• Journal of the Korean Institute of Telematics and Electronics
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• v.14 no.2
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• pp.1-9
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• 1977

The waveform, power spectrum and optimal receiver structure for fast FSK are analysed. It is shown and verified experimentally that for the observation interval of two bit periods, the bit error rate is as good as that of PSK. At the same time, it is shown to have the advantage of narrower band width than PSK. In experiment, the FFSK signal is generated by using a binary transversal filter and a low pass filter.

• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.185-185
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• 2000

This paper designs an interference mitigation filter for GPS receiver using a time-domain signal processing techniques (Adaptive Transversal filter). ATF is designed using verilog and simulated using COSSAP for evaluating its performance.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2035-2038
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• 2002

The necessity of the filter coefficients compensation for the LTJ adaptive filter was explained generally and easily by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%

• 제어로봇시스템학회:학술대회논문집
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• 1997.10a
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• pp.1379-1382
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• 1997

We introduce a fast implementation of the adaptive transversal filter which uses least-mean-square(LMS) algorithm. The fast Hadamard transform(FHT) is used for the implementation of the filter. By using the proposed filter we can get the significant time reduction in computatioin over the conventional time domain LMS filter at the cost of a little performance. By computer simulation, we show the comparison of the propsed Hadamard-domain filter and the time domain filter in the view of multiplication time, mean-square error and robustness for noise.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.22-33
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• 2005

In this paper, an efficient signal interference control technique to improve the high convergence speed of LMS algorithms is introduced in the adaptive transversal filter of DS/SS. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, is analyzed to prove theoretically the improvement of high convergence speed. According as the step-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Also, an increase in the stop-size parameter ${\mu}$ has the effect of reducing the variation in the experimentally computed learning curve. Increasing the eigenvalue spread has the effect of controlling which is downed the rate of convergence of the adaptive equalizer. Increasing the steady-state value of the average squared error, proposed algorithm also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.260-263
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• 2000

A technique that uses linear prediction to achieve noise reduction in a voice signal which has been mixed with an ambient noise (Signal to Noise (S-N) ratio = about 0dB) is proposed. This noise reduction method which is based on the linear prediction estimates the voice spectrum while ignoring the spectrum of the noise. The performance of the noise reduction method is first examined using the transversal linear predictor filter. However, with this method there is deterioration in the tone quality of the predicted voice due to the low level of the S-N ratio. An additional processing circuit is then proposed so as to adjust the noise reduction circuit with an aim of improving the problem of tone deterioration. Next, we consider a practical application where the effects of round on errors arising from fixed-point computation has to be minimized. This minimization is achieved by using the lattice predictor filter which in comparison to the transversal type, is Down to be less sensitive to the round-off error associated with finite word length operations. Finally, we consider a practical application where noise reduction is necessary. In this noise reduction method, both the voice spectrum and the actual noise spectrum are estimated. Noise reduction is achieved by using the linear predictor filter which includes the control of the predictor filter coefficient’s update.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.4
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• pp.996-1007
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• 1996

In this paper, a blind adaptation method for a decision feedback equalizer (DFE) is proposed to deal with nominimum phase channels. This equalizer is composed of a linear transversal filter and a prediction error filter which are trained separately using constant modulus and decision feedback prediction algorithms, respectively, during the learnign time. The proposed algorithm guaranetees the DFE to converge to a suboptimal point on the condition that a linear transversal of the proposed scheme is illustrated and the performance is compared with conventional blind equlization algorithms.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1983.10a
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• pp.85-87
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• 1983

In this paper, a improved whitening filter for narrowband interference rejetion in PN spread spectrum communication systems is proposed. This whitening filter has the feedback from the output to the input of adaptive transversal filter and is superior to conventional LMS whitening filter in respect to SNR and power spectrum since the calculation of tap weight is based on maximum likelihood estimation

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.10B
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• pp.1707-1713
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• 2000

We have proposed a tunable optical ADM using a fiber-optic transversal filter which is composed of fiber couplers and metal-film coated fiber-optic tapped delay-lines with a flat spectral response in abroad range of wavelength. Simulation results show that the optical loss at the DROP and PASS wavelengths of the OADM is negligible and the wavelength tunability is 0.78${\mu}{\textrm}{m}$/ps for the unit time delay of 2 ps. In order to investigate the effects of wavelength drift of the imput optical signal on the OADM the loss at the DROP port and the crosstalks to the other ports have been calculated. The maximum bit rates have been calculated at 46.26 Gb/s for the input Gaussian pulse width of 10 ps.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.10 no.2
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• pp.82-87
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• 1985

The performance comparison method against ground clutter, when the trasnfer function of MTI filter or integrator is given, is considered for the MTI signal processors using a constant PRF. The MTI signal processors are modelled as the transversal filters, and the ground clutter power density spectrum as Gaussian type and the performance of the MTI signal processors are compared by calculating the MTI improvent factors. The MTI imrpovement factors versus normalized spectral width is depicted as examples for the MTI filters, the integrator using Hanning weighting function and the cascading of the two.