this study. All current methods employ a filtering technique to obtain CG position from CP position time histories during standing trials. In most cases the mean square error is less than 0.1 cm/sup 2/. Finite-duration impulse response filters with periods of 1.0 s to 1.50 s gave the best results when compared with the CG position based on kinematic data. A low-pass filter with cutoff frequency of 0.4 Hz to 0.5 Hz provides the best comparison for this approximation method. The accuracy of the methods diminishes as more dynamics are introduced to the trial. The average mean square differences for walking-in-place trials is from 1 to 10 times greater than that for the standing. CG position estimates from CP data must be interpreted cautiously when nonstatic tasks are monitored, especially in unstable patients.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Optimized windows for FIR filter design to perform maximally flat decimation stages in signal conditioning

New cosine windows, the coefficients of which have been computed through a simple optimization procedure, are proposed to minimize the sidelobe ripples. This allows maximally flat low-pass finite-duration impulse-response (FIR) filter realizations in a multistage-decimation process. In digital measuring devices this strategy is performed in order to limit the amount of data to the storage capabilities of the processor, in conjunction with filtering the data set available after oversampling to remove digitally the noise power produced by quantization. The consequent increase of the number of effective bits improves the dynamic performance of digital instrumentation, especially at lower frequencies.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

The design of low-complexity in linear-phase FIR filter banks using powers-of-two coefficients with an application to subband image coding

An optimization technique is presented for the design of multiplierless two-channel linear-phase finite-duration impulse-response (FIR) filter banks. It is shown to yield filter banks with good filtering performance and nearly perfect signal reconstruction. The design employs filters whose coefficients are represented by a canonic signed-digit (CSD) code. When applied to subband image coding this technique provides an easy way to design low-complexity analysis/synthesis filter banks for high-performance codecs. Examples concerning filter design and the application of such filters to subband image coding are given.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Polyphase FIR filters for the demodulation of chrominance signals

The author presents an estimation scheme and its polyphase finite-duration impulse-response (FIR) filter implementation for the demodulation of composite chrominance signals into color difference components such as in-phase and quadrature-phase components. A decoding scheme with three-tap polyphase FIR filters is presented. The least square error estimators, which estimate the chrominance components, are derived and their performance analyzed. The composite video signal is reviewed and the digitization of the video signal is described. The author formulates the demodulation of the digitized video signal as an estimation problem. A polyphase FIR filter implementation of the resultant estimators for the chrominance components is then described. >

Prefiltering: cascaded stages of decimation-by-two

A very efficient method of reducing the bandwidth of a sampled signal is presented. The method is implemented as a sequence of filtering and decimation stages, where the data rate is halved in each stage. Part of the efficiency is gained through the use of symmetrical linear-phase finite-duration impulse response filters, which have the added benefit that the operation is free of any phase distortion. >

Lagrange multiplier approach to the design of FIR filters for multirate applications

The method of Lagrange multipliers is presented for the design of optimal finite-duration impulse-response (FIR) digital filters for multirate signal processing applications. The filter passbands are constrained to be maximally flat, and the integral of the aliasing error is minimized. The use of these optimality criteria for practical applications is justified. It is shown that the overall frequency response can be represented as a linear combination of individual frequency responses corresponding to the separate design constraints, with the Lagrange multipliers indicating the sensitivity of the integrated aliasing error to changes in the constrained values. The constraints can be modified and the filter can be updated with only a small number of computations. In the case of maximally flat filters it is established that the constraint matrix is of the Vandermonde type, and, using this result, it is shown that a unique, optimal filter always exists. Its coefficients can be determined by solving a linear system of equations.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Subband coder design incorporating recursive quadrature filters and optimum ADPCM coders

This paper presents the results of an analytical and experimental study of the use of IIR quadrature mirror filters and forward adapting ADPCM with optimum quantizers for octave band subband coders for speech. It is Shown that the underlying analysis-synthesis systems can be designed such that there is no interband aliasing distortion in the coded speech and such that the analysis-synthesis transfer function has no frequency distortion or no phase distortion, but not both. The experimental study showed that subband coders based on a mixture of infinite impulse response (IIR) and finite-duration impulse response (FIR) filters resulted in higher quality coding systems using fewer multiplies than for systems based on FIR filters alone. It was also found that a signal-to-noise ratio (SNR) gain of about 4 dB could be obtained using optimum quantizers for forward adapting ADPCM coders over linear quantizers in backward adapting APCM codes, but no perceptual quality gain was observed.

Lunar perturbations in columnar electron content and their interpretation in terms of dynamo electrostatic fields

Both the sun and the moon exert influences on the ionosphere, causing fluctuations in its electron content. The small lunar effects, though not negligible, are difficult to analyze because their periodicities differ little from the periodicity of the dominant solar effects. A finite duration impulse response filter was perfected, permitting the efficient splitting of our columnar electron content data into a solar, a lunar, and a residual component. The solar component plus the lunar component and the solar component alone were processed by a dynamic ionospheric simulation program that yields values of vertical plasma drifts when electron content data are used as input. The difference between the two plasma drifts so obtained was taken as being the plasma drift caused by the electric field generated by the lunar tides in the dynamo region. This technique appears to be the first to allow a direct estimation of the lunar-induced electric fields in the ionosphere.

Statistical design of nonrecursive digital filters

The problem of designing a finite duration impulse response (FIR) digital filter to approximate a desired spectral response is treated in this paper. The philosophy adopted is that for a given FIR filter structure, the filter coefficients can be designed to provide a minimum mean-squared error (MMSE) estimate of a random signal sequence (the design-signal) imbedded in a random noise sequence. By treating the signal and noise covariance functions as design parameters, one can design FIR filters with spectral responses that approximate the power spectral density of the design-signal. For signal processing applications that require some attention to signal fidelity, as well as noise rejection, the MMSE philosophy seems appropriate (as opposed to a maximum signal-to-noise ratio philosophy, for example). Several practical designs are presented that emphasize the simplicity of the design technique and illustrate the selection of design parameters. The designs show quite dramatically that the MMSE design technique can be competitive with existing low-pass and bandpass design techniques. Finally, considerable attention is given to an efficient Toeplitz matrix inversion algorithm that permits rapid inversion of the covariance matrices that arise in the MMSE design. The resulting computation times for the design of high-order filters (N = 128, e.g.) appear to be shorter than computation times for competing algorithms.

Design and simulation of a speech analysis-synthesis system based on short-time Fourier analysis

This paper discusses the theoretical basis for representation of a speech signal by its short-time Fourier transform. The results of the theoretical studies were used to design a speech analysis-synthesis system which was simulated on a general-purpose laboratory digital computer system. The simulation uses the fast Fourier transform in the analysis stage and specially designed finite duration impulse response filters in the synthesis stage. The results of both the theoretical and computational studies lead to an understanding of the effect of several design parameters and elucidate the design tradeoffs necessary to achieve moderate information rate reductions.