• Title/Summary/Keyword: microphone

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A Experimental study on frequency characteristics of the microphone array covered with Kevlar in closed test section wind tunnel (폐쇄형 시험부에서 케블라 덮개가 장착된 마이크로폰 어레이의 주파수 특성에 대한 실험적 연구)

  • Hwang, Eun-sue;Choi, Youngmin;Han, Huyngsuk;Kim, Yangwon;Cho, Taehwan
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2014.04a
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    • pp.128-134
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    • 2014
  • A Experimental study on frequency characteristics of the microphone array covered with Kevlar in closed test section wind tunnel. Microphones that are flush mounted in a closed test section wall of wind tunnel are subject to very high flow noise resulting from the turbulence in the wall boundary layer. At this time the microphones measure the strong hydrodynamic fluctuations generated by the flow. The phenomena are referred to a microphone self-noise and a method for reducing this has studied. In this paper the array that covered with acoustically transparent Kevlar sheet was designed and made to reduce the flow-induced self-noise. For the validation frequency characteristics of the Kevlar, the microphone array was installed on the wall and test was performed for white noise and sine wave of several frequencies using loudspeaker. In addition, the paper compared the signals as a tension of Kevlar. The results were presented that tend to decrease the sound pressure level at high frequency above 3500Hz according to existence of Kevlar.

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The Limit and Application of Two-microphone Impedance Tube Method to the Sound Transmission Loss Measurement of Sound Isolation Materials (차음재의 음향투과손실 측정에 Two-Microphone Impedance Tube Method의 적용과 한계)

  • Lee, Seung;Ahn, Min-Hong;Lee, Dong-Hoon
    • 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.

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Designing a Microphone Array System for Noise Measurements on High-Speed Trains (고속철도 차량의 소음 측정을 위한 마이크로폰 어레이 설계 연구)

  • Noh, Hee-Min;Choi, Sung-Hoon;Hong, Suk-Yoon;Kim, Seog-Won
    • Journal of the Korean Society for Railway
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    • v.14 no.6
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    • pp.477-483
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    • 2011
  • In this paper, noise source localization of the Korean high speed train was conducted by using delay and sum beam-forming method of a microphone array. At first, the microphone array having irregular configuration was designed and the resolution of which was analyzed from parameters such as 3-dB bandwidth and maximum side-lobe level. After the demonstration, the microphone array was applied on the high speed train and noise localization of the high speed train driving at 300 km/h was performed successfully.

Active Audition System based on 2-Dimensional Microphone Array (2차원 마이크로폰 배열에 의한 능동 청각 시스템)

  • Lee, Chang-Hun;Kim, Yong-Ho
    • 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.

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DSP Implementation of Speech Enhancement System Using Microphone Array with Adaptive Post-processing (적응 후처리 과정을 갖는 마이크로폰 배열을 이용한 잡음제거기의 DSP 구현)

  • 권홍석;김시호;배건성
    • Proceedings of the IEEK Conference
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    • 2002.06d
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    • pp.413-416
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    • 2002
  • In this paper, a speech enhancement system using microphone array with adaptive Post-Processing is implemented in real-lime with TMS320C6201 DSP. It consists of delay-and-sum beamformer and adaptive post-processing filters with NLMS (Normalized Least Mean Square) algorithm. THS1206 ADC is used for collection of 4-channel microphone signals. Sizes of program memory, data ROM and data RAM of the implemented system are 15,744, 748 and 47,540 bytes, respectively. Finally 21.839${\times}$106 clocks per second is required for real-time operation.

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A User friendly Remote Speech Input Unit in Spontaneous Speech Translation System

  • Lee, Kwang-Seok;Kim, Heung-Jun;Song, Jin-Kook;Choo, Yeon-Gyu
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2008.05a
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    • pp.784-788
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    • 2008
  • In this research, we propose a remote speech input unit, a new method of user-friendly speech input in speech recognition system. We focused the user friendliness on hands-free and microphone independence in speech recognition applications. Our module adopts two algorithms, the automatic speech detection and speech enhancement based on the microphone array-based beamforming method. In the performance evaluation of speech detection, within-200msec accuracy with respect to the manually detected positions is about 97percent under the noise environments of 25dB of the SNR. The microphone array-based speech enhancement using the delay-and-sum beamforming algorithm shows about 6dB of maximum SNR gain over a single microphone and more than 12% of error reduction rate in speech recognition.

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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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Performance of Different Sensors for Monitoring of the Vibration Generated during Thermosonic Non-destructive Testing

  • Kang, Bu-Byoung
    • Journal of the Korean Society for Nondestructive Testing
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    • v.31 no.2
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    • pp.111-117
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    • 2011
  • Vibration monitoring is required for reliable thermosonic testing to decide whether sufficient vibration is achieved in each test for the detection of cracks. From a practical point of view, a cheaper and convenient monitoring method is better for the application to real tests. Therefore, the performance of different sensors for vibration monitoring was investigated and compared in this study to find a convenient and acceptable measurement method for thermosonics. Velocity measured by a laser vibrometer and strain provide an equivalent HI when measured at the same position. The microphone can provide a cheaper vibration monitoring device than the laser and the heating index calculated by a microphone signal shows similar characteristics to that calculated from velocity measured by the laser vibrometer. The microphone frequency response shows that it underestimates high frequency components but it is applicable to practical tests because it gives a conservative value of HI.

Considering Microphone Positions in Sound Source Localization Methods: in Robot Application (로봇 플랫폼에서 마이크로폰 위치를 고려한 음원의 방향 검지 방법)

  • Kwon, Byoung-Ho;Kim, Gyeong-Ho;Park, Young-Jin
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2007.05a
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    • pp.1080-1084
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    • 2007
  • Many different methods for sound source localization have been developed. Most of them mainly depend on time delay of arrival (TDOA) or on empirical or analytic head related transfer functions (HRTFs). In real implementation, since the direct path between a source and a sensor is interrupted by obstacles as like a head or body of robot, it has to be considered the number of sensors as well as their positions. Therefore, in this paper, we present the methods, which are included sensor position problem, to localize the sound source with 4 microphones to cover the 3D space. Those are modified two-step TDOA methods. Our conclusion is that the different method has to be applied in case to be different microphone position on real robot platform.

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