• Proceedings of the KSPS conference
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• 2002.11a
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• pp.187-190
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• 2002

In this paper, a speech enhancement system using microphone array with MMSE-STSA (Minimum Mean Square Error-Short Time Spectral Amplitude) estimator based post-processing is proposed. Speech enhancement is first carried out by conventional delay-and-sum beamforming (DSB). A new MMSE-STSA estimator is then obtained by refining MMSE-STSA estimators from each microphone, which is applied to the output of conventional DSB to obtain additional speech enhancement. Computer simulation for white and pink noises show that the proposed system is superior to other approaches.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.1667-1674
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• 2000

A new method is proposed to measure the flow rate in a pipe by multiple measurements of acoustic pressure using a microphone array. It is based on the realization that variation in flow velocity affects the change in wave number. The method minimizes measurement random errors and sensor mismatch errors thereby providing practically realizable flow rate measurement. One of the advantages of the method is that it does not obstruct the flow field and can provide the time-spatial mean flow rate. Numerical simulations and experiments were conducted to verify the utility of this method.

• 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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2013.10a
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• pp.440-441
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• 2013

• Journal of the Korean Society for Precision Engineering
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• v.26 no.11
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• pp.20-28
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• 2009

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.723-726
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• 2009

In this research, We simulated performances of error reduction algorithm for the speech signal based on the microphone array-based beamforming method in speech recognition system and analyzed its performance. Also, we processed speech signal adopted from microphone array and maximum signal to noise ratio for each channel, and then compared them with signal to noise ratio of speech signal. Speech recognition rate is improved from 54.2% to 61.4% in case 1 and is improved from 41.2% to 50.5% in case 2 of the lower signal to noise ratio. Therefore the average reduction rates are showed 15.7% in case 1.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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• pp.38-46
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• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.5
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• pp.808-811
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• 2022

In this paper, a wide-range microphone system for lectures using dual 3D sensors is proposed. A previous work using a single sensor had lowering the detecting threshold to support wide-area. However it was found that an error occurred when lecturer wears clothes with low reflectivity or has small body size. When multiple sensors are used to expand the coverage it could be cause various problems. Each sensor could show different distance to the same target. We derive the rotation angle and and compensate for lecturing microphone system using sensors on the line. The proposed method shows a little improvement in performance by about 1dB compared to the previous works but the performance is uniform in all areas regardless of reflectivity.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1996.04a
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• pp.291-297
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• 1996

To visualize sound field or to identify noise sources, we can use many methods such as intensity method, acoustic holographic method, source identification method using line array, etc. Conventionally all these methods are performed with the assumption of stationary condition in space and time. But for moving source, spatial characteristics and frequency components are changing, so we need another processing algorithm. This paper shows some experimental results - sound field by moving noise sources. In the experiment cross type microphone line array is used for sensing pressure and cars and a motorcycle are used as moving sources that are assumed to have constant speed. The processing methods are acoustic holographic method, spherical beamforming and spectrogram.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.14 no.12
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• pp.1241-1248
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• 2004

This paper presents an acoustic source localization technique on 2D cavity model in flow using a phased microphone array. Investigation was performed on cavity flows of open and closed types. The source distributions on 2D cavity flow were investigated in an anechoic open-jet wind tunnel. The array of microphones was placed outside the flow to measure the far field acoustic signals. The optimum sensor placement was decided by varying the relative location of the microphones to improve the spatial resolution. Pressure transducers were flush-mounted on the cavity surface to measure the near-filed pressures. It is shown that the propagated far field acoustic pressures are closely correlated to the near-field pressures and their spectral contents are affected by the cavity parameter L/D.