• KSII Transactions on Internet and Information Systems (TIIS)
• /
• 제6권3호
• /
• pp.906-918
• /
• 2012

Wireless Fidelity (WiFi) and Wireless microphone are assumed to operate on adjacent channels in TV White Spaces(TVWS). The Scenario of WiFi potentially interfering with Wireless microphone is analyzed through Minimum Coupling Loss (MCL) and Spectrum Engineering Advanced Monte Carlo Analysis Tool (SEAMCAT) based on the Monte-Carlo simulation method. In the case of single WiFi interfering with Wireless microphone, the protection distance between WiFi and the Wireless microphone should be at least 25.12 m to avoid WiFi impact on Wireless microphone. When the active number of WiFi is 12, the guard band between WiFi and Wireless microphone should not be less than 4.97 MHz to guarantee that WiFi does not interfere with the Wireless microphone.

• Transactions on Electrical and Electronic Materials
• /
• 제9권3호
• /
• pp.128-133
• /
• 2008

In this paper, in order to analyze property of frequency response in microphone using optical sensor, acousto-optic sensor system has been implemented. The capacitance microphone and fiber-optic transmission path type fiber-optic microphone (FOM) have weaknesses in directivity, size, weight, and price. However suggested optical microphone can be constituted by cheap devices, so it has many benefits like small size, light weight, high directivity, etc. Head part of optical microphone which is suggested in this paper is movable back and forth by sound pressure with the attached reflection plate. Operating point has also been determined by measuring the response characteristics. The choosing the point, which has maximum linearity and sensitivity has changing the distance between optical head and vibrating plate. We measured the output of the O/E transformed signal of the optical microphone while frequency of sound signal is changed using sound measurement /analysis program, "Smaart Live" and "USBPre", which are based on PC, and compared the result from an existing capacitance microphone. The measured optical microphone showed almost similar output characteristics as those of the compared condenser microphone, and its bandwidth performance was about 4 kHz at up to 3 dB.

• 한국콘텐츠학회논문지
• /
• 제8권6호
• /
• pp.8-15
• /
• 2008

본 논문에서는 광 마이크로폰의 주파수 응답특성 분석을 위해 시스템을 제안하고 구현하였다. 현재 사용되어지고 있는 캐패시턴스 마이크로폰과 광섬유 전송로형 광 마이크로폰(FOM)은 낮은 지향성과 크기, 무게, 비싼 가격 등에서 단점을 나타내고 있다. 반면 제안된 광 마이크로폰은 저가의 소자로 구성할 수 있으며, 소형, 경량화, 높은 지향성 등의 장점을 갖는다. 본 논문에서 제안하는 광 마이크로폰의 헤드 부는 인가된 음압에 따라서 전후로 움직이게 되어있으며, 반사판이 부착되어 있는 형태로 구성되어 있다. 광 헤드와 진동판사이의 거리를 변화시켜가며 응답특성을 측정하여 선형성과 민감도가 최대인 지점을 동작 점으로 결정하였다. 컴퓨터 기반 음향 측정/분석 프로그램인 Smaart Live와 USBPre를 사용하여 음향 신호의 주파수를 변화시켜 가며 구현한 광 마이크로폰의 광전 변환된 신호의 출력을 측정하였으며, 기존의 캐패시턴스 마이크로폰과 비교 분석하였다. 측정된 광 마이크로폰은 비교 콘덴서 마이크로폰과 거의 동일한 출력 특성을 나타내었으며, 주파수 대역폭은 약 ${\pm}3$[dB] 이내에서 300[Hz]-3[kHz] 정도의 성능을 나타내었다.

• 대한의용생체공학회:의공학회지
• /
• 제26권3호
• /
• pp.139-144
• /
• 2005

An implantable microphone that can be utilized as part of a totally implantable hearing aid is designed and implemented. The proposed microphone is implanted in the center of the pinna, and designed to ensure the speech frequency range and the appropriate sensitivity. The characteristics of the proposed microphone are evaluated using a finite element analysis (FEA). The microphone is composed of a small electric condenser microphone, titanium case 6.2mm in diameter and 3mm high, and $10{\mu}m$ SUS316L vibrating membrane in contact with hypodermic tissue to maintain the sensitivity of the microphone. The microphone components are all made of biocompatible materials, then the assembled microphone is hermetically sealed using a polymer and ceramic. Experiments with the fabricated microphone confirm an operational bandwidth of up to 5kHz without any decline of sensitivity in 6mm of hypodermic tissue.

• 제어로봇시스템학회:학술대회논문집
• /
• 제어로봇시스템학회 1999년도 제14차 학술회의논문집
• /
• pp.266-269
• /
• 1999

An electronically Scannable microphone system is in the Planning stage. For this Purpose, a multiple microphone array with controllable delay is available. To achieve effective point-listening characteristics, we proposed an enclosed microphone array system with a complex weighting method. In this system, both the microphone arrangement and the value of the complex weighting are important. In this report, the construction of microphone array system and the signal-processing method are explained, and the calculation method for optimal complex weighting is also presented. A prototype experimental setup is designed and fabricated to verify the expected characteristics.

• 한국음향학회지
• /
• 제4권1호
• /
• pp.35-42
• /
• 1985

This paper suggests a method on the use of very long linear array microphone in order to obtain clear loud sound reinforcement system without howling. According to the results of the theoretical investigation, we have made that a linear array microphone. This is made of one hundred small condenser microphone having 2 cm of spatical period. To estimate the effect of sound reinforcement system physically and subjectively, four cases have been experimented : In case of using no sound reinforcement, nondirectivity microphone, rectangular window and Hnning window in linear array microphone. The experimental results prove that the case of Hnning window in linear array microphone is more excellent.

• 한국음향학회:학술대회논문집
• /
• 한국음향학회 1985년도 학술발표회 논문집
• /
• pp.37-42
• /
• 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.

• The Journal of the Acoustical Society of Korea
• /
• 제17권4E호
• /
• pp.41-47
• /
• 1998

Microphone is the first link in the speech recognition system. Depending on its type and mounting position, the microphone can significantly distort the spectrum and affect the performance of the speech recognition system. In this paper, characteristics of the speech signal for different microphones and microphone distances are investigated both in time and frequency domains. In the time domain analysis, the average signal-to-noise ration is measure ration is measured for the database we collected depending on the microphones and microphone distances. Mel-frequency spectral coefficients and mel-frequency cepstrum are computed to examine the spectral characteristics. Analysis results are discussed with our findings, and the result of recognition experiments is given.

• 한국수산과학회지
• /
• 제13권4호
• /
• pp.145-149
• /
• 1980

Various methods have been so far proposed to find out the directions and spectra of sound waves from the sources for provisions of noise controls. The conventional methods are generally classified into three systems such as, single microphone system, moving microphone system and multi-microphone system, which composes a resultant super directivity by giving a appropriate delay and a weighting coefficient in the output of each microphone. In case of using a single microphone there is a difficulty in providing it with desirable super directivity in the low frequency range, while in case of using multi-microphone system there has been a disadvantage that the measurement of directivity could not separately be done with the spectrum analysing. And in case of the use of a moving microphone system it needs a condition that the sound source to be detected should be stationary state and in rest. However here we introduce a method that the spectral analysing and the directivity of synthesis can be separately carried out by using a linear array of many microphones, in which each output of the microphone is multiplied by appropriate weighting coefficient and all of those products are summed after passing through adequate filters. The resultant signal is then sampled with an adequate sampling frequency and taken average for processing.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2000년도 추계학술대회논문집
• /
• pp.731-736
• /
• 2000

The recently developed BEM-based NAH(nearfield acoustical holography) is a useful technique for identifying the sound source of vibrating objects. The acoustic parameters of a sound source can be reconstructed by using the vibro-acoustic transfer matrix, which is determined by means of BEM, and the sound pressure measured in the nearfield. Theoretically, one can come up with a very nice reconstructed result as the field plane gets near to the source surface. However, when a microphone is placed in the very close nearfield of the source surface, the scattering, reflection, or resonance in the gap between the source and the microphone can distort the acoustic field, and therefore, the measured field pressure would differ from the actual one in the absence of the microphone. In order to analyze this problem, the interference effect of the microphone is numerically calculated by using the nonsingular BEM that yields very small error in the nearfield. From this analysis, it is found that the prediction error of the field pressure decreases firstly and then increases as the microphone approaches the vibrating surface from the farfield to the close nearfield. It is noted that the microphone should be separated from the source surface by at least a diameter of the microphone for an error ratio less than 2% in the low frequency range less than about 2.7kHz. This means that if one wants to put a microphone in the very close nearfield. a microphone with small diameter should be used.