• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1917-1919
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• 2002

An adaptive algorithm for the acoustic echo canceller is presented. This paper proposes a modified LMS algorithm for the adaptive filter and applys the algorithm to he acoustic echo canceller, An objective of the proposed algorithm is to reduce the hardware complexity. In order to est the performances, a model of the echo path is established, and a program is described. The impulse reponses of the echo path have the length of 125msec or ore, and then the FIR filter with 1000 taps is required. he results from simulations show that the acoustic echo canceller adopting the proposed algorithm achieves the ERLE of 25dB or more within 1sec. If an echo canceller is implemented with this algorithm, its computation quantity s reduced to two times less than the one that is implemented with the normal LMS algorithm, without the degradation of performances.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.46-49
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• 2000

An iterative deblocking algorithm for DCT-compressed images using two-dimensional FIR filters adapted for local directionality of each block, is proposed. First, we introduce a set of simple lowpass filters, which are adapted for edges of different angles. In conventional deblocking methods based on lowpass-filtering and convex projections, a single filter is applied to a whole image. In the proposed method, on the other hand, a suitable filter is chosen out of the directional filters designed previously in every subimage (typically $8{\times}8$ block). Experimental results indicate that adaptive filtering improves PSNR at each iteration.

• Proceedings of the KIEE Conference
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• 2007.10a
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• pp.433-434
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• 2007

In this paper, active control of noise in a HVAC duct is considered. Most adaptive control filters have used FIR structures based on filtered-x LMS algorithms. But, the IIR structures are more desirable for the active control of duct noise in order to remove the poles introduced by the acoustic feedback and presented an algorithm to adjust the coefficients of an IIR filter using the recursive least mean square (RLMS) algorithm. A smoothed LMS algorithm is proposed to improve a convergent speed of filter parameters when the noise is wide band and power of input is time varying. And computer simulations have performed to show the effectiveness of the proposed algorithm.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.36 no.11
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• pp.819-825
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• 1987

The ALE problem, which tries to recover a sinusoidal signal corrupted by noise, has been solved using FIR filters. Recently several methods have been proposed using a norch filter of IIR type. In this study, the notch filter was represented with a parameter and auxiliary signals were generated by using an adaptive observer. A simple method is proposed to estimate the parameter. This method is tested under various circumstances by changing the input frequency, S/N ratio, and the type of the noise. The simulation shows that this method gives much better results than the other known methods with respect to the input S/N ratio and converging times. This method is simple and does not require much conputation, so it can be easily implemented in real time applications.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.5
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• pp.70-75
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• 2014

Super-resolution near-field structure (super-RENS) read-out samples are affected by a nonlinear and noncausal channel, which results in inter-symbol interference (ISI). In this study, we investigate asymmetry or domain bloom in super-RENS in terms of equalization. Domain bloom is caused by writing process in optical recording. We assume in this work that the asymmetry symbol conversion scheme is to generate asymmetric symbols, and then a linear finite impulse response filter can model the read-out channel. For equalizing this overall nonlinear channel, the read-out signals are deconvolved with the finite impulse response filter and its output is decided based on the decision rule table that is developed from the asymmetry symbol conversion scheme. The proposed equalizer is investigated with the simulations and the real super-RENS samples in terms of raw bit error rate.

• Journal of the Korea Society of Computer and Information
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• v.10 no.5 s.37
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• pp.313-322
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• 2005

In this paper, we implement and verify performance monitor for parallel signal processing system, using DSP Starter Kit(DSK) of which the basic Processor is TMS302C6711 chip. The key ideas of this performance monitor is, using Real Time Data Exchange(RTDX) for the Purpose of real-time data transfer and function of DSP/BIOS, the ability to measure the Performance measure like DSP workload, memory usage, and bridge traffic. In the simulation, FFT, 2D FFT, Matrix Multiplication, and Fir Filter, which are widely used DSP algorithms, have been employed. Using performance monitor and Code Composer Studio from Texas Instrument(Tl) , the result has been recorded according to different frequencies, data sizes, and buffer sizes for a single wave file. The accuracy of our performance monitor has been verified by comparing those recorded results.

• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.389-396
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• 1997

Several types of electrical artifact seen on electroencephalogram( EEG) records are described. Those are the EOG and the PVC roller pump noise, and so on. An adaptive digital filtering of the electroencephalogram( EEG) is a successful way of suppressing mains interference, but it affects some of the frequency components of the signal, whore artifacts may not be acceptable in some cafes of automatic EEG processing. Thus we studied the method for cancelling these artifacts. This proposed method does not use the reference channel, and is realized by connecting the linear predictor and the fixed FIR filter for the EOG artifact, and by cascading the linear predictor and the noise canceller for the pump artifact. The simulation results illustrate the performances of the proposed method in terms of the capability of interferences suppression. In the results we obtained about 20 dB noise reduction.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.11 no.1
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• pp.18-25
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• 2000

In this paper, we have designed a microwave level meter based on the FM-CW radar theory using a YIG-tuned oscillator (YTO). YTO has an excellent frequency linearity, so a linearizer circuit is not necessary for the level meter. It is shown that interference signals reflected from nearby obstacles can be removed by using a digital band-pass filter. An FIR band-pass filter is designed using the Kaiser window. The distance measurement has been performed in the outdoor test field. The measurement data have been obtained for the range of 1~40m with 1m step, and the results show that the standard deviation of the measured data is 2.33 cm. The level meter manufactured in this study can be applied usefully in the industrial facilities which are not accessible easily, for example, to measure the level of molten metal in the iron and steel company.

• JSTS:Journal of Semiconductor Technology and Science
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• v.16 no.4
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• pp.425-435
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• 2016

This paper describes a spread-spectrum clock generation method by utilizing a ${\Delta}{\Sigma}$ digital PLL (DPLL) which is solely based on binary phase detection and does not require a linear time-to-digital converter (TDC) or other linear digital-to-time converter (DTC) circuitry. A 1-bit high-order ${\Delta}{\Sigma}$ modulator and a hybrid finite-impulse response (FIR) filter are employed to mitigate the phase-folding problem caused by the nonlinearity of the bang-bang phase detector (BBPD). The ${\Delta}{\Sigma}$ DPLL employs a two-point modulation technique to further enhance linearity at the turning point of a triangular modulation profile. We also show that the two-point modulation is useful for the BBPLL to improve the spread-spectrum performance by suppressing the frequency deviation at the input of the BBPD, thus reducing the peak phase deviation. Based on the proposed architecture, a 3.2 GHz spread-spectrum clock generator (SSCG) is implemented in 65 nm CMOS. Experimental results show that the proposed SSCG achieves peak power reductions of 18.5 dB and 11 dB with 10 kHz and 100 kHz resolution bandwidths respectively, consuming 6.34 mW from a 1 V supply.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.64-73
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• 1995

We developed a surface acoustic wave (SAW) computer aided design (CAD) for mobile communication using Kaier window function. The systems are composed of modules for designing apodization weighted IDT-uniform IDT, withdrawal weighted IDT-withdrawal weighted IDT, and resonator type. The design of SAW bandpass with center frequencies from 222MHz to 343MHz were simulated by the developed CAD system. Although the method proposed in this paper is formulated primarily for SAW filters, it is equally applicable to finite impulse response (FIR) digital filter design.