• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.6 no.1
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• pp.23-28
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• 2013

In this paper, background noise reduction method using dual microphone is proposed in mobile environment. Each Signal, reference and primary, would be replaced by microphone input signals, which were measured by reference and primary microphones, and then, noise reduction was performed using FDAF. After then, residual and background noise would be estimated and reduced by MMSE-LSA. For consistent noise reduction performance, result of VAD that could be caculated by PLD between two microphones was used.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2003.05b
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• pp.96-102
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• 2003

Today, stage technology is developing highly by application of digital computer. Performance is composed of audio/video and acoustic technology takes very important position in field of stage technology. Generally speaking, four factors of sound are loudness, pitch, sound timbre and duration. Loudness depends on sound pressure level, yet partly related with spectrum and dulation. Pitch depends mainly on frequence and have a relation with sound pressure and duration. sound timbre depends strongly on spectrum and have a relation with frequence. In this paper, I designed a multi-microphone system which can used in broadcasting and performance stage with vicboss 200MHz-VHF wireless microphone and vicboss 900MHz-VHF wireless microphone. I also studied about multi-microphone which can use conveniently in the super play that needs many microphones. If this multi-microphone is prodused, we could expect better sound quality and a big progress in stereo recording technology.

• Journal of KSNVE
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• v.5 no.2
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• pp.197-206
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• 1995

Microphone array is designed to measure the equivalent source height of vehicle noise. The equivalent source position is defined for an arbirary distribution of acoustic sources above a perfectly reflecting plane and a microphone array for its measurement is developed. The normalized errors of the measured equivalent source heights are defined including the effects of background noise, the geometric near field, and source size. Normalized errors of the measured source heights obtained by a nemerical simulation for each parameter lead to optimization of the microphone spacing and to the design of an array which gives the equivalent source height as a function of frequency. The performance of the designed array is verified using the stationary loudspeaker experiments.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.4
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• pp.611-614
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• 2021

In this paper, we propose a widerange array microphone for lecturer tracked with Frequency Modulated Continuous Waveform (FMCW) radar sensor. Time Difference-of-Arrival (TDoA) is often used as audio tracking, but the tracking accuracy is poor because the frequency of the voice is low and the relative frequency change is large. FMCW radar has a simple structure and is used to detect obstacles for vehicles, and the resolution can be archived to several centimeter. It is shown that the sensor is useful for detecting a speaker in open area such as a lecture, and we propose an wide range 4-element array microphone beamforming system. Through some experiments, the proposed system is able to adequately track the location and showed a 8.6dB improvement over the selection of the best microphone.

• ETRI Journal
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• v.29 no.3
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• pp.292-299
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• 2007

In this paper, we present a speech recognition system using a throat microphone. The use of this kind of microphone minimizes the impact of environmental noise. Due to the absence of high frequencies and the partial loss of formant frequencies, previous systems using throat microphones have shown a lower recognition rate than systems which use standard microphones. To develop a high performance automatic speech recognition (ASR) system using only a throat microphone, we propose two methods. First, based on Korean phonological feature theory and a detailed throat signal analysis, we show that it is possible to develop an ASR system using only a throat microphone, and propose conditions of the feature extraction algorithm. Second, we optimize the zero-crossing with peak amplitude (ZCPA) algorithm to guarantee the high performance of the ASR system using only a throat microphone. For ZCPA optimization, we propose an intensification of the formant frequencies and a selection of cochlear filters. Experimental results show that this system yields a performance improvement of about 4% and a reduction in time complexity of 25% when compared to the performance of a standard ZCPA algorithm on throat microphone signals.

• Journal of Sensor Science and Technology
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• v.16 no.1
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• pp.62-67
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• 2007

This paper presents a novel MEMS condenser microphone with rigid backplate to enhance acoustic characteristics. The MEMS condenser microphone consists of membrane and backplate chips which are bonded together by gold-tin (Au/Sn) eutectic solder bonding. The membrane chip has 2.5 mm${\times}$2.5 mm, $0.5{\mu}m$ thick low stress silicon nitride membrane, 2 mm${\times}$2 mm Au/Ni/Cr membrane electrode, and $3{\mu}m$ thick Au/Sn layer. The backplate chip has 2 mm${\times}$2 mm, $150{\mu}m$ thick single crystal silicon rigid backplate, 1.8 mm${\times}$1.8 mm backplate electrode, and air gap, which is fabricated by bulk micromachining and silicon deep reactive ion etching. Slots and $50-60{\mu}m$ radius circular acoustic holes to reduce air damping are also formed in the backplate chip. The fabricated microphone sensitivity is $39.8{\mu}V/Pa$ (-88 dB re. 1 V/Pa) at 1 kHz and 28 V polarization voltage. The microphone shows flat frequency response within 1 dB between 20 Hz and 5 kHz.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.12 s.117
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• pp.1272-1278
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• 2006

This paper presents a novel MEMS condenser microphone with rigid backplate to enhance acoustic characteristics. The MEMS condenser microphone consists of membrane and backplate chips which are bonded together by gold-tin(Au/Sn) eutectic solder bonding. The membrane chip has $2.5mm{\times}2.5mm$, 0.5${\mu}m$ thick low stress silicon nitride membrane, $2mm{\times}2mm$ Au/Ni/Cr membrane electrode, and 3${\mu}m$ thick Au/Sn layer. The backplate chip has $2mm{\times}2mm$, 150${\mu}m$ thick single crystal silicon rigid backplate, $1.8mm{\times}1.8mm$ backplate electrode, and air gap, which is fabricated by bulk micromachining and silicon deep reactive ion etching. Slots and $50{\sim}60{\mu}m$ radius circular acoustic holes to reduce air damping are also formed in the backplate chip. The fabricated microphone sensitivity is 39.8 ${\mu}V/Pa$(-88 dB re. 1 V/Pa) at 1 kHz and 28 V polarization voltage. The microphone shows flat frequency response within 1 dB between 20 Hz and 5 kHz.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.14 no.10
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• pp.4753-4761
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• 2013

This study recorded the sound of Korean traditional musical instruments by considering the radiation characteristics of traditional Korean musical instruments and setting the diverse mounting angle and distance of microphone. By following recorded results, the study analyzes the optimum position of microphone and the characteristic of timbre through frequency range spectrum and repeated listening. This experiment of study is for technique of close microphone which is used in popular music, therefore it is recorded in the popular music recording studio. The study also suggests the optimum position for the microphone to record, and explains the difference of timbre at different position. The result of study will suggest the best technique of Close Microphone Techniques for Korean Traditional Musical Instruments not only to accurately express the beautiful and unique sound of traditional Korean musical instruments but also to obtain diverse timbre.

• Journal of rehabilitation welfare engineering & assistive technology
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• v.2 no.1
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• pp.27-33
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• 2009

Generally, implantable microphone has been implanted in the temporal bone for implantable middle ear hearing devices (IMEHDs). In this case, the microphone's membrane can be damaged and can be generated biological noise. In order to overcome the these problems, the location of implanted microphone should be changed. As an alternative, the microphone can be implanted in the external auditory canal. However, the sound emission can be produced because of vibration transducer toward reverse direction from the tympanic membrane to the external auditory canal. In this paper, an amount of the emitted sound is measured using a probe microphone as the changing the position of microphone in the external auditory canal of a physical ear model, which is similar to acoustical and vibratory properties of the human ear. Through the measured value, the location of the microphone was assumed in the external auditory canal. According to the analysis, the microphone input sound can be decreased when microphone position become more distance from the tympanic membrane in the auditory canal. However, the external auditory canal is not appropriated to implantable microphone position, because sound emission is not completely eliminated.

• The Journal of the Acoustical Society of Korea
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• v.17 no.4E
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• pp.22-27
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• 1998

A modified perturbation method is proposed to optimize the beam pattern of thinned microphone arrays. Both microphone spacing and array weight are iteratively adjusted via successive perturbation to achieve an optimum beam pattern in a Dolph Chebyshev sense. To improve the sidelobe performance, an alternative perturbation with respect to microphone spacing and array weight is implemented. Also, a linear space-tapering is employed in the perturbation process. It is demonstrated that the proposed approaches successfully yield sidelobe performances comparable to that of a normal array. Computer simulation results are presented.