• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.333-341
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• 2008

In this paper, a new interpolation model for the head related transfer function (HRTF) was proposed. In the method herein, we assume that the impulse response of the HRTF for each azimuth angle is given by linear interpolation of the time-delayed neighboring impulse responses of HRTFs. The time delay of the HRTF for each azimuth angle is given by sum of the sound wave propagation time from the ears to the sound source, which can be estimated by using azimuth angle, the physical shape of the underlying head and the distance between the head and sound source, and the refinement time yielding the minimum mean square error. Moreover, in the proposed model, the interpolation intervals were not fixed but varied, which were determined by minimizing the total number of HRTFs while the synthesized signals have no perceptual difference from the original signals in terms of sound location. To validate the usefulness of the proposed interpolation model, the proposed model was applied to the several HRTFs that were obtained from one dummy-head and three human heads. We used the HRTFs that have 5 degree azimuth angle resolution at 0 degree elevation (horizontal plane). The experimental results showed that using only $30\sim40%$ of the original HRTFs were sufficient for producing the signals that have no audible differences from the original ones in terms of sound location.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1991.04a
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• pp.17-22
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• 1991

최근 기계장치로부터 발생하는 소음을 감소시키는 새로운 방법으로서, 능동 적으로 소음을 제어하는 방법에 대한 연구가 활발히 진행되고 있다. 이것은 원하지 않는 소음을 그 신호의 역위상을 갖는 부가음을 이용하여 능동적으 로 감쇠시키는 방법으로서, 저주파수 대역에서 비효율적인 수동적인 방법인 소음기둥에 대한 대안으로 많은 학자들의 관심의 대상이 되어 왔다. 초기에 는 소음을 줄이기 위해 요구되는 여러가지 음향요소의 전달함수를 제어하는 데 대한 불가능성으로 인해 능동 소음제어에 대한 실질적인 발전이 지연되 어 왔으나 최근 마이크로 컴퓨터를 비롯한 전자공학의 발전으로 인해 적응 신호처리 분야가 등장하게 되었으며, 음향계의 소음을 원하는 수준까지 제어 하는 능동 소음제어의 실시간 구현이 가능하게 되었다. 그 중에서도 음이 1 차원적으로 전파한다고 볼 수 있는 길이가 긴 덕트구조물에서의 능동 소음 제어는 가장 기본적이며 현실적으로 자동차 배기계나 냉동.공조설비에 있어 서 실용적으로 적용할 수 있는 문제임 만큼 많은 연구가 이루어지고 있다. 이러한 능동 소음제어 방법을 음향계에 적용하였을 때, 부가적인 음을 발생 하는 제어용 스피커로 인해 입력마이크로폰으로의 음향궤환이 존재하고 이 에 따라 제어계가 불안정해질 수 있으며, 또한 변환기의 사용으로 인한 부가 적인 전달함수가 존재하므로 이에 대한 중요한 의미를 갖고 고려하여야 한 다. 본 연구에서는 적응 필터링 이론에 의한 소음원의 입력신호에 대한 최적 한 예측으로써 부가음을 발생시키고, 입력신호 및 제어된 출력신호간의 차인 오차를 최소화 시키도록 하는 오차적응제어법을 이용한 능동소음 제어 방법 을 제시하였다. 이와 아울러 제어계의 환경변화에 따른 파라메타의 변화에 적응적으로 응답이 가능해야 하는 적응 소음제어 시스템에서, 음향궤환과 함 께 필히 고려해야 하는 부가적인 전달함수의 영향을 고려한 능동 소음제어 에 대해 연구하였다. 경량화 추세에 따라 지반이나 케이싱이 경량이거나 유연하여 회전축과 동적으로 연성된 경우 회전축-베어링-지반으로 이루어진 2중구조의 회전축 계 동특성을 해석할 수 있는 프로그램을 개발하므로서 회전 기계류의 진동 전반에 걸친 문제점에 대한 그 원인과 현상을 명확히 분석하여 국내의 전기 계류의 보다 신뢰성있는 설계 및 제작자료를 확보하는데 기여할 수 있게 하 였다.존의 small molecular Gd-chelate에 비해 매우 큼을 알 수 있었다. MnPC는 간세포에 흡수된 후 담도계로 배출되는 간특이성 조영제임을 확인하였다. 장비 내에서 반복 시행한 평균값의 차이는 대체적으로 유의한 차이가 없었으나, 다른 장비에서 반복 시행한 장비간의 사이에는 유의한 차이가 있는 경우가 더 많았다. 따라서 , MRS 검사를 소뇌나 뇌교의 어떤 절환에 적용하기 전에 각 장비 마다 정상 기준치를 반드시 얻은 후에 이상여부를 판 정하는 것이 필수적이라고 생각된다.EX> 이상이 적절한 진단기준으로 생각되었다. $0.4{\;}\textrm{cm}^3$ 이상의 좌우 부피차를 보이는 모든 증례에서 육안적으로도 해마위축이 뚜렷이 나타났다. 결론 : MR영상을 이용한 해마의 부피측정은 해마경화증 환자의 진단에 있어 육안적인 MR 진단이 어려운 제한된 경우에만 실제적 도움을 줄 수 있는 보조적인 방법으로 생각된다.ofile whereas relaxivity at high field is not affected by τS. On the other hand, the change in τV does not affect low field profile but strongly in fluences on both inflection fie이 and the maximum relaxivity value. The results shows a fluences on both inflection field

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.14 no.5
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• pp.43-49
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• 2000

In this paper, a new control method is presented to reduce acoustic resonance phenomena of metal halide lamps with electronic ballasts. A behavior of frequency controlled resonant inverter is analyzed and the transfer function is obtained. An integrator as a current controller for ballast is employed. By using both a disturbance of the lamp power and spread spectrum effects for reducing acoustic resonance, an amplitude modulation is done by using the current controller with 200[Hz] sinusoidal wave. Arc stabilities of MH 250[W] lamps are studied with changing the modulation index. Experiments with two MH 250[W] lamps, made by different manufacturers, showed the validity of the proposed method.

• The Journal of the Acoustical Society of Korea
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• v.29 no.5
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• pp.339-346
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• 2010

For the implementation of the crosstalk canceller, the filters with large length are needed, which is because that the length of the filters greatly depends on the length of the head-related impulse responses. In order to reduce the length of the crosstalk cancellation filters, many methods such as frequency warping, common acoustical pole and zero (CAPZ) modeling have been researched. In this paper, we propose a new method combining these two methods. To accomplish this, we design the filters using the CAPZ modeling on the warped domain, and then, we implement the filters using the poles and zeros de-warped to the linear domain. The proposed method provides improved channel separation performance through the frequency warping and significant reduction of the complexity through the CAPZ modeling. These are confirmed through various computer simulations.

• 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.

• Transactions of the Korean Society of Automotive Engineers
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• v.2 no.6
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• pp.29-37
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• 1994

Active noise control of random noise which propatate in the vehicle cabin as a form of spherical wave is the target of this study. In the previous study, the adaptive algorithm for adaptive controller is presented for the application in active noise control system. And for the preliminary study of adaptive active noise control in vehicle cabin as a real system, a computer simulation is performed on the effectiveness of the adaptive algorithm in the amplitude of the pressure fluctuation. This work studies the implementation of multi-channel feedforward adaptive algorithm for the reduction of the noise inside a vehicle cabin using a number of secondary sources derived by adaptive filtering of reference noise source. Multi-channel adaptive feedforward algorithm are verified in numerical simulation and semi-experimental justification of developed system is made on a domestic passenger car. In the results of semi-experimental study, the noise of specific region in the interior of automobile are reduced for the appreciabe sound pressure level in the operating engine rpm and finally this study suggests the capabilities of the real time active noise control in 3 dimensional acoustic fields.

• The Journal of the Acoustical Society of Korea
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• v.25 no.4
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• pp.184-190
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• 2006

Recently, Lee et al. have proposed an algorithm utilizing the signals from different paths by using bottom mounted simple linear array to estimate 3-D location of oceanic target. But this algorithm assumes that sound velocity is constant along depth of sea. Consequently, serious performance loss is appeared in real oceanic environment that sound speed is changed variously. In this paper, we present a 3-D near field localization algorithm for inhomogeneous sound speed. The proposed algorithm adopt localization function that utilize ray propagation model for multipath environment with linear sound speed profile(SSP), after that, the proposed algorithm searches for the instantaneous azimuth angle, range and depth from the localization cost function. Several simulations using linear SSP and non linear SSP similar to that of real oceans are used to demonstrate the performance of the proposed algorithm. The estimation error in range and depth is decreased by 100m and 50m respectively.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.16 no.2
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• pp.9-17
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• 2002

A high-efficiency electronic ballast for HID lamps is presented. The ballast consists of a PFC and a resonant inverter. To reduce losses of the ballast, DC link voltage should be determined by taking into account the peak voltage of lamp and the maximum flux density should be kept 0.2[T] on all of inductors. AR inductor at bridge diode is employed in order to remove currant harmonics from PFC. An inductor is connected in series with an electrolytic capacitor at DC link to reject high-frequency current. The acoustic resonance is eliminated using the stead spectrum technique. The electronic ballast for 250[W] metal-halide discharge lamp is implemented and 96[%] efficiency, no acoustic resonance and low conducted EMI level are accomplished.

• Environmental and Resource Economics Review
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• v.14 no.2
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• pp.291-336
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• 2005

This paper investigates how oil price changes have an impact on the Korean manufacturing production activities. For this investigation, we use a structural VAR motel to estimate impulse response functions of industrial production, producer price and export price to an oil price increase over manufacturing industries. It finds that in most manufacturing industries, an oil price increase leads to decreases in industrial productions and domestic prices except energy intensive industries, but to increases in industrial export prices except non-metallic (26), computers and offic machinery (30), electronic components, radio, television and communication (32) industries. This result explains that an oil price increase makes negative impacts on the manufacturing production activities not only through demand slowdown in the domestic markets but also through supply contraction in the export markets.

• The Journal of the Acoustical Society of Korea
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• v.29 no.7
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• pp.458-467
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• 2010

In this paper, a new audio reproduction system was developed in which the cross-talk signals would be reasonably cancelled at an arbitrary listener position. To adaptively remove the cross-talk signals according to the listener's position, a method of tracking the listener position was employed. This was achieved using the two microphones, where the listener direction was estimated using the time-delay between the two signals from the two microphones, respectively. Moreover, room reverberation effects were taken into consideration where linear prediction analysis was involved. To remove the cross-talk signals at the left-and right-ears, the paths between the sources and the ears were represented using the KEMAR head-related transfer functions (HRTFs) which were measured from the artificial dummy head. To evaluate the usefulness of the proposed listener tracking system, the performance of cross-talk cancellation was evaluated at the estimated listener positions. The performance was evaluated in terms of the channel separation ration (CSR), a -10 dB of CSR was experimentally achieved although the listener positions were more or less deviated. A real-time system was implemented using a floating-point digital signal processor (DSP). It was confirmed that the average errors of the listener direction was 5 degree and the subjects indicated that 80 % of the stimuli was perceived as the correct directions.