• Title/Summary/Keyword: speaker-microphone characteristics

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Dynamic Characteristic Analysis and Transfer Function Estimate of Acoustic System for Transformer Noise Control (변압기 소음제어를 위한 음향 시스템의 동특성 해석 및 전달함수 추정)

  • 김영달;정창경;심재명
    • Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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    • v.13 no.3
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    • pp.17-24
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    • 1999
  • This paper presents a method of ANC for transfonrer noise control utilizing a sproker and microphone pair. In this study, the main focus is on identifying the dynamic characteristics of speaker - amplifier microphone path. This study presents a theoretical method to identify the dynamic characteristics of speaker-microphone pairs. The transfer functions of microphone - speaker pair have been estimated utilizing sequential least square(SLS) algorithm. We identified the estimated transfer function has stable JXlles and zeros in z-plane. This paper also propose an architecture far the noise cancellation to which we applied the estimated transfer function.nction.

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Implementation of Active Noise Control with DSP56001 (DSP56001을 이용한 능동소음제어의 구현)

  • Kim, Young-Hoon;Park, Jang-Kwan;Koo, Choon-Keun;Chung, Chan-Soo
    • Proceedings of the KIEE Conference
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    • 1998.07b
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    • pp.654-656
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    • 1998
  • This paper deal with the implementation of Active Noise Control (ANC) in a short duct. In case of ANC in the air duct, input microphone, control speaker, error microphone are used. But we can't use input microphone because of the characteristics of short duct. It is difficult to avoid howl. So we propose single-channel adaptive feedback ANC which is composed only error microphone and control speaker without input microphone. FXLMS algorithm is used to compensate for the time delay of the error path. Experimental results show that the controller reduce noise signal sufficiently.

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A Single Sensor Active Noise Control Considering The Characteristics of The Speaker and The Microphone (스피커와 마이크의 전달특성을 고려한 단일 센서 능동소음제어)

  • 김현태;박장식
    • Journal of Korea Multimedia Society
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    • v.6 no.7
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    • pp.1131-1138
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    • 2003
  • Active noise control(ANC) is an approach to noise reduction in which a secondary noise source destructively interferes with the unwanted noise is introduced. Generally, the performance of ANC is determined how well a secondary noise tracks noises. A secondary noise is generated from the cancelling speaker and a error sensor pick up error signal. The transfer function between the cancelling speaker and the error sensor is not flat and distorts secondary noises. Consequently, the performance of ANC is degraded by the transfer function. In this paper, a single sensor ANC which considers the characteristics of the speaker and the error sensor is proposed. To reduce distortion of secondary noises, the transfer function is estimated by adaptive inverse modelling and the primary noises are estimated by Kalman filter. Experimental results show that the proposed single sensor ANC effectively attenuates noises.

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Optimal Selection of Microphone and Speaker Positions in Adaptive Noise Control Systems (능동소음제어 시스템의 마이크로폰 및 스피커 최적위치 선정)

  • Lee, Hong-Won;Seo, Sung-Dae;Nam, Hyun-Do
    • Proceedings of the KIEE Conference
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    • 2003.07d
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    • pp.2766-2768
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    • 2003
  • In this paper, positions of microphones and speakers in adaptive noise control systems are selected using optimization techniques. To get the optimal control characteristics of adaptive noise control system, It is necessary to optimize positions of microphones and speakers. Assume that control speakers are placed in a experimental enclosure several place indoors. Acoustic transfer functions from control speaker positions to microphone positions are measured for simulation program by experiments, and simulated annealing methods are used to select optimal positions of speakers and microphones.

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Design of the broadband and compact phase-calibrator for array microphones (어레이 마이크로폰용 광대역 소형 위상교정기의 설계)

  • Ju, Hyeong-Sick;Kim, Yang-Hann
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2004.11a
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    • pp.1032-1035
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    • 2004
  • Pressure distribution is measured by way microphones to identify noise sources in the space. For example, beam-forming method or acoustic holography use phase information to identify the source. Therefore, the phase is significant information to correctly identify the source position. However, due to the microphone characteristics and measuring systems, measured signals always have errors, which make the identification difficult. Therefore, phase calibration of microphones is needed. Duct and speaker systems are generally used as calibrators. Acoustic characteristics of the calibrator are, of course, functions of many Parameters of the system: i.e. duct size, frequency, and microphone spacing. In this paper, design parameters which effect on the performance and size of the calibrators are considered. Then the parameters would be applied to design and real product of the phase-calibrator.

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The Design of Temporal Bone Type Implantable Microphone for Reduction of the Vibrational Noise due to Masticatory Movement (저작운동으로 인한 진동 잡음 신호의 경감을 위한 측두골 이식형 마이크로폰의 설계)

  • Woo, Seong-Tak;Jung, Eui-Sung;Lim, Hyung-Gyu;Lee, Yun-Jung;Seong, Ki-Woong;Lee, Jyung-Hyun;Cho, Jin-Ho
    • Journal of Sensor Science and Technology
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    • v.21 no.2
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    • pp.144-150
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    • 2012
  • A microphone for fully implantable hearing device was generally implanted under the skin of the temporal bone. So, the implanted microphone's characteristics can be affected by the accompanying noise due to masticatory movement. In this paper, the implantable microphone with 2-channels structure was designed for reduction of the generated noise signal by masticatory movement. And an experimental model for generation of the noise by masticatory movement was developed with considering the characteristics of human temporal bone and skin. Using the model, the speech signal by a speaker and the artificial noise by a vibrator were supplied simultaneously into the experimental model, the electrical signals were measured at the proposed microphone. The collected signals were processed using a general adaptive filter with least mean square(LMS) algorithm. To confirm performance of the proposed methods, the correlation coefficient and the signal to noise ratio(SNR) before and after the signal processing were calculated. Finally, the results were compared each other.

The comparison of the voice between the free field and the external auditory canal (음장과 외이도 내부에서의 음성 비교)

  • Heo, Seung-Deok;Kim, Lee-Suk;Ko, Do-Heung;Lee, Jung-Hak
    • Speech Sciences
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    • v.7 no.4
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    • pp.83-90
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    • 2000
  • The purpose of this study was to examine some acoustic characteristics in the ear canal. It was assumed that a sound outside the external auditory canal could be different from the sound inside the external auditory canal. The acoustic signals were captured by a probe microphone placed at a distance within 1 cm from the tympanic membrane, and a reference microphone was placed over the upper pinna. Three vowels /a/, /i/, /u/ were recorded from a normal adult male speaker. The parameters such as the formant frequency ($Fl\simF5$) and the peak intensity were measured using a speech analyser, PCquirer. It was found that the entering part of the external auditory canal functions as a narrowing point as to the speech that passes through the free field. Results show that acoustic characteristics were changed for speech discrimination rather than speech perception.

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The Measurement Algorithm for Microphone's Frequency Character Response Using OATSP (OATSP를 이용한 마이크로폰의 주파수 특성 응답 측정 알고리즘)

  • Park, Byoung-Uk;Kim, Hack-Yoon
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.2
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    • pp.61-68
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    • 2007
  • The frequency response of a microphone, which indicates the frequency range that a microphone can output within the approved level, is one of the most significant standards used to measure the characteristics of a microphone. At present, conventional methods of measuring the frequency response are complicated and involve the use of expensive equipment. To complement the disadvantages, this paper suggests a new algorithm that can measure the frequency response of a microphone in a simple manner. The algorithm suggested in this paper generates the Optimized Aoshima's Time Stretched Pulse(OATSP) signal from a computer via a standard speaker and measures the impulse response of a microphone by convolution the inverse OATSP signal and the received by the microphone to be measured. Then, the frequency response of the microphone to be measured is calculated using the signals. The performance test for the algorithm suggested in the study was conducted through a comparative analysis of the frequency response data and the measures of frequency response of the microphone measured by the algorithm. It proved that the algorithm is suitable for measuring the frequency response of a microphone, and that despite a few errors they are all within the error tolerance.

Sound Detection Characteristics Using Fabry-Perot Fiber Optic Sensor which Simply Supported in Structure (양단이 지지된 Fabry-Perot 광섬유센서의 음압 감지 특성 연구)

  • 이종길;이진우;이준호
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.7
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    • pp.585-591
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    • 2003
  • In this paper, fiber optic sensor using Fabry-Perot interferometer which had benefit of minimize and light-weight was used. The sensor head has 1cm in length, total length of fiber is 9.5 chi and the sensor supported at both ends, simply. To analyze the acoustic characteristic non-directional speaker is used as a sound source. Acoustic applied in lateral direction and detected two signals were compared each other. Below 1㎑ fiber optic sensor has more sensitive than microphone, but in 2㎑ fiber optic sensor has less sensitive than microphone. This characteristic varies to the supporting system of fiber optic sensor. It was confirmed that the Fabry-Perot interferometric sensor detected acoustic signal, effectively. This kind of sensor can be applied to the structural health monitoring field of intellectual structure.

Thermoacoustic Refrigerating System, Part II : Implementation and Experiment (열음향 냉장시스템 (II) : 제작 및 실험)

  • Hah, Zae-Gyoo;Ahn, Chul-Yong;Sung, Keong-Mo
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.6
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    • pp.13-20
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    • 1995
  • In this paper, the thermoacoustic refrigerating system was implemented and its operation was experimentally verified. The system is composed of several parts ,4 inch midrange speaker, speaker housing, chamber, stack housing, stack of plates, heat exchangers, thin pipe and cavity. The system is filled with He gas at 10 bar and contains T-type thermocouples and condenser microphone for measuring the temperature and pressure inside, respectively. In addition, cooling water is used for protecting speaker from thermal destruction and cooling down the hot heat exchanger. For the experimental verification of the implemented refigerating system, electrical impedance and resonance characteristics were measured. The results showed that it was most efficient to drive the system at 340 Hz. When operated at 340 Hz, $30^\circ{C}$ environments and 50 electical watts, the temperature of the cold region decreased by $16^\circ{C}$. The dissatisfaction mainly comes from the incomplete thermal insulation of the cold region. We also pointed out some guidelines to improve the performance for later study.

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