• Journal of KSNVE
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• v.9 no.2
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• pp.265-272
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• 1999

A LMS algorithms has been widely used for an adaptive filter algorithm in active noise control systems. But this algorithm has poor convergence and it is very difficult to select optimal convergence parameters in this algorithm. In this paper, a fuzzy LMS algorithm where the convergence parameters are computed using a fuzzy logic controller was proposed. A proposed algorithm was applied to active noise control system in HVAC(central Heating Ventilation and Air Conditioning) ducts. The experimental ducts and experimental apparatus were designed and manufactured for experiments, and the modelling of the experimental ducts was also performed for computer simulations. Computer simulations and experiments were performed to show the effectiveness of a proposed algorithm.

• Proceedings of the KIEE Conference
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• 1990.07a
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• pp.106-112
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• 1990

This paper presents the FIR filtering scheme for digital distance protection. In most faults, there exists severe oscillation of apparent impedance according to distortion in transient waveforms. In these cases, this scheme provides more accurate fault detection and fault location, as compared with conventional schemes for digital distance protection or Kalman filtering schemes. The test of these facts were performed on EMTP result data. For the real time implementation of the proposed schemes, the motorola DSP 56001 was used to the main calculation board of the digital protective relaying system.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.11-20
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• 1997

The passive noise control techniques used until now cancel the noise in terms of the characteristics of materials, which increase the mass and the dimension and have a limit that is effective only to attenuate the high frequency components of the noise. But the active noise control techniques developed in recent years have merits that they decrease the mass and the dimension and are effective to attenuating the low frequency noises. In this paper, the real-time active noise control system attenuating the engine booming noise in a car using the digital signal processing(DSP) techniques in terms of the principle of active noise control. The multiple-error filtered-x LMS(Least-Mean Square) algorithm is used as the adaptive algorithm for active noise control and is implemented using the DSP processor Motorola DSP56001 as a controller. According to the result that the experiments are performed for the engine as the RPM changes in a car, the noise attenuating performances are achieved in an overall car interior and is verified to be 20 dB higher for pure-tone and globally, 15 dB.

• Proceedings of the KIEE Conference
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• 1992.07a
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• pp.63-66
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• 1992

This paper presents the real time digital differential protection scheme for transformer. Inrush and Internal fault conditions are distinguished by the relative magnitudes of fundamental and second harmonic components which are extracted from differential currents. The algorithm is simulated and implemented using a prototype relay which is made up of Intel 80286 CPU and Motorola DSP-56001. The testing data of inrush and Internal fault signals which are sampled at a rate of 12 times a cycle are obtained by EMTP, and tested In real time using simulator which downloaded those data.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.2
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• pp.51-58
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• 1993

This paper describes the development of a speaker-dependent isolated word recognizer as applied to voice dialing in a car noise environment. for this purpose, several methods to improve performance under such condition are evaluated using database collected in a small car moving at 100km/h The main features of the recognizer are as follow: The endpoint detection error can be reduced by using the magnitude of the signal which is inverse filtered by the AR model of the background noise, and it can be compensated by using variants of the DTW algorithm. To remove the noise, an autocorrelation subtraction method is used with the constraint that residual energy obtainable by linear predictive analysis should be positive. By using the noise rubust distance measure, distortion of the feature vector is minimized. The speech recognizer is implemented using the Motorola DSP56001(24-bit general purpose digital signal processor). The recognition database is composed of 50 Korean names spoken by 3 male speakers. The recognition error rate of the system is reduced to 4.3% using a single reference pattern for each word and 1.5% using 2 reference patterns for each word.