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      • 상대적으로 안정한 Lattice 필터 구조

        지석근 群山大學校 情報通信技術硏究所 1999 情報通信技術硏究論文集 Vol.3 No.-

        This paper proposes a new lattice filter structure that has the following properties. When the filter is Linear Time Invariant(LTI), it is equivalent to the celebrated Gray Markel Lattice filter. When the lattice parameters vary with time it sustains arbitrary rate of time variations without sacrificing a prescribed degree of stability, provided that the lattice coefficients are magnitude bounded in a region where all LTI lattices have the same degree of stability. It is also presented that the resulting Linear Time Varying lattice obeys an energy contraction condition. This structure thus generalizes the normalized Gray-Markel lattice which has similar properties but only with respect to stability as opposed to relative stability.

      • 複數音源에서의 騷音除去

        池石根 군산대학교 1993 群山水産專門大學 硏究報告 Vol.28 No.1

        This paper described ANC(Active Noise Control) systems for noise Cancelling in multiple channel. Until this time, it was reported that the study of ANC for the mutiple inputs and the multiple outputs had no correlation between input signals of adaptive filters. In fact it is difficult independently checking of noise source signals and insufficient about this study. In this paper we present the correlation elimination filter between input signals in adaptive filters for noise cancelling. As the results of simulation, high convergence speed using the correlation elimination filter was obtained.

      • 다중 채널 능동 소음 제어를 위한 Filtered-x LMS와 고속 LMS 알고리즘

        지석근 群山大學校 情報通信技術硏究所 1998 情報通信技術硏究論文集 Vol.2 No.-

        In some situations where active noise control could be used, the well known multichannel version of the filtered-x LMS adaptive filter is too comptuationally complex to implement. In this paper, we develop a fast, exact implementation of this multichannel system whose complexity is approximately O(2L) per filter channel, where L is the FIR filter length. In addition, we provide a computationally-efficient method for effectively removing the delays of the secondary paths within the coefficient updates, thus yielding a fast implementation of the LMS adaptive algorithm for multichannel active noise control. Examples illustrate both the equivalence of the algorithms there original counterparts and the computational gains provided by the new algorithms.

      • 가동중인 변압기 이상음의 초음파 분석

        지석근 국립7개대학공동논문집간행위원회 2002 공업기술연구 Vol.2 No.-

        A running high voltage equipments produce ultrasonic wave that has unique sound by the specific characteristics of the electricity. The generation of the ultrasonic wave is made by the electric transform like arcing, corona, and tracking so on. The mechanical losses and fatal human damages are happened by the electric failure of high voltage equipments. To prevent and diagnose the obstacle factors of the high voltage equipments, the measurement of the ultrasonic wave became to be prominent. However standardized data have been a deficient situation by now. This paper measures the ultrasonic wave coming from the real running transformer equipments and transforms it as an audio frequency. Measured data represents as frequency and time domain through the FFT(Fast Fourier Transform) transform. In conclusion, the purpose of this paper is to standardize the analyzed data.

      • 多衆채널 適應알고리즘을 利用한 能動騷音制御

        지석근(Suk-Kun Jee) 군산대학교 수산과학연구소 1994 水産科學硏究 Vol.- No.10

        다중능동 소음제어 시스테에서 계산량을 줄이는 새로운 ES(Error Scanning) 알고리즘을 제안한다. 음장의 소음원 입력신호 수 K, 음장의 오차 측정용 에러 마이크로폰 M, 제어용 신호의 스피커 수 L로 음장의 소음제거 시스템을 나타낸다. 제안한 알고리즘으로 ANC(Active Noise Control) chair에 적용하여 소음제거부분을 실현했다. 실험은 CASE(111), (112), (122)에 대해 구성을 하고 대역 제한된 랜던잡음을 부가 스피커 전방 약 2[m]의 위치에 배치했다. 능동필터는 샘플링 주파수 700[Hz], 탭 길이는 128점을 사용했다. 실험결과 주파수 60[Hz]이상에서 약 15~25[dB]의 잡음 억압효과가 확인되었다. 본 연구의 응용으로서는 공장내, 선박의 엔진실내 등에서 작동자의 머리주변에 소음제거를 실현함으로서 작업환경의 개선 피로경감을 기대하고 있다. This paper proposes new error scanning algorithm to reduce the calculation speed in multiple active noise control system. And theory was introduced about noise input signal number K, error microphone number M for error measurement and secondary loudspeaker number L for noise control in the sound, that is CASE(K, M, L). The noise control spot was realized using proposed algorithm of active noise control chair. Experiments are origanized CASE(111), (112), (122) and secondary loudspeakers are located about 2[m] in front of noise source which filtered random noise. Sampling frequency 700[Hz], tap length 128 points was used in active adaptive filters. In the results of experiments, obtained 15~25[dB] noise suppression in the above 60[Hz]. In the applications of the experiments, expected that improve of working environment and fatigue reduction as noise suppression around head of operator in the facory line or in the engine room of the ship.

      • 多衆채널 適應알고리즘을 利用한 能動騷音制御

        지석근(Suk-Kun Jee) 군산대학교 수산과학연구소 2004 군산대학교 수산과학연구소 연구논문집 Vol.3 No.-

        다중능동 소음제어 시스템에서 계산량을 줄이는 새로운 ES(Error Scanning) 알고리즘을 제안한다. 음장의 소음원 입력선호 수 K, 음장의 오차 측정용 에러 마이크로폰 M, 제어용 신호의 스피커 수 L 로 음장의 소음제거 시스템을 나타낸다. 제안한 알고리즘으로 ANC(Active Noise Control) chair 에 적용하여 소음 -제거부분을 실현했다. 실험은 CASE(111), (112), (122) 에 대해 구성을 하고 대역 제한된 랜던잡음을 부가 스피커 전방 약 2[m] 의 위치에 배치했다 . 능동필터는 샘플링 주파수 700[Hz], 탭 걸이는 128 점을 사용했다. 실험결과 주파수 60[Hz] 이상에서 약 15-25[dB] 의 잡음 억 압효과가 확인되였다. 본 연구의 응용으로서는 공장내, 선박의 엔진실내 등에서 작동자의 머리주변 에 소음제거를 실현하므로서 작업환경의 개선 피로경감을 기대하고 있다. This paper proposes new error scanning algorithm to reduce the calculation speed in multiple active noise control system. And theory was introduced about noise input signal number K, error microphone number L for error measurement and secondary loudspeaker number L for noise control in the sound, that is CASE(K, !'vI, L). The noise control spot was realized using proposed algorithm of active noise control chair. Experiments are origanized CASE(111), (112), (122) and secondary loudspeakers are located about 2[m] in front of noise source which filtered random noise. Sampling frequency 700[Hz], tap length 128 points was used in active adaptive filters. In the results of experiments. obtained 15 - 25 [dB] noise suppression in the above 60[Hz]. In the appli. cations of the experiments. expected that improve of working environment and fatigue reduction as noise suppression around head of operator in the facory line or in the engine room of the ship.

      • LSP 音聲分析과 認識에 關한 硏究

        지석근(Jee Suk.Kun) 군산대학교 수산과학연구소 1990 水産科學硏究 Vol.- No.6

        This paper described a speech analysis method by using a LSP(Line Spectrum Pair) parameter which is directly related to spectral enevlope of speech signal. The LSP parameter can be estimated by a distance between the two adjacent formant frequencies which are high resonance frequency in the frequency domain and uses idea that the higher resonance frequency, the narrower distance between the adjacent formant frequencies. In the experiment for the LSP analysis of Korean vowel, 256 data samples are used for the analysis frame and the order of analysis is 10. As the results of experiment, it was obtained that the formant frequency of LSP is the same with the LPC envelope for Korean vowel and the scores of recognition when used the Euclidian Distance method for the PARCOR parameter and the LSP parameter are 91.8% and 93.3%, respectively.

      • 體育館에서의 音聲明瞭度의 향상

        지석근(Jee Suk Kun) 군산대학교 수산과학연구소 1991 水産科學硏究 Vol.- No.7

        When using Gymnasium with audiotorium, it's important that the audience very well listen the sound in audiotorium. The curve array speaker and the auxiary line array speaker were so made as to listen the sound articultoary. This paper discribed the electric loudspeaker system for audiotorium, not to alter the inside structure of a Gymansium. This system was checked for the articulation index test, the sound pressure and the reverberation time. In result of the articulation index test or the loudspeaker system, the index showed 80% approximately.

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