• 제어로봇시스템학회:학술대회논문집
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• 1999.10a
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• pp.266-269
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• 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.

• The Journal of the Acoustical Society of Korea
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• v.4 no.1
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• pp.35-42
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• 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.

• KEPCO Journal on Electric Power and Energy
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• v.1 no.1
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• pp.99-104
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• 2015

In this study, it has been initiated to investigate the specific abnormal vibration signal that has been captured in the power equipment. Array Microphone can be used in order to detect the direction and the position of the noise source. It is possible to track the abnormal mechanical noise in the power plant by utilizing the program and the microphone array system developed from this research. Array microphone system can be operated as a constant monitoring system.

• 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.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.1 no.1
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• pp.21-27
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• 2008

In this paper, a PC-based real-time microphone array system with small aperture is presented. The microphone array system is based on the generalized sidelobe canceler (GSC) but it employs a new adaptation mode controller (AMC). The performance of the proposed system was evaluated in the Multimedia Room modeled on an office situation. Evaluation experiments show that the proposed system can perform with stable noise suppression.

• Journal of KIISE
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• v.43 no.1
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• pp.13-26
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• 2016

This paper proposes a new sound localization algorithm that can improve localization performance based on a tetrahedron-shaped microphone array. Sound localization system estimates directional information of sound source based on the time delay of arrival(TDOA) information between the microphone pairs in a microphone array. In order to obtain directional information of the sound source in three dimensions, the system requires at least three microphones. If one of the microphones fails to detect proper signal level, the system cannot produce a reliable estimate. This paper proposes a tetrahedron- shaped sound localization system with a coordinate transform method by adding one microphone to the previously known triangular-shaped system providing more robust and reliable sound localization. To verify the performance of the proposed algorithm, a real time simulation was conducted, and the results were compared to the previously known triangular-shaped system. From the simulation results, the proposed tetrahedron-shaped sound localization system is superior to the triangular-shaped system by more than 46% for maximum sound source detection.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2011.10a
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• pp.717-722
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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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.11a
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• pp.735-741
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• 2003

A simplified view of array design and application process was introduced. Array design is critical to achieve a successful phased array measurements. A planar microphone array is designed to produce optimum performance and also to fit economic requirement in integrating data acquisition system. Certain performance characteristics are of primary concern when designing arrays. These characteristics include array resolution, spatial aliasing and array sidelobe suppression. Every array has its directional pattern that shows such characteristics. Assuming that a monopole source is located in center, beam-patterns have been simulated varying measurement conditions such as number of sensors. array aperture size, distance between array and source, frequency of interest and so on. Sensor correction was conducted on very channel using magnitudes and phased of FRF with respect to a reference microphone channel. Then with a spiral type array, measurements have been made with two point sources of same frequency in order to investigate array resolving abilities. It is observed that higher frequency source achieves better resolution than lower one does.

• 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.

• 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.