• Journal of Sensor Science and Technology
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• v.13 no.6
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• pp.479-487
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• 2004

The prevention of noise induced hearing loss is very important, because there is no treatment for it. There are some kinds of devices for hearing protection, and those are effective in preventing a noise induced hearing loss. However, people often resist the use of hearing protection devices because it is difficult to have a conversation. Therefore, a hearing protection device is must effective not only in hearing protection but also in preserving communication ability. In this paper, we proposed a small acoustic filter for ITE type noise protector to solving the problem about the difficulties of conversation. That is applied a principle of acoustic filters that have been used for a muffler of automobiles, guns, and etc. To find out the sound transmission characteristic at the eardrum, we regarded an acoustic filter and external ear canal as a coupled system. So, we simulated the coupled system with OrCad, and experimented with a designed acoustic filter and a 2 cc coupler which has the same transmission characteristic as the external ear canal has. We confirmed that it is possible to adjustment acoustic transmission characteristics through simulation of electrical model for acoustic filter and external ear and experiments using designed small acoustic filters.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.205-209
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• 2013

A conventional acoustic echo suppressor (AES) considering only room impulse response between a loudspeaker and a microphone eliminates acoustic echo from the microphone input. However, in a nonlinear acoustic echo environment, the conventional AES degraded because of a nonlinearity of the loudspeaker. In this paper, we adopt AES based on the frequency-domain second-order Volterra filter using Least Square method. For comparing performances, we conduct objective tests including Echo Return Loss Enhancement (ERLE) and Speech Attenuation (SA). The proposed algorithm shows better performance than the conventional in both linear and nonlinear acoustic echo environments.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.24 no.12
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• pp.91-99
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• 2010

HVDC converter station consists of a number of equipment such as converter transformer, ac filter, thyristor valve and so on. They can be acoustic noise sources. In this paper, we analyzed the simulation results of the outdoor acoustic noise for HVDC converter station. It shows that maximum noise level in boundary of HVDC converter station exceeds regulation value. The main factors in generating maximum noise level are ac filter and converter transformer. Then we applied some soundproof countermeasures in HVDC converter station. Shielding wall is enough to reduce transformer noise level but not enough to reduce ac filter noise level. In case of ac filter, soundproof building is effective in satisfying noise level regulation in boundary of HVDC converter station. In addition, we also studied effects of season, soundproof woods, ground.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2001.05a
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• pp.991-996
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• 2001

It is well known that there are two experimental errors on acoustic decay measurements. One is due to the influence of the band pass filter the other one is that of an averaging device. In this paper the influence of the filter is in detail investigated. To minimize the influence of filter, the product of the filter bandwidth Β (3㏈ bandwidth) and the reverberation time T$\sub$60/ of the room under test is at least 16. Moreover, if the initial part of an acoustic decay curve is important, the strong requirement, i.e. BT$\sub$60/＞ 64, must be satisfied. In this paper, the wavelet filter bank instead of the band pass filter bank is applied to obtain an acoustic decay curve. As results, the influence of filter is reduced and then the value of BT$\sub$60/ required for obtaining an acceptable decay curve is at least 4. The strong requirement for the initial part of a decay curve is also replaced by the BT$\sub$60/＞ 16 instead of BT$\sub$60/＞ 64.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.5-13
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• 2001

It is well known that there are two experimental errors on acoustic decay measurements. ,One is due to the influence of the band pass filter the other one is that of an averaging device. In this paper the influence of the filter is investigated in detail. To minimize the influence of the filter, the product of the filter bandwidth B (3dB bandwidth) and the reverberation time T/sub 60/ of the room under test should be at least 16. Moreover, if the initial part of an acoustic decay curve is important, the strong requirement, i. e. BT/sub 60/〉64, must be satisfied. In this paper, the wavelet filter bank instead of the band pass filter bank is applied to obtain an acoustic decay curve. As a result, the influence of filter is reduced and then the value of BT/sub 60/ required for obtaining an acceptable decay curve becomes at least 4. The strong requirement for the initial part of a decay curve is also replaced by the BT/sub 60/〉16 instead of BT/sub 60/〉64.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.1077-1080
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• 2002

In building acoustics, reverberation time is an important acoustic parameter. However, it is often difficult to measure short reverberation times at low frequencies using the traditional band pass filter bank if the product of bandwidth (B) and reverberation time (T) is small. It is well known that the minimum permissible product of bandwidth and reverberation time of the traditional band pass filter is at least 16 ［F. Jacobsen, J. Sound Vib. 115, 163-170 (1987)]. This strict requirement makes it difficult to measure short reverberation times of an acoustic room at low frequencies exactly. In order to reduce this strict requirement, recently, the wavelet filter bank is developed and the minimum permissible product of bandwidth and reverberation time is replaced with 4 [S. K. Lee, J, Sound Vib. 252, 141-153 (2002)]. In the present paper, it is demonstrated how the short reverberation times at low frequencies are successfully measured by using the wavelet filter bank. In order to present this job, two synthetic signals and one measured signal are used for impulse responses of an acoustic room.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.9
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• pp.1632-1652
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• 2011

This paper considers the object tracking problem in three dimensional (3-D) space when the azimuth and elevation of the object are available from the passive acoustic sensor. The particle filtering technique can be directly applied to estimate the 3-D object location, but we propose to decompose the 3-D particle filter into the three planes' particle filters, which are individually designed for the 2-D bearings-only tracking problems. 2-D bearing information is derived from the azimuth and elevation of the object to be used for the 2-D particle filter. Two estimates of three planes' particle filters are selected based on the characterization of the acoustic sensor operation in a noisy environment. The Cramer-Rao Lower Bound of the proposed 2-D particle filter-based algorithm is derived and compared against the algorithm that is based on the direct 3-D particle filter.

• The Journal of the Acoustical Society of Korea
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• v.38 no.1
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• pp.114-119
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• 2019

The transmitted acoustic signals are severely influenced by multiply reflected signals from boundaries, such as sea surface and bottom in the shallow water. Very large reflection signals from boundaries cause inter-symbol interference so that the performance of the underwater acoustic communication is degraded. Usually, the waveform shaping filters are used to prevent the reflected signals under this kind of acoustic channel. Especially, the raised cosine filter is widely used, which can also be used to restrict the bandwidth of the transmitted signal. In this study, we evaluate the raised cosine filter for image data transmission in the shallow water, and propose a new modified raised cosine filter. The QPSK (Quadrature Phase Shift Keying) system is used for the underwater acoustic communication simulations with different distances and symbol rates. As a result, the bit error rate was reduced from the minimum 1.0 % to the maximum 32 %.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.12 no.9
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• pp.694-701
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• 2002

In this paper, through the study on locations of structural transducers for active control of the radiated sound from the vibrating plate, the active structural acoustic control (ASAC) system is proposed. And, for the evaluation of the proposed location, the experiment of the active structural acoustic control is implemented using the multi-channel filtered-x LMS algorithm and an additional filter (Acoustic Prediction Filter) to estimate the radiated sound using the acceleration signals of the plate. The structural transducers are piezoceramic actuator (PZT) and accelerometer. PZT is used as an actuator to reduce the vibration and the radiated sound. To maximize the control performance, each PZT actuator is located at the position that has the largest control sensitivity of the plate bending moment in the direction of x and y coordinates and the optimal PZT location is validated experimentally. Also, to find the acoustic prediction filter accurately, two accelerometers are located at the positions that have the largest radiation efficiencies of the plate, and the proposed locations are validated by simulation using the Rayleigh integral. The multi-channel filtered-x LMS algorithm is introduced to control a complex 2-D structural vibration mode. Finding the locations of structural transducers for active structural acoustic control of the radiated sound, the active structural acoustic control (ASAC) system can be presented and validated by experiments using a real time control system.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.6
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• pp.1126-1130
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• 2010

Speech production can be viewed as a filtering operation in which a sound source excites a vocal tract filter. The vocal tract is modeled as a chain of cylinders of varying cross-sectional area in linear prediction acoustic tube modeling. In this modeling the most common implementation assumes equal length of tube sections. Therefore, to model complex vocal tract shapes, a large number of tube sections are needed. This paper proposes a new vocal tract model with unfixed sectionlengths, which uses the reduced lattice filter for modeling the vocal tract. This model transforms the lattice filter to reduced structure and the Burg algorithm to modified version. When the conventional and the proposed models are implemented with the same order of linear prediction analysis, the proposed model can produce more accurate results than the conventional one. To implement a system within similar accuracy level, it may be possible to reduce the stages of the lattice filter structure. The proposed model produces the more similar vocal tract shape than the conventional one.