• The Transactions of The Korean Institute of Electrical Engineers
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• v.63 no.4
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• pp.551-558
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• 2014

Baseline drift reduction and removal of power line noises in electrocardiogram are often necessary to avoid the distortions in extracting the fiducial features. With this aim, the multirate digital filtering algorithm is suggested to design and implement Finite Impulse Response or Infinite Impulse Response Filter by changing the sampling rate with omitting or interpolating intermediate ECG data. After the experimental simulations performed, we can conclude the fact that we can suppress the baseline wander and power line disturbances in ECG signal with reducing the computational complexities in which we do not keep the original and high sampling frequency.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.310-312
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• 2011

In complex noise environment, digital filter is being used to obtain, transport and storage original voice or image signal. Digital filter can be largely separated FIR(Finite duration impulse response) filter and IIR(Infinite duration impulse response) filter. Among FIR filter, window function has characteristic of linear phase and as can be easily set pass-band frequency, cutoff frequency and so on. In this paper, We compared with established method using transient characteristic and peak side-lobe in order to check filter characteristics after we designed the existing variants of the window function.

• ETRI Journal
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• v.34 no.3
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• pp.379-387
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• 2012

In this paper, an error compensation technique for a dead reckoning (DR) system using a magnetic compass module is proposed. The magnetic compass-based azimuth may include a bias that varies with location due to the surrounding magnetic sources. In this paper, the DR system is integrated with a Global Positioning System (GPS) receiver using a finite impulse response (FIR) filter to reduce errors. This filter can estimate the varying bias more effectively than the conventional Kalman filter, which has an infinite impulse response structure. Moreover, the conventional receding horizon Kalman FIR (RHKF) filter is modified for application in nonlinear systems and to compensate the drawbacks of the RHKF filter. The modified RHKF filter is a novel RHKF filter scheme for nonlinear dynamics. The inverse covariance form of the linearized Kalman filter is combined with a receding horizon FIR strategy. This filter is then combined with an extended Kalman filter to enhance the convergence characteristics of the FIR filter. Also, the receding interval is extended to reduce the computational burden. The performance of the proposed DR/GPS integrated system using the modified RHKF filter is evaluated through simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.6A
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• pp.341-350
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• 2003

In this paper, we evaluate the performance of a moving average (MA) channel estimation filter when space-time block coding transmit diversity (STBC-TD) is applied to the wideband direct sequence code division multiple access (WCDMA) forward link. And we present the infinite impulse response (IIR) filter scheme that can reduce the required memory buffer and the channel estimation delay time. This paper also compares the performance between MA filter scheme and IIR filter scheme in various Rayleigh fading channel environments through the bit error rate (BER) and the frame error rate (FER). Extensive computer simulation results show that transmission with STBC-TD provides a significant gain in performance over no transmit diversity technique, particularly at pedestrian speeds. If STBC-TD technique is employed in the channel estimator based on MA filter, it provides considerable performance gains against Rayleigh fading and reduces the optimum filter tap number. Consequently, the channel estimation delay time and the complexity of the receiver are reduced. In addition, the channel estimator based on IIR filter has the advantages such as little memory requirement and no delay time compared to the MA scheme. However, IIR filter coefficients is very sensitive to the mobile speed change and it exerts a serious influence upon the performance. For that reason, it is important to set uP the optimum IIR filter coefficients.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1E
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• pp.15-23
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• 1997

Zero-Crossings with Peak Amplitudes(ZCPA) model motivated by human auditory periphery was proposed to extract reliable features speech signals even in noisy environments for robust speech recognition. In this paper, some practical considerations for digital hardware implementations of the ZCPA model are addressed and evaluated for recognition of speech corrupted by several real world noises as well as white Gaussian noise. Infinite impulse response(IIR) filters which constitute the cochliar filterbank of the ZCPA are replaced by hamming bandpass filters of which frequency responses are less similar to biological neural tuning curves. Experimental results demonstrate that the detailed frequency response of the cochlear filters are not critical to performance. Also, the sensitivity of the model output to the variations in microphone gain is investigated, and results in good reliability of the ZCPA model.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.3
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• pp.145-155
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• 1990

In this paper, an adaptive Infinite Impulse Response (IIR) filter algorithm using output error method, which prevents poles of a system transfer function from being out of unit circle, is proposed, and it is proved that the proposed algorithm always satisfies hyperstability. The proposed algorithm is applied to an Adaptive Noise Canceller (ANC), and compared with a Least Square (LS) method adaptive IIR filter algorithm and an adaptive Finite Inpulse Response (FIR) filter algorithm. As a result, the validity of the proposed algorithm is proved.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.955-959
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• 2003

A technique for realizing linear phase IIR filters has been proposed by Powell-Chau which gives a real-time implementation of H(z-1).H(z), where H(z) is a causal nonlinear phase IIR filter. Powell-Chau system is linear but not timeinvariant system. Therefore, that system has group delay response that exhibits a minor sinusoidal variation superimposed on a constant value. In the signal processing, this oscillation seriously degrade the signal quality. Unfortunately, that system has a large sample delay of 4L and also more computational complexity. Proposed system is present a reduced computational complexity technique by moved the numerator polynomial of H(1/z) out to cascade with causal filter H(z) and remain only all-pole of H(1/z), then applied truncated infinite impulse response to finite with truncated IIR filtel $H_L$(z) and L sample delay to subtract the output sequence from the top and bottom filter. Proposed system is linear time invariance and group delay response and total harmonic distortion are also improved.

• KSCE Journal of Civil and Environmental Engineering Research
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• v.12 no.4
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• pp.71-79
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• 1992

In general, the accelerogram of earthquake ground motion or the accelerogram obtained from dynamic tests contain various errors. In these errors of the accelerograms, there are instrumental errors(magnitude and phase distortion) due to the response characteristics of accelerometer and the digitizing error concentrated in low and high frequency components and random errors. Then, these errors may be detrimental to the results of data processing and dynamic analysis. An efficient method which can correct the errors of the accelerogram is proposed in this study. The correction of errors can be accomplished through four steps as followes ; 1) using an interpolation method a data form appropriate to the error correction is prepared, 2) low and high frequency errors of the accelerogram are removed by band-pass filter between prescribed frequency limits, 3) instrumental errors are corrected using dynamic equilibrium equation of the accelerometer, 4) velocity and displacement are obtained by integrating corrected accelerogram. Presently, infinite impulse response(IIR) filter and finite impulse response (FIR) filter are generally used as band-pass filter. In the proposed error correction procedure, the deficiencies of FIR filter and IIR filter are reduced and, using the properties of the differentiation and the integration of Fourier transform, the accuracy of instrument correction and integration is improved.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.351-354
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• 2007

In recent years, digital processing techniques have been applied diversity of fields. Typical signal processing techniques are speech processing and image processing. And filters for the signal processing can be divided in FIR (finite impulse response) filter and IIR (infinite impulse response) filter. Compared with IIR filter, the FIR Filter has a defect of high-degree, but has a merit of stability and uses simply. Futhermore, FIR filter also has linear phase response characteristics, it is using in fields regarding wave information importantly. To FIR Filter design, the main issue is to remove the Gibbs phenomenon. Therefore, in this paper I was proposed a method using FIR digital filter applied a modified window function and the method was compared with conventional methods.

• ETRI Journal
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• v.35 no.4
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• pp.697-705
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• 2013

Emerging trends in the area of digital very large scale integration (VLSI) signal processing can lead to a reduction in the cost of the cochlear implant. Digital signal processing algorithms are repetitively used in speech processors for filtering and encoding operations. The critical paths in these algorithms limit the performance of the speech processors. These algorithms must be transformed to accommodate processors designed to be high speed and have less area and low power. This can be realized by basing the design of the auditory filter banks for the processors on digital VLSI signal processing concepts. By applying a folding algorithm to the second-order digital gammatone filter (GTF), the number of multipliers is reduced from five to one and the number of adders is reduced from three to one, without changing the characteristics of the filter. Folded second-order filter sections are cascaded with three similar structures to realize the eighth-order digital GTF whose response is a close match to the human cochlea response. The silicon area is reduced from twenty to four multipliers and from twelve to four adders by using the folding architecture.