• The Journal of the Acoustical Society of Korea
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• v.18 no.2
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• pp.18-24
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• 1999

In this paper, we have investigated on the development of ultrasonic speakers that have higher directivity and much wider bandwidth than those of conventional loudspeakers. For optimal structure of speaker, we have analyzed material properties and dimension of the piezoelectric element and the vibrating metal plate, and their support type as well as the housing case. Based on the design, we have fabricated ultrasonic speaker prototypes, measured their performance, and verified validity of the design theory. For higher sound pressure level, the ultrasonic array speaker prototype has been fabricated in the form of an array. The design and fabrication method worked in this paper can be utilized in development of various ultrasonic speakers with higher directivity and broader bandwidth.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.05a
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• pp.551-554
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• 2005

This study uses a characteristic function to explain correlations between the objective function and design variables. Analysis of means and table of orthogonal array were carried out. The change of shape of diaphragm, thickness of diaphragm and voice coil weight based on the table of orthogonal array is made. Therefore this study carried to decide shape of diaphragm, voice coil weight and thickness of diaphragm for minimizing 1st natural frequency and maximizing 2nd natural frequency of diaphragm using design of experiments and characteristic function with constraints. we showed improved design factors that minimized 1st natural frequency and maximized 2nd natural frequency of diaphragm.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.573-578
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• 2003

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this paper, it was described to be able to enhance the speech quality through microphone array, and received the only signal of speaker. Unfortunately, as it using estimated the signal in advance, it is not matched in a real acoustic environment so it has poor performance. In this paper is proposed for Adaptive Matched Filter Microphone Array that estimated acoustic room environment from the received the signal and study of the efficiency through simulations.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.15 no.7 s.100
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• pp.775-780
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• 2005

This study uses a characteristic function to explain correlations between the objective function and design variables. Analysis of means and table of orthogonal array were carried out. The change of shape of diaphragm, thickness of diaphragm and voice coil weight based on the table of orthogonal array Is made. Therefore this study carried to decide design variables for minimizing 1st natural frequency and maximizing 2nd natural frequency of diaphragm using design of experiments and characteristic function with constraints. we showed improved design variables.

• The Magazine of the IEIE
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• v.37 no.1
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• pp.52-62
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• 2010

• Proceedings of the KIEE Conference
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• 2003.11b
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• pp.175-178
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• 2003

This paper describes a active audition system for robot-human interface in real environment. We propose a strategy for a robust sound localization and for -talking speech recognition(60-300cm) based on 2-dimensional microphone array. We consider spatial features, the relation of position and interaural time differences, and realize speaker tracking system using fuzzy inference profess based on inference rules generated by its spatial features.

• Proceedings of the KSME Conference
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• 2005.11a
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• pp.186.1-186.1
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• 2005

• Journal of Satellite, Information and Communications
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• v.8 no.4
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• pp.30-36
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• 2013

Recently the globalization of the media content industry, various activities have been made in the field of art and the speaker system is very important in the sound industry which is one of the arts. The directional characteristics of a loudspeaker refer to the radiation of sound in certain directions and are among the most important features of a loudspeaker. Designing a loudspeaker that can keep all of its constant directivity at all frequencies is difficult due to the wavelengths of audio frequencies and the size of horns and transducers. This study proposed gradient array methods to improve low frequency pattern control of full-range speakers to maximize Direct to Reverberant Ratios at the listeners.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.4
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• pp.857-866
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• 2009

In this paper, a noise cancellation-method using microphone array for digital hearing aids is proposed. The microphone array is located around the ear of a dummy. Speech sound is generated from the forward speaker positioned in the front of the dummy and noise sound is generated from the backward speaker. The speech and noise are mixed in the air space and entered into the microphones. VAD(voice activity detector) and ANC(adaptive noise cancellation) methods were used to eliminate noise in the sound of the microphones. 10 two-syllable words and 4 sentences were used for speech signals. Babble and car interior noise were used for noise signals. The performance of the proposed algorithm was evaluated by SNR(signal-to-noise ratio) and PESQ-MOS(perceptual evaluation of speech quality-mean opinion score). In babble noise condition, SNR was improved as much as $7.963{\pm}1.3620dB\;and\;3.968{\pm}0.6659dB$ for words and sentences respectively. In the case of car interior noise, SNR was improved as $10.512{\pm}2.0665dB\;and\;6.000{\pm}1.7642dB$ for words and sentences respectively. PESQ-MOS of the babble noise was improved as much as $0.1722{\pm}0.0861$ score for words and $0.083{\pm}0.0417$ score for sentences. And PESQ-MOS of the car interior noise was improved as $0.2661{\pm}0.0335$ score and $0.040{\pm}0.0201$ score for words and sentences respectively. It is verified that the proposed algorithm has a good performance in noise cancellation of microphone array for digital hearing aids.

• Journal of Broadcast Engineering
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• v.20 no.5
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• pp.667-675
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• 2015

As the interest on the high-definition and high-quality broadcasting is increased, the request on the high quality sound signal is enlarged as well as on the video signal's quality. One factor contributing to the high-quality of audio signal is an expansion of reproduction channels like 10.2channel and 22.2channel, but there is a problem of speaker installation issue of these many channels. One solution to solve this problem, we can use frontal loudspeaker array reproduction technique making virtual surround sound. So in this paper, we introduce theocratical analysis on the Wave Field Synthesis used for speaker array based sound reproduction and also present the result about the subjective listening test of reproduction performance based on this technique to check the perfoemance of this system. As a result, we showed WFS based frontal loudspeaker array reproduction method could provide sufficient performance compared to conventional discrete 5.1 channel reproduction method.