• Proceedings of the Acoustical Society of Korea Conference
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• 1985.10a
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• pp.37-42
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• 1985

The errors caused by developmental finite difference approximations in the two microphone acoustic intensity measurement technique are considered in this paper. The frequencies and pressure responses obtained experimentally, are presented for both face to face (FF) and side by side(SS) microphone configurations. The implications of these results by the use of such a cofiguration for two microphone sound intensity measurement are discussed.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.883-888
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• 2002

This study describes the limit and application of the two-microphone impedance tube method to the sound transmission loss measurement of several sound isolation materials with different physical properties. For the sound isolation materials having small flexural rigidity, it is shown that the two-microphone impedance tube method is validated to practically measure the sound transmission loss. For the sound isolation materials having large flexural rigidity, on the other hand, it is found that the two-microphone impedance tube method is no longer valid to measure the sound transmission loss because the regions of resonance and mass law are moved into the higher frequencies. In addition, in order to accurately measure the sound transmission loss of sound isolation materials, their size should be decided based on the consideration of the effect of acoustic excitation on their vibration response.

• Journal of Information Technology Applications and Management
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• v.23 no.3
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• pp.87-97
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• 2016

This paper proposes a fast sound source localization method using a TDOA sign-based algorithm. We present an L-shaped microphone set-up which creates four major regions in the range of $0^{\circ}{\sim}360^{\circ}$ by the intersection of the positive and negative regions of the individual microphone pairs. Then, we make an initial source region prediction based on the signs of two TDOA estimates before computing the azimuth value. Also, we apply a threshold and angle comparison to tackle the existing front-back confusion problem. Our experimental results show that the proposed method is comparable in accuracy to previous three microphone array methods; however, it takes a shorter computation time because we compute only two TDOA values.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.24 no.2E
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• pp.38-45
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• 2005

The noise reduction of the In-The-Ear (ITE) hearing aid (HA) can be achieved by arrays of microphones. Each of the right and the left ears was assumed to have two HA microphones. These arrays of HA microphones produce particular patterns of directivity by some time delay between two microphones. The directivity pattern geometrically increase the S/N ratio. The boundary element method (BEM) was used for the three dimensional simulation of the HA directivity pattern with the two pairs' microphone arrays. The separation between two microphones was fixed to 10 mm. The time delay between the two microphones was calculated to produce the most narrow directivity pattern in the fore front of the head. The variation of the time delay was examined in accordance with input frequencies. This numerical analysis may be then applied for the calculation of the time delay parameter of the digital hearing aid DSP chip.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.4
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• pp.343-352
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• 2012

To follow a sound source by a mobile robot, the relative position and orientation of the sound source from the mobile robot have been estimated using a microphone array. In this research, the difference among the traveling times of the sound source to each of three microphones has been used to calculate the distance and orientation of the sound source from the mobile robot which carries the microphone array. The cross-correlation between two signals has been applied for detecting the time difference between two signals, which provides reliable and precise value of the time difference comparing to the conventional methods. To generate the tracking direction to the sound source, fuzzy rules are applied and the results are used to control the mobile robot in a real-time. The efficiency of the proposed algorithm has been demonstrated through the real experiments comparing to the conventional approaches.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.1661-1666
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• 2000

Nowadays, the two microphone method is accepted as the standard as specified in ASTM E1050-90 for measuring in-duct acoustic properties. However, research results on using the least square method with multiple measurement points and broadband excitation have been reported for enhancing the frequency response of the two microphone method. In this paper, the effects of varying the relative measurement positions on errors in the estimation of the acoustic quantities is studied for the multiple microphone method. Both of the theoretical and experimental results show that, among every possible sensor positioning configurations, the equidistant positioning of sensors yields the smallest error within the effective measurement frequency range. In addition, it is noted that the measurement accuracy can be increased and the effective frequency range can be widened by increasing the number of equidistant sensors. Measurement examples are shown and the results support the findings.

• 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.

• Journal of Advanced Marine Engineering and Technology
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• v.11 no.1
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• pp.63-71
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• 1987

The measuring of acoustic intensity has been investigated by many researchers and practicians during the last several decades. But due to the lack of measurement accuracy, they have had no practical use. In recent years, the two microphone acoustic intensity method has been developed by the advancement of FFT analysis technique and the digital data processing equipment. This new method of using two microphones gives informations on the noise source survey and the acoustic power of sound radiation source without the anechoic room. In this paper, theoretical formulae for the two microphone acoustic intensity method and the sound transmission loss are checked. The obtained results for the acoustical enclosure of gas heat pump were compared with the classical field incidence mass law. The surface vibration modes for a panel of enclosure were also estimated.