• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.449-453
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• 1998

일반적으로 음성신호는 파형의 특성에 따라 파형이 준주기적인 유성음과 주기성 없이 잡음과 유사한 무성음 그리고 배경 잡음에 해당하는 묵음의 세 종류로 분류된다. 기존의 유성음/무성음/묵음 분류 방법에서는 피치정보, 에너지 및 영교차율 등이 분류를 위한 파라미터로 널리 사용되었다. 본 논문에서는 음성신호를 웨이브렛 변환한 신호에서 스펙트럼상에서이 변화를 파라미터로 하는 유성음/무성음/묵음 분류 알고리즘을 제안하고 제안된 알고리즘으로 검출한 결과와 이에 따른 문제점을 검토하였다.

• The Journal of the Acoustical Society of Korea
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• v.41 no.6
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• pp.610-620
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• 2022

An acoustic vector sensor can simultaneously receive vector quantities, such as particle velocity and acceleration, as well as acoustic pressure at one location, and thus it can be used as a single input multiple output receiver in underwater acoustic communication systems. On the other hand, vector signals received by a single vector sensor have different channel characteristics due to the azimuth angle between the source and receiver and the difference in propagation angle of multipath in each component, producing different communication performances. In this paper, we propose a channel parameter-based weighting method to improve the performance of an acoustic communication system using a single vector sensor. To verify the proposed method, we used communication data collected from the experiment conducted during the KOREX-17 (Korea Reverberation Experiment). For communication demodulation, block-based time reversal technique which is robust against time-varying channels were utilized. Finally, the communication results showed that the effectiveness of the channel parameter-based weighting method for the underwater communication system using a single vector sensor was verified.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.211-215
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• 1995

명료하고 자연스러운 한국어 음성을 생성하기 위하여 다중대역 여기신호를 이용한 음성 합성 시스팀을 제안한다. 분석계에서는 켑스트럼 파라미터를 사용하여 유성/무성 판별 스펙트럼을 이용한 유/무성 구간 자동판별법을 제안하고, 현재 단순 임펄스와 백색잡음만으로도 구성된 음원과 간단한 유성/무성 판별로 구동되어지는 합성음의 음질상의 한계를 개선하기 위하여 합성계에서는 음질개선 방안으로 유성음 구동시 다중대역 여기신호를 도입하여 합성시 이용한다. 제안된 방법에 대한 청취실험을 한 결과, 유성음 부분 특히 잡음이 많이 섞여 있는 유성음화 마찰음과 모음의 천이부분 등에서 일반적으로 사용되고 있는 간단한 유성/무성 파라미터를 사용한 합성음에 비하여 다중대역 여기신호를 사용한 합성음의 명료도가 매우 우수함을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• v.10 no.5
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• pp.27-35
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• 1991

음소를 인식의 기본 단위로 하는 소규모 음성 인식 시스템을 구현하기 위한 기초 연구로서 마 찰음(/ㅅ, ㅆ, ㅎ/) 과 파찰음(/ㅈ, ㅉ, ㅊ/) 에 대하여 지속시간, 평균패턴, 분산비를 이용하여 각 음소 의 특징을 분석하고 각 음소군 내에서의 식별에 유효한 parameter들을 추출하여 인식 실험을 실시하 였다. 지속시간의 분포, 평균패턴의 분포, 분산비의 분포를 이용하여 분석한 결과 6차원 정도의 cepstrum 계수만으로 마찰음 및 파찰음의 식별이 가능하고, 시간 방향의 정보는 음성의 시단으로부터 14 frame 정도의 특징을 인식 파라미터로 할 경우가 최적임을 알 수 있었다. 이를 이용한 인식실험 결과에서는 조음방법별로 분류된 음소군내의 각 음소에 대한 인식실험의 인식률 보다는 발음방법별 인식실험시의 인식률이 높게 나타나 동일 음소군 내에서의 각 음소에 대한 식별이 더 어려움을 알 수 있었고, 특징 파라미터의 길이를 음성의 시단으로부터 14 frame 정도로 했을 때 조음방법별 인식률은 평균 81.1%, 발음방법별 인식률은 평균 97.9%로 최고의 인식률을 나타내었다. 특징 파라미터의 길이 를 14 frame 이상으로 증가시켜도 인식률은 큰 변화가 없어 분석 결과를 잘 설명하고 있음을 알 수 있었다.

• Journal of the Korean Society for Nondestructive Testing
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• v.28 no.2
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• pp.144-150
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• 2008

In this study, the PVDF (polyvinylidene fluoride) piero-film sensor was applied to measure the internal quality of apple. The developed sensor detected the response signal through apple after mechanical impact on the surface of apple. The acoustical parameters at time domain such as rise time (RT), ring down count (RC), energy (EN), event duration (ED) and peak amplitude (PA) and acoustical parameter at frequency domain such as spectral density (SE) were analyzed. The size of waveform decreased as storage time of apple increased. The frequency at maximum magnitude was shifted to lower frequency band according to the storage time. The acoustical parameters showed strong relationship with storage time. The multiple linear regression equation was developed to estimate storage time of apple using the acoustical parameters at time domain and its coefficient of determination was 0.97. The internal quality of apple according to storage time is predictable using developed PVDF sensor and acoustical parameters defined in this study.

• Journal of the Korean Society for Nondestructive Testing
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• v.29 no.5
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• pp.415-420
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• 2009

This paper reports the results of a case study for the evaluation of surface damage by using acoustic nonlinearity of surface wave. In this study, the experimental system was constructed to measure the acoustic nonlinear parameter of surface wave in an Aluminum 6061 T6 specimen of which surface was damaged by the three point bending fatigue test, and magnitudes of nonlinear parameter measured before and after the fatigue test were compared. Especially, since the surface fatigue damage by the three point bending is concentrated at the central position of loading, the change in the nonlinear parameter around this position was monitored. Experimental results showed that the measured nonlinear parameter at the outside of this position after the fatigue test was almost same as the initial value before the fatigue test, since the fatigue damage at this position was little. However, clear increase in the nonlinear parameter was noticed after the fatigue test at the central position of specimen where the surface fatigue damage is expected to be concentrated.

• The Journal of the Acoustical Society of Korea
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• v.40 no.2
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• pp.139-147
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• 2021

This paper introduces and implements a Sound Event Detection (SED) system based on weakly-supervised learning where only part of the data is labeled, and analyzes the effect of parameters. The SED system estimates the classes and onset/offset times of events in the acoustic signal. In order to train the model, all information on the event class and onset/offset times must be provided. Unfortunately, the onset/offset times are hard to be labeled exactly. Therefore, in the weakly-supervised task, the SED model is trained by "strongly labeled data" including the event class and activations, "weakly labeled data" including the event class, and "unlabeled data" without any label. Recently, the SED systems using the mean-teacher model are widely used for the task with several parameters. These parameters should be chosen carefully because they may affect the performance. In this paper, performance analysis was performed on parameters, such as the feature, moving average parameter, weight of the consistency cost function, ramp-up length, and maximum learning rate, using the data of DCASE 2020 Task 4. Effects and the optimal values of the parameters were discussed.

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.369-379
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• 2016

This paper deals with an efficient method for skipping inter-channel phase differences (IPD) in the MPEG surround of the unified speech and audio coding (USAC). Based on the psycho-acoustic sensitivity on the IPD, we estimate a threshold on IPD, below which we can not notice degradation in spatial cue. We propose an IPD skip method, in which any IPDs within the threshold are set to zero and are not transmitted. The proposed IPD skip method gives about 38% savings in terms of bit amount for IPD. Nevertheless, in the MUSHRA test, the proposed method does not show any noticeable degradation in the decoded audio quality.

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.435-440
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• 2012

We propose optimal contexts for context-adaptive coding of stereo parameters in parametric stereo (PS) of enhanced aacPlus. For the quantized indexes of stereo parameters, 8 context candidates were proposed based on the index values and their combinations adjacent to a source index in the time-stereo band domain, where the time-stereo band region was further divided into 4 regions based on refresh/non-refresh frames and stereo bands. The optimal contexts for each region were proposed by experiments, which are expected to be used for context-adaptive coding of PS for improved performance.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.559-565
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• 2002

In this paper, we proposed an efficient method that estimates the HMM (Hidden Marke Model) parameters of the noisy speech. In previous methods, noisy speech HMM parameters are usually obtained by analytical methods using the assumed noise statistics. However, as they assume some simplication in the methods, it is difficult to come closely to the real statistics for the noisy speech. Instead of using the simplication, we used some useful statistics from the clean speech HMMs and employed the deterministic noise model. We could find that the new scheme showed improved results with reduced computation cost.