• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1987.04a
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• pp.167-172
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• 1987

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.19 no.4
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• pp.418-426
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• 2008

In this paper, investigations on the impedance characteristics of a DGS(Defected Ground Structure) slot in the groud plane of microstripline are presented in spectral domain and applied to the characteristic improvement of stepped impedance microstrip low pass filter(LPF). In this method, expressions for the impedance of a DGS slot are derived from self-reaction of the angular spectrum of plane waves and the discontinuity in the modal voltage. The numerical results are compared with those of the rigorous full-wave method and are shown to produce reasonably accurate data. And the stepped impedance microstrip low pass filter is designed and fabricated with the uniform and nonuniform DGS slots for improving the frequency responses. The experiments show that the proposed filter with slots in the ground plane has a wider stopband and sharper cutoff response.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.1
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• pp.35-44
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• 2004

We Propose the methods to design the finite impulse response (FIR) and the infinite impulse response (IIR) lattice filters using Schur algorithm through the spectral factorization of the covariance matrix by circulant matrix factorization (CMF). Circulant matrix factorization is also very powerful tool used for spectral factorization of the covariance polynomial in matrix domain to obtain the minimum phase polynomial without the polynomial root finding problem. Schur algorithm is the method for a fast Cholesky factorization of Toeplitz matrix, which easily determines the lattice filter parameters. Examples for the case of the FIR filter and for the case of the In filter are included, and performance of our method check by comparing of our method and another methods (polynomial root finding and cepstral deconvolution).

• Korean Journal of Optics and Photonics
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• v.18 no.1
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• pp.79-83
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• 2007

A reconfigurable photonic microwave (MW) channel selective filter was demonstrated incorporating a $1{\times}2$ switch based on two tunable polymeric resonators with different free spectral ranges. Each resonator, consisting of two cascaded rings with an electrode formed on one of them, plays a role as an on/off switch through the thermooptic effect. The optical signal carrying the MW signal is routed to either port of the switch and detected to show the filtered output at the frequency determined by the free spectral range of the corresponding resonator. When the channel centered at 10 GHz was chosen, the extinction ratio was ${\sim}30dB$, the bandwidth 1 GHz, and the electrical power consumption 4.1 mW. For the other channel located at 20 GHz, we have achieved the extinction ratio of ${\sim}30dB$, the bandwidth of 2 GHz, and the required power of 8.0 mW. Finally the crosstalk between the selected and blocked channels was higher than 24 dB.

• International Journal of Control, Automation, and Systems
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• v.6 no.3
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• pp.378-385
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• 2008

This paper aims to present a polynomial approach to the steady-state optimal filtering for delayed systems. The design of the steady-state filter involves solving one polynomial equation and one spectral factorization. The key problem in this paper is the derivation of spectral factorization for systems with delayed measurement, which is more difficult than the standard systems without delays. To get the spectral factorization, we apply the reorganized innovation approach. The calculation of spectral factorization comes down to two Riccati equations with the same dimension as the original systems.

• ETRI Journal
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• v.35 no.3
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• pp.542-545
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• 2013

State-of-the-art voice codecs have been developed to extend the input bandwidth to enhance quality while maintaining interoperability with a legacy codec. Most of them employ a modified discrete cosine transform (MDCT) for coding their extended band. We propose a source filter model-based coding algorithm of MDCT spectral coefficients, apply it to the ITU-T G.711.1 super wideband (SWB) extension codec, and subjectively test it to validate the model. A subjective test shows a better quality over the standardized SWB codec.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.9
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• pp.22-26
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• 1994

In this paper a method of designing FIR filters without digital arithmetic operationsi is persented. The filter coefficients are represented by resistors combined with a differential amplifier. With this method an FIR filter can be simply impemented without refering to complex digital arithmetic operations. Furthermore, in this scheme, no additional D/A converter is needed for D/A conversion. Spectral response response of a pulse shaping filter of 17 coefficients is shown as an illustration.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.32A no.8
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• pp.133-139
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• 1995

In this paper, the frequency separation locking and interval stabilization of 4-channel DFB-LDs have been demonstrated using a fiber Fabry-Perot filter with an free spectral range of 100GHz. Frequency fluctuation and locking range of each channel were appeared to be within 15MHz and over 12GHz, respectively. Back-reflection curve from the fiber Fabry-Perot filter was used for the extraction of an error signal in order to increase the number of accomodable channels and extinction ratio.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.743-749
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• 2007

Today, we can use radio communication device anywhere-anytime. Sometimes, we use the device in acoustic noise environment. The acoustic noise makes many problems in communication system. In acoustic noise environment, speaker cannot send clear information to receiver, because the received signal includes both speech signal and noise signal. A digital filter is useful to remove noise to get desired signal. One of methods is the adaptive digital filter using the adaptive noise canceller that automatically adjust filter parameters. This thesis addresses articulation algorithms against actual acoustic noises by means of two adaptive filtering methods. One is the adaptive noise canceller with two input channels and another is the spectral subtraction filter with one input channel. The experimental result from the proposed filter shows that the adaptive noise canceller is useful to reduce the non-stationary noises, while the spectral amplitude filter is effective for stationary noises.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.10
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• pp.1867-1872
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• 2016

In the area of digital signal processing, it is necessary to improve the quality of the speech signal after removing the background noise which exists in a various real environments. The important thing to consider when removing the background noise acoustically is that to solve the problem, depending on the information of the human auditory mechanism is mainly the amplitude spectrum of the speech signal. This paper introduces the characteristics of a frequency-weighted spectrum shaping filter for the extraction of the amplitude spectrum of the speech signal with the primary purpose. Therefore, this paper proposes an algorithm using the methods of a Wiener filter and the frequency-weighted spectrum shaping filter according to the acoustic model, after extracted the amplitude spectral information in the noisy speech signal. The spectral distortion (SD) output of the proposed algorithm is experimentally improved more than 5.28 dB compared to a conventional method.