• Transactions of the Korean Society of Mechanical Engineers
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• v.15 no.3
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• pp.809-823
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• 1991

본 연구에서는 Wiener 필터링 이론에 의하여 소음원의 입력신호에 대한 최적 한 예측을 할 수 있는 최적예측기(optimal predictor)로써 부가적인 음을 발생시키고 입력신호 및 출력신호 간의 차인 오차를 최소화시키도록 하는 적응신호처리방법에 대 해 설명하고 이러한 적응 신호처리 방법을 이용한 능동 소음 제어 방법을 제시하였다. 이와 아울러 제어계의 환경 변화에 따른 파라메타의 변화에 적응적으로 응답이 가능해 야 하는 적응 소음 제어계에서, 음향궤환과 함께 필히 고려해야하는 부가적인 전달함 수-모델과 스피커를 포함하는 보조경로 및 오차미이크로폰을 포함하는 오차경로의 전 달함수의 영향을 고려한 능동소음제어에 대해 연구하였다.

• Proceedings of the KIEE Conference
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• 1994.11a
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• pp.373-375
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• 1994

In this paper, the fuzzy LMS algorithm where the convergence coefficient is computed by a fuzzy logic controller was proposed. The proposed fuzzy LMS algorithm showed better convergence property and stability than conventional LMS algorithms. The estimation error and misadaptation degree were used for Input of the fuzzy logic controller. In a airconditioning duct case, various conditions were investigated to design active noise controllers. A case with acoustic feedback, the proposed algorithm showed good performances through computer simulations.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.11
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• pp.2299-2301
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• 2009

The adaptive IIR filters for active noise control systems are more effective when acoustic feedback exists, but the adaptive IIR filters could be unstable when the filter algorithm is not yet converged. In this paper, auto-secondary path estimation techniques and a stabilizing process for adaptive Multi-Channel Recursive LMS (MCRLMS) filters are developed to improve the stability of multi-channel active noise control systems. Experiments using a TMS320VC33 digital signal processor in a three dimensional enclosure have performed to show the effectiveness of the proposed algorithm.

• 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.

• 제어로봇시스템학회:학술대회논문집
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• 1987.10b
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• pp.631-636
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• 1987

The design of ultrasonic transducer energy processing systems requires highly reliable command featuring mechanical frequency tracking and constant velocity control of the ultrasonic transducer with an acoustic load. This paper presents a new conceptional instantaneous current resultant control base high-frequency inverter using self turn-off devices driving an electrostrictive ultrasonic transducer system and its optimum control technique, which is implemented by feed-back of the ultrasonic transducer applied voltage and instantaneous velocity of the transducer vibrating system through a Phase-Locked-Loop control scheme. The feedback voltage corresponding to instantaneous velocity is averaged over a half-period with respect to constant amplitude/constant velocity control strategy. Described are the theory of this signal detection technique and the experimental set-up.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.20 no.3
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• pp.570-575
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• 2019

Active muffler is implemented by applying active noise control technique to reduce exhaust noise of automobile muffler. Conventional Filtered_x LMS algorithm has a problem that the degree of control filter becomes very large and convergence deteriorates when acoustic feedback is present. The recursive LMS algorithm can compensate for this problem because it can be easily diverted in the adaptive filter adaptation process. In this paper, the structure of the primary path and the secondary path transfer function is designed as the IIR filter to improve the convergence performance and the computational burden, and the stabilization filter algorithm is applied to secure stability which is a disadvantage of the IIR filter structure. The stabilization filter algorithm plays a role of pulling the pole into the unit circle to prevent the pole of the transfer function corresponding to the acoustic feedback from diverging during the adaptation process. In this way, the computational burden of the active muffler system and the convergence performance can be improved. In order to show the usefulness of the proposed system, we compared the performance of the proposed Filtered_x LMS algorithm with the performance of the proposed system for the exhaust sound of a diesel engine, which is a variable environment. Compared to conventional algorithm, proposed algorithm's computational burden is less than half, and convergence performances are more than 4 times.

• Phonetics and Speech Sciences
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• v.10 no.1
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• pp.75-81
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• 2018

Diplophonia is generally defined as the perception of more than one fundamental frequency component in a voice. Its perceptual aspect has traditionally been used to evaluate diplophonia because the perceptions can be easily evaluated, but there are limitations in the validity of the reliability of the intra- and inter-raters, examination situation, and variation of voice sample. Therefore, the purpose of this study is to confirm the reliability and accuracy of auditory perceptual evaluation by comparing non-invasive indirect assessment methods (sound waveform and EGG analysis), and to identify their usefulness with diplophonia. A total of 28 diplophonic voices and 39 non-periodic voices were assessed. Three raters assessed the diplophonia by performing an auditory perception evaluation and identifying the quasi-periodic perturbations of the acoustic waveform and EGG. Among the three discrimination methods, intra- and inter-rater reliability, sensitivity, specificity, accuracy, positive likelihood ratio, and negative likelihood ratio were examined, and the McNemar test was performed to compare the discriminant agreement. The accuracy of the auditory perceptual evaluation (86.57%) was not significantly different from that of sound waveform acoustic (88.06%), but it was significantly different from that of EGG (83.33%). The reading time (6.02 s) for the auditory perceptual evaluation was significantly different from that for sound waveform analysis (30.15 s) and EGG analysis (16.41 s). In the discrimination of diplophonia, auditory perceptual evaluation has sufficient reliability and accuracy as compared to sound waveform and EGG. Since immediate feedback is possible, auditory perceptual evaluation is more convenient. Therefore, it can continue to be used as a tool to discriminate diplophonia in clinical practice.

• Journal of Biomedical Engineering Research
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• v.23 no.3
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• pp.227-233
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• 2002

The IME(implantable middle ear) system is Promising due to its ability to free from sound feedback and Produce a good sound quality and intelligibility with low distortion even if it is operated with high gain for severe hearing impaired. The differential electromagnetic vibration transducer. which was developed for using in IME system and has two small magnets attached the same Pole facing in the coil. is not influenced by environmental external magnetic field. Besides, it has high vibration efficiency and good frequency response characteristics. In this Paper, using acoustic model of the transducer and ear model of normal Person. the signal transfer characteristics of the IME system are analyzed and investigated From the differences of the characteristics between normal ear and the IME system, it is Possible that design of the IME system that have the signal transfer characteristics similar to normal person's ear.

• Journal of IKEEE
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• v.5 no.1 s.8
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• pp.16-23
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• 2001

Passive noise reduction is a classical approach to attenuate industrial noise, and an active noise cancellation has several advantages over the passive noise cancellation. The active noise reduction system offers a better low frequency performance with a smaller and lighter system. This paper presents a simple active closed loop control system which consists of an controller for inverting and compensating the phase delay, a microphone for picking up the external noise, and a loudspeaker for radiating the acoustic out of phase signal to reduce the external noise, and external noise can be reduced after compensating the phase difference to be $180^{\circ}$ in the frequency of maximum value in the amplitude response. The noise of the phase delay covered from $50^{\circ}\;to\;310^{\circ}$ tends to be reduced in the active noise control system and it is possible to obtain a noise cancelling of up to approximately 20[dB] at the ears in the enclosurer.

• Proceedings of the Korea Committee for Ocean Resources and Engineering Conference
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• 2003.05a
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• pp.149-156
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• 2003

This paper presents an underwater hybrid navigation system for a semi-autonomous underwater vehicle (SAUV). The navigation system consists of an inertial measurement unit (IMU), an ultra-short baseline (USBL) acoustic navigation sensor and a doppler velocity log (DVL) accompanying a magnetic compass. The errors of inertial measurement units increase with time due to the bias errors of gyros and accelerometers. A navigational system model is derived to include the error model of the USBL acoustic navigation sensor and the scale effect and bias errors of the DVL, of which the state equation composed of the navigation states and sensor parameters is 25 in the order. The conventional extended Kalman filter was used to propagate the error covariance, update the measurement errors and correct the state equation when the measurements are available. Simulation was performed with the 6-d.o.f. equations of motion of SAUV in a lawn-mowing survey mode. The hybrid underwater navigation system shows good tracking performance by updating the error covariance and correcting the system's states with the measurement errors from a DVL, a magnetic compass and a depth senor. The error of the estimated position still slowly drifts in horizontal plane about 3.5m for 500 seconds, which could be eliminated with the help of additional USBL information.