• The Journal of the Acoustical Society of Korea
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• v.37 no.4
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• pp.196-205
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• 2018

This paper presents an algorithm for passive localization of a ship by applying the ray back-propagation technique to the ship radiation noise data. The previous method [S. H. Abadi, D. Rouseff and D. R. Dowling, J. Acoust. Soc. Am. 131, 2599-2610 (2012)] estimates the position of a sound source in the near-field environment with no array tilt by using the RBD (Ray-based Blind Deconvolution) and ray back-propagation techniques. However, when there exists an array tilt, the above method leads to a large position estimation error. In order to overcome the problem, this study proposes an algorithm that estimates the position of a sound source by correcting the array tilt using the RBD and ray back-propagation techniques. The proposed algorithm was verified by using the ship noise of SAVEX15 (Shallow-water Acoustic Variability EXperiment in 2015) experimental data.

• 한국신재생에너지학회:학술대회논문집
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• 2008.10a
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• pp.203-207
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• 2008

Prestack depth migration is used to image for complex geological structure such as faults, folds, and subsalt. In this case, it is widely used the surface reflection data as a input data. However, the surface reflection data have intrinsic problems to image the subsalt and the salt flank due to the complex wavefields and multiples which come from overburden. For overcoming the structural defect of the surface reflection data in the imaging, I used the virtual sources in terms of seismic interferometry to image the subsurface and suppress the multiples using the velocity model of the lower part of the virtual sources. The results of the prestack depth migration using virtual source gathers and velocity model below receivers are similar geological interfaces to the results from shot gathers of the conventional ocean bottom seismic survey. And especially artificial interfaces by multiples were suppressed without applying any other data processing to eliminate multiples. This study results by numerical modeling can make a valuable imaging tool when it is applied to satisfied field data for specific condition.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.1
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• pp.42-49
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• 2009

In this paper, an alternative method was introduced to conduct a transfer path analysis for airborne noise. The method used the transfer function matrix composed of acoustic transfer functions that are referenced by the input voltage of a calibration source. A calibration factor which is converting a virtual voltage to source strength was deduced by vibro-acoustical reciprocity theorem. The calibration factor is then multiplied to the virtual input voltage to estimate the operational source strength. Three loudspeakers were used to noise sources of acrylic half car model. The method was applied to airborne noise transfer path analysis of the half car. The estimated source strength by transfer path analysis was compared the deduced source strength by vibro-acoustical reciprocity to verify the method.

• Economic and Environmental Geology
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• v.39 no.6 s.181
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• pp.711-717
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• 2006

In order to study gas hydrate, potential future energy resources, Korea Institute of Geoscience and Mineral Resources has conducted seismic reflection survey in the East Sea since 1997. one of evidence for presence of gas hydrate in seismic reflection data is a bottom simulating reflector (BSR). The BSR occurs at the interface between overlaying higher velocity, hydrate-bearing sediment and underlying lower velocity, free gas-bearing sediment. That is often characterized by large reflection coefficient and reflection polarity reverse to that of seafloor reflection. In order to apply depth migration to seismic reflection data. we need high performance computers and a parallelizing technique because of huge data volume and computation. Phase shift plus interpolation (PSPI) is a useful method for migration due to less computing time and computational efficiency. PSPI is intrinsically parallelizing characteristic in the frequency domain. We conducted conventional data processing for the gas hydrate data of the Ease Sea and then applied prestack depth migration using message-passing-interface PSPI (MPI_PSPI) that was parallelized by MPI local-area-multi-computer (MPI_LAM). Velocity model was made using the stack velocities after we had picked horizons on the stack image with in-house processing tool, Geobit. We could find the BSRs on the migrated stack section were about at SP 3555-4162 and two way travel time around 2,950 ms in time domain. In depth domain such BSRs appear at 6-17 km distance and 2.1 km depth from the seafloor. Since energy concentrated subsurface was well imaged we have to choose acquisition parameters suited for transmitting seismic energy to target area.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.17 no.7
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• pp.49-53
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• 2016

Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.509-517
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• 2013

In this paper, we researched a method that makes a good acoustic-speech system using a digital signal processing technique with dynamic microphone as a transducer. Good acoustic-speech system should deliver the original sound input to electric signal without distortion. By measuring the frequency response of the microphone, adjustment factors are obtained by comparing measured data and standard frequency response of microphone for each frequency band. The final sound levels are obtained using the developed adjustment factors of frequency responses from the microphone and speaker to match the original sound levels using the digital signal processing technique. Then, we minimize the changes in the frequency response and level due to the variation of the distance from source to microphone, where the frequency responses were measured according to the distance changes.

• The Journal of the Acoustical Society of Korea
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• v.39 no.5
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• pp.490-496
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• 2020

We present a method to improve the performance of the multi-channel Wiener filter in noisy environment. To build subspace-based multi-channel Wiener filter, in the case of single target source, the target speech component can be effectively estimated in the principal subspace of speech correlation matrix. The speech correlation matrix can be estimated by subtracting noise correlation matrix from signal correlation matrix based on the assumption that the cross-correlation between speech and interfering noise is negligible compared with speech correlation. However, this assumption is not valid in the presence of strong interfering noise and significant error can be induced in the principal subspace accordingly. In this paper, we propose to adjust the principal subspace vector using speech presence probability and the steering vector for the desired speech source. The multi-channel speech presence probability is derived in the principal subspace and applied to adjust the principal subspace vector. Simulation results show that the proposed method improves the performance of multi-channel Wiener filter in noisy environment.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.397-400
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• 2004

소음원 또는 방사음장을 가시화하기 위하여 소음원을 둘러싸는 한 면에서의 음압측정을 통해 임의의 3차원 공간상의 음향 물리량을 예측하는 음향 홀로그래피 방법이 사용되고 있으며 이때 반사파가 존재하지 않는 기본 가정을 만족해야 한다. 반사파가 존재하는 경우에는 반사파를 보정하거나 또는 반사파의 음장이 미치는 영향이 무시할 만큼 작다고 가정하여 음장 예측이 가능하게 된다. 최근 해양연구원에서는 음향 홀로그래피 방법을 이용하여 수중음장을 가시화하는 시스템을 개발하였으며 시스템 검증을 위해 무향수조 내 단순음원을 이용하여 음장예측을 수행하였다. 무향수조 표면에는 흡음처리가 되지 않아 표면 반사파가 존재하나 해석 결과 반사파의 영향이 작은 경우에는 반사파를 무시하고 음장해석의 수행이 가능함을 확인하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.289-292
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• 1999

다수의 마이크로폰 어레이를 사용하여 소음원에서 방사하는 음장을 예측, 가시화하고 소음원의 시, 공간적 특성을 파악하기 위하여 음향 홀로그래피 방법에 대한 연구를 수행하였다. 음향 홀로그래피 방법은 실험적으로 소음원의 특성을 규명할 수 있기 때문에 많은 연구가 활발히 진행되고 있지만, 많은 개수의 마이크로폰과 신호수집장치 등이 필요하기 때문에 그 사용에 많은 제약이 있어 왔다. 음향 홀로그래피 방법중에서 대표적인 평면 음향 홀로그래피 방법을 중심으로 마이크로폰의 개수, 간격등과 같은 측정조건과 함께 마이크로폰을 스캐닝하는 방법둥에 대한 해석을 통하여 장, 단점 및 제한성을 논하였다. 또한 이러한 측정방법에서 나타나는 오차요인을 해석하고 이를 보정하는 방법에 대한 설명과 함께 실험을 통하여 이를 확인해 보았다.

• Proceedings of the Korea Information Processing Society Conference
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• 2014.11a
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• pp.975-977
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• 2014

본 논문에서는 판소리 자동채보에 중요한 요소인 '합'과 '궁'의 위치 즉 마디를 인공신경망과 히스토그램을 이용하여 추정한다. 기존의 합과 궁을 추정하는 방법으로는 NCC(Normalized Cross Correlation)를 이용한 대표치 추정 윈도우와 칼만 필터를 이용하였다. 하지만 대표치 추정 윈도우를 구성하는 과정에서 단순히 15개의 특징벡터 각각의 평균을 이용하기 때문에 분별력이 떨어지고, 마디위치를 보정하는 과정에서 칼만 필터를 사용하면 전체음원이 길이가 짧을 경우 오차가 발생할 가능성이 크다. 본 논문에서 제안한 마디 추정 알고리즘은 장단별로 최대 90%이상의 정확도로 마디를 추정할 수 있다.