• Journal of the Korea Institute of Military Science and Technology
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• v.3 no.2
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• pp.71-80
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• 2000

In Underwater Flight Vehicle depth control system, the followings must be required. Firstly, It need robust depth control performance which can get over parameter variation, modeling error and disturbance. Secondly, It need no oveshoot phenomenon to avoid colliding with ground surface and obstables. Thirdly, It need continuous control input to reduce the acoustic noise and propulsion energy consumption. Finally, It need effective interpolation method which can reduce the dependency of control parameters on speed. To solve these problems, we propose the Fuzzy-PID depth controller with the control parameter interpolators. Simulation results show the proposed control scheme has robust and accurate performance with continuous control input.

• 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.

• Speech Sciences
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• v.9 no.2
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• pp.1-11
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• 2002

This paper aims to provide a guideline for modelling speaker identification and speaker verification by comparing voice similarities between brothers. Five pairs of brothers who are believed to have similar voices participated in this experiment. Before conducted in the experiment, perceptual tests were measured if the voices were similar between brothers. The words were measured in both isolation and context, and the subjects were asked to read five times with about three seconds of interval between readings. Recordings were made at natural speed in a quiet room. The data were analyzed in pitch and formant frequencies using CSL (Computerized Speech Lab), PCQuirer and MDVP (Multi -dimensional Voice Program). It was found that data of the initial vowels are much more similar and homogeneous than those of vowels in other position. The acoustic data showed that voice similarities are strikingly high in both pitch and formant frequencies. It was also found that the correlation coefficient was not significant between parameters above.

• Speech Sciences
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• v.7 no.4
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• pp.49-57
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• 2000

This paper proposes an effective decision method focused on evaluating the 'stress position'. Conventional methods usually extract the acoustic parameters and compare them to references in absolute scale, adversely producing unstable results as testing conditions change. To cope with environmental dependency, the proposed method is designed to be model-based and determines the stressed interval by making relative comparison over candidates. The stressed/unstressed models are then induced from normal phone models by adaptive training. The experimental results indicate that the proposed method is promising, and that it is useful for automatic detection of stress positions. The results also show that generating the stressed/unstressed model by adaptive training is effective.

• Phonetics and Speech Sciences
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• v.2 no.4
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• pp.137-143
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• 2010

In this paper, a speech endpoint detection algorithm is proposed. The proposed algorithm is a kind of state transition-based ones for speech detection. To reject short-duration acoustic pulses which can be considered noises, it utilizes duration information of all detected pulses. For the optimization of parameters related with pulse lengths and energy threshold to detect speech intervals, an exhaustive search scheme is adopted while speech recognition rates are used as its performance index. Experimental results show that the proposed algorithm outperforms the baseline state-based endpoint detection algorithm. At 5 dB input SNR for the beamforming input, the word recognition accuracies of its outputs were 78.5% for human voice noises and 81.1% for music noises.

• Phonetics and Speech Sciences
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• v.9 no.2
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• pp.53-59
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• 2017

The present study examined if individual listeners' perceptual variations were associated with their cognitive characteristics indexed by the Autistic Spectrum Quotient (AQ). This study first investigated the perception of the lexical pitch accent contrast in the Kyungsang Korean currently undergoing a sound change, and then tested if listeners' perceptual variations were correlated with their AQ scores. Eighteen Kyungsang listeners in their 20s participated in the perception experiment where they identified two contrastive accent words for auditory stimuli systematically varying F0 scaling and timing properties; the participants then completed the AQ questionnaire. In the results, the acoustic parameters reporting reduced phonetic differences across accent contrasts for younger Kyungsang generation played a reliable role in perceiving the HH word from HL, suggesting the discrepancy between the perception and the production in the context of sound change. This study also observed that individuals' perceptual variations were negatively correlated with their AQ sub scores. The present findings suggested that the sound change might appear differently between production and perception with a different time course, and deviant percepts could be explained by individuals' cognitive measure.

• The Journal of the Acoustical Society of Korea
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• v.15 no.4
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• pp.77-82
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• 1996

In this paper an identification method is presented to obtain the equivalent electric model of a sandwitch type piezoelectric transducer. Unknown parameters related to the equivalent circuit are identified by solving a nonlinear optimization problem which can minimize an error between the experimental and analytical admittances in air. The proposed method is applied to an example transducer. The validity of equivalent circuit model is demonstrated by the comparison between the experimental measurements and analytical calculations of transmitting voltage response(TVR) and receiving voltage response(RVS).

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.385-388
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• 2002

현재 복부용 초음파 진단장치에 가장 일반적으로 사용 되고 있는 3.5MHz의 굴곡형 선형배열(curved linear array) 탐촉자에 대하여, 탐촉자 요소의 폭과 높이, 측 방향 및 높이 방향의 곡률 반경 둥의 구조적인 파라메터 변화가 음장에 미치는 영향을 시물레이션을 통하여 체계적으로 해석하였다. 시물레이션에 있어서, 탐촉자는 128개의 요소변환자 중 32개에 의해 초음파 빔을 형성하며, 매질 중에 방사된 파형은 코사인 포락선을 갖는 3주기의 펄스인 것으로 일정하게 가정하였다. 시뮬레이션의 결과, 탐촉자의 측 방향 및 높이 방향 곡률보다는 탐촉자 요소의 폭 및 개구높이가 음장에 더 많은 영향을 미치고 있음을 알 수 있었다.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.8
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• pp.1356-1361
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• 2008

A switched reluctance generator(SRG) has simple magnetic structure, and needs simple power electronic driving circuit. But, a SRG are no windings or permanent magnets on the rotor, and there are concentrated windings placed around each salient pole on the stator. Because of the characteristics of time-sharing excitation, the control of SRG is very flexible. And there are several parameters for controlling SRG, such as switch turn-on angle, switch turn-off angle, and exciting voltage and controlling mode, all these will affect the generation greatly. A SRG has positive torque at increasing inductance region and negative torque at decreasing inductance region. In this paper, we studied characteristics about the switch turn-on and off angles according to switch method for constant output voltage of the fixed speed SRG. It is the acoustic noise and torque ripple characteristics. Characteristics for a switch angle and method are presented by experiment using a 50W SRG with 12/8 poles.

• Proceedings of the KSPS conference
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• 2005.11a
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• pp.17-20
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• 2005

The purpose of this project is to develop a device that can automatically measure pronunciation of English speech produced by Korean learners of English. Pronunciation proficiency will be measured largely in two areas; suprasegmental and segmental areas. In suprasegmental area, intonation and word stress will be traced and compared with those of native speakers by way of statistical methods using tilt parameters. Durations of phones are also examined to measure speakers' naturalness of their pronunciations. In doing so, statistical duration modelling from a large speech database using CART will be considered. For segmental measurement of pronunciation, acoustic probability of a phone, which is a byproduct when doing the forced alignment, will be a basis of scoring pronunciation accuracy of a phone. The final score will be a feedback to the learners to improve their pronunciation.