• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.177-183
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• 1998

보통 말씨와 clear speech에서, 영어 전설모음[ i, , , e ]를 일정한 강세를 갖도록 하고, 모음의 길이를 변화시켜 음향적 분석을 하였다. 그 결과, 모음의 음가가 주변의 자음과 비슷한 값으로 변화하였으며, 그 변화 정도는 모음의 길이와 직접적인 관련이 있었으며, 이러한 변화의 크기는 clear speech에서 더 적었다. 이러한 결과는 clear speech 가 단순히 보통 말씨보다 소리가 큰 것일 뿐 아니라, 체계적으로 모음축약 현상을 보상하기 위하여 음향적인 특성을 재구성하는 것임을 시사한다.

• The Journal of the Acoustical Society of Korea
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• v.38 no.2
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• pp.240-245
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• 2019

This paper is a study result on long distance transmission of underwater acoustic sensor data over cable. The data transceiver is designed using the LVDS (Low Voltage Differential Signaling) transmission scheme, and the jitter characteristics are analyzed by measuring the long distance transmission signal through the cable. In order to reduce the jitter, a pre-emphasis technique is applied to compensate the transmitting signal to be attenuated by long distance transmission, and the transmission characteristics were verified according to the distance.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.12-21
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• 2000

In this paper, we propose a novel hearing aid algorithm for sensorinural hearing loss restoration with multi-channel(band) dynamic range compression and psychoacoustics. In this way, we can present a normal perception condition to the impaired listener. The proposed algorithm make loudness scaling function achieve proper loudness level, and analysis masking property for the signal will be perceived to impaired listener, and then, restore normal spectral contrast using SMR(signal-to-masking ratio) defined by distance between the level of each frequency and masking threshold.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.69-75
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• 2001

Intra-speaker variation, noise environments, and mismatch between training and test conditions are the major reasons for the speaker verification system unable to use it practically. In this study, we propose robust end-point detection algorithm, noise cancelling with the microphone property compensation technique, and inter-speaker discriminate technique by weighting cepstrum for robust speaker verification system. Simulation results show that the average speaker verification rate is improved in the rate of 17.65% with proposed end-point detection algorithm using LPC residue and is improved in the rate of 36.93% with proposed noise cancelling and microphone property compensation algorithm. The proposed weighting function for discriminating inter-speaker variations also improves the average speaker verification rate in the rate of 6.515%.

• The Journal of the Acoustical Society of Korea
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• v.31 no.3
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• pp.197-204
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• 2012

In this thesis, studied the method to minimize the changes in frequency response and level due to the variation of the distance from the source to the microphone. selecting three microphones (omni directional, cardioid, super cardioid) which are being used generally, frequency responses were measured in accordance with the distance changes. Gotten the difference from the reference as the result of measurement, changed responses for each frequency range were compensated in comparison of the original human vocal source. In low frequency range, the low frequency boost caused by the proximity effect and decrease in accordance with the distance were compensated. The variation in mid-frequency range is comparatively small, however since the mid-range is the most important part of the human vocal signal, were compensated the mid-frequency range in comparison of the reference. The human vocal signal variation in high frequency range is extremely small and the high frequency is compensated close to the original source without difficulty. Understanding the microphone characteristics and compensations, this study showed that the response can be maintain among the change of the distance from the source to the microphone.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.238-247
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• 2005

In this Paper. we Presents subband design techniques of a compensating filter and adaptive feedback canceller for the digital hearing aid. The sensorineural hearing loss has a hearing threshold that shows a nonlinear characteristic in frequency domain. and its compensation suffers from an echo that produced by an undesired time varying feedback path. Therefore. the digital hearing aid requires the compensator that can adjust gains nonlinearly in frequency bands and eliminate the echo rapidly In the Proposed digital hearing aid. the compensating filter is designed by the adaptive system identification method in subband structure, and the adaptive feedback canceller is designed by the subband affine projection algorithm. The designed compensation filter can control the nonlinear gain in each subband respectively, therefore precise compensation is possible. And the feedback canceller using the subband adaptive filter achieves fast convergence rate. The Performances of the Proposed method are verified by computer simulations as comparing with the behaviors of the previous trials.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1
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• pp.25-31
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• 1999

This paper proposes a new wide-band beamforming algorithm which has Gaussian weighting at nested linear array for acoustic measurement. The beamforming is performed one-shot by using all sensors simultaneously, not octave-by-octave. Gaussian weighting of frequency-dependent function is applied to each sensor before time-delay compensation so as to control the frequency receiving band of each sensor. As the results of the simulations, it is confirmed that the proposed algorithm can form the one-shot beams having uniform directivity index and also it can be applied to the broad-band acoustic measurement.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.257-260
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• 2004

HRTF 를 사용하여 가상적인 음향 효과를 내는 시스템에서의 음의 왜곡은 필수적이다. 하지만 음성과 같은 사람이 민감하게 느끼는 부분에서의 음의 선명도 저하는 다른 대역의 왜곡보다 더욱 크게 느껴진다. 그리고 모노 및 두 채널의 주파수 특성이 비슷한 스테레오에서는 음질을 보상할 수 없다는 단점이 있다. 본 논문에서는 이러한 음의 선명도의 저하를 유발하게 하는 가상 음향 발생 필터의 특징을 분석하고, 간단한 필터 설계에 의해서 본래의 가상 음향을 그대로 유지하면서 선명도를 높일 수 있는 방법을 제시한다. 제시한 방법은 특히 모노 및 뉴스모드와 같이 음성이 많이 들어 있는 부분에서 뛰어난 성능을 보인다.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.577-578
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• 2014

• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.159-165
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• 2022

In this paper, a frame synchronization structure for the Binary Phase Shift Keying (BPSK) modulation method in underwater acoustic communication was proposed. The proposed frame synchronization structure is largely divided into two. First, the approximate position and frequency offset of the frame are obtained by non-coherent correlation and sliding Fast Fourier Transform (FFT) method. Second, after compensating for the frequency error to the received signal, the exact position of the frame is obtained by coherent correlation method. Maritime experiments were conducted to confirm the performance of the 2-STEP frame synchronization structure. It was showed that the limitations of the non-coherent correlation and sliding FFT method can be verified when the power of the received signal was greatly reduced due to the channel characteristics. As a result, stable frame synchronization could be obtained by compensating for the frequency error and then using the coherent correlation method.