• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.25 no.3
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• pp.150-159
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• 2015

An experimental study on frequency characteristics of the microphone array covered with Kevlar sheet was conducted in the closed test section. Microphones that were flush-mounted in the wall of wind tunnel were subjected to very high flow noise resulting from the turbulence in the wall boundary layer. This noise interference by the boundary layer was referred as 'a microphone self-noise' and various approaches were studied to reduce this interference. Recessed microphone array with high tensioned cover was one of the good approaches to reduce this self-noise. But, the array cover could cause an unexpected interference to the measuring results. In this paper the frequency characteristics of the microphone array with Kevlar cover was experimentally studied. The white noise was used as a reference noise source. Three kinds of tensions for the Kevlar cover were tested and those results were compared with the test results without the Kevlar cover. The gap effect between the cover and microphone head was also tested to find out the proper position of microphone in the array module. Test results show that the mid-tension and 10mm gap was the best choice in the tested cases.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.10
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• pp.1448-1451
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• 2021

In this paper, 3D range sensor is applied to the sensor-based widerange microphone system for lectures. Since the 2D range sensor measures the shortest distance of the speaker, an error occurs and the performance is degraded. The 3D sensor provides a 160×60 distance image so that the position of the speaker can be obtained with accuracy. We propose a method for obtaining the distance per pixel required to determine the absolute position of the speaker from the distance image. The proposed array microphone system using the 3D sensor shows the improvement of 0.8~1.5dB compared to the previous works using 2D sensor.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.4
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• pp.624-633
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• 2018

While the multimedia lecture system has been getting smart using immerging technology, the microphone still relies on the classical approach such as holding in hand or attaching on the body. In this paper, we propose a ceiling mounted array microphone system that allows a wide reception coverage and instructors to move freely without attaching microphone. The proposed system adopts cell and handover of mobile communication instead of a complicated beamforming method and implements a wide range microphone over several cells with low cost. Since the characteristics of unvoiced speech is similar to Pseudo Noise it is shown that soft handover are possible with 3 microphones connected to delay-sum multipath receiver. The proposed system is tested in $6.3{\times}1.5m$ area. For real-time processing the correlation range can be reduced by 82% or more, and the output latency delay can be improved by using the delay adaptive filter.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2001.11b
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• pp.1335-1340
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• 2001

The information technology brought us many kinds of conveniences to our life, but it also caused social problems such as privacy interference, unexpected personal information leaks, and nose generation by telephonic talks, etc. In this paper, the microphone system of the cellular phone is developed to prevent these problems caused by progress of information technology. The developed system was designed to detect only acoustic signals from a human being in the presence of various kinds of background noises. A windscreen was designed by use of micro-channels to eliminate the popping noise by the wind from the mouth of a speaker and four microphone array and signal processing techniques are applied to reduce background noise. The impact of the developed system was evaluated by experimental tests. The results show that the system can improve the required functions considerably.

• Korean Journal of Fisheries and Aquatic Sciences
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• v.13 no.4
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• pp.145-149
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• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.54-60
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• 2005

In this Paper, an adaptive microphone array system with self-delay estimator is proposed. By showing that the adaptive blocking matrix (ABM) of the generalized sidelobe canceller (GSC) can estimate the relative time delay between each sensor, the proposed system utilizes the ABM not only for blocking target components in the blocked signal path, but also for estimating the relative time delay. Therefore, the proposed system requires only the GSC structure while maintaining the system performance similar to the conventional system using an additional time delay estimator as a preprocessor. Simulation results show that the performance of the proposed system is identical to the conventional system that uses an additional time delay estimation module.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.4
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• pp.708-717
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• 2017

Tracking of sound-location is widely used in various area such as intelligent CCTV, video conference and voice commander. In this paper we introduce the real-time sound-location tracking method for smart lecture system using TDoA(Time Difference of Arrival) with orthogonal microphone array on the ceiling. Through discussion on some models of TDoA detection, cross correlation method using linear microphone array is proposed. Orthogonal array with 5 microphone could detect omni direction of sound-location. For real-time detection we adopt the threshold of received energy for eliminating no-voice interval, signed cross correlation for reducing computational complexity. The detected azimuth angles are processed using median filter for lowering the angle deviation. The proposed system is implemented with high performance MCU of TMS320F379D and MEMs microphone module and shows the accuracy of 0.5 and 6.5 in degree for white noise and lectured voice, respectively.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.05a
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• pp.484-488
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• 2007

In this paper, the measurement system using the microphone array developed for analyzing cabin noise of the vehicle and its applications are discussed. The sensor is a three dimensional microphone array, the microphones and cameras are equipped on the rigid sphere. The cameras are used for acoustic visualization. As applications, the experiments in both reverberation chamber and anechoic chamber are discussed. These results show that this system is very useful to evaluate or improve the vehicle sound insulation performance.

• 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.