• Economic and Environmental Geology
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• v.43 no.4
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• pp.349-358
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• 2010

For imaging complex subsurface structures such as salt dome, faults, thrust belt, and folds, seismic prestack reverse-time migration in depth domain is widely used, which is performed by the cross-correlation of shot-domain wavefield extrapolation with receiver-domain wavefield extrapolation. We apply the prestack reverse-time migration, which had been developed at KIGAM, to the seismic field data set of Block 5 in Jeju basin of Korea continental shelf in order to improve subsurface syncline stratigraphy image of the deep structures under the shot point 8km at the surface. We performed basic data processing for improving S/N ratio in the shot gathers, and constructed a velocity model from stack velocity which was calculated by the iterative velocity spectrum. The syncline structure of the stack image appears as disconnected interfaces due to the diffractions, but the result of the prestack migration shows that the syncline image is improved as seismic energy is concentrated on the geological interfaces.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.11a
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• pp.266-269
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• 2010

급속한 정보 통신 기술의 발달로 인해 멀티미디어 재생 개발 기술들은 단순히 수동적으로 보고 듣는 재생 기술에서 벗어나 청취자 감성, 취향 등에 따라 보다 실감 있고 사용자가 능동적으로 재생할 수 있는 기술로 진화 하고 있다. 지금까지의 오디오 서비스는 음원 개발자 중심의 오디오 서비스, 즉 보컬 및 모든 악기가 믹스된 단일음원이기 때문에 사용자는 단순히 오디오 음원 개발자나 음반 제작사가 발매한 단일 음원을 일방적으로 수동적 청취할 수밖에 없다. 하지만 사용자 능동형 오디오 서비스에서는 사용자가 능동적으로 자신이 원하는 음악적 취향에 따라 능동적으로 각각의 객체 기반의 독립 음원을 선택, 감성에 따른 음원 효과 추가, 최적의 음원 청취 위치(Sweet Spot) 변경, 음원 및 스피커 재생 공간 및 위치 변경 재생 등을 할 수가 있다. 본 논문에서는 디지털 음원들을 입력받아 임의의 필터링을 실행하고, 사용자 음원 보정 정보, 출력 유닛의 공간적, 음향적 특성을 상위제어기로부터 입력받아 전신호경로 상에 디지털 신호처리 하여 출력신호를 생성하는 DSP 시스템 플랫폼 및 알고리즘에 관하여 소개한다.

• Journal of Internet Computing and Services
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• v.18 no.5
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• pp.17-22
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• 2017

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. This paper present PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) used V/UV/S( Voiced / Unvoiced / Silence ) switching, position compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. Also, I was implemented that the PCFBD-MPC( Position Compensation Frequency Band Division-Multi Pulse Coding ) system and evaluate the SNRseg of PCFBD-MPC in coding condition of 8kbps. As a result, SNRseg of PCFBD-MPC was 13.4dB for female voice and 13.8dB for male voice respectively. In the future, I will study the evaluation of the sound quality of 8kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or a smart phone.

• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.112-120
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• 1992

수심 100m 이하 천해에서 저주파 대역 음파의 전달손실 양상을 규명하기 위해 한국 동해 남부해역 3개 정점에서 특정 주파수 모의신호 발생을 통한 전달손실을 측정하였다. 10개의 특정 주파수에 대해서 연속파(Continuous Wave)를 방생시킬 수 있는 저주파 음원기를 5 kts의 속도로 예인하고, 다시 육상으로 무선전송하여 각 센서에서의 수진준위를 정확하게 보정하였다. 음원으로부터 DIFAR 센서까지의 전달손실은 거리에 대한 Log 함수로 표시할 수 있었으며, 주파수별 전달손실을 비교, 분석한 결과 동해 남부해역에서의 최적 주파수는 800Hz 내외에서 존재하는 것으로 추정된다.

• Journal of Digital Convergence
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• v.11 no.5
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• pp.285-290
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• 2013

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of position compensation(PC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and PC-MPC using multi-pulses position compensation method. As a result, $SNR_{seg}$ of PC-MPC was improved 0.4dB for female voice and 0.5dB for male voice respectively. Compared to the MPC, $SNR_{seg}$ of PC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.9
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• pp.4119-4124
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• 2011

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of increasing or decreasing of speech signal amplitude in a frame. This is caused by normalization of synthesis speech signal in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of amplitude compensation(AC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and AC-MPC using amplitude compensation method. As a result, SNRseg of AC-MPC was improved 0.7dB for female voice and 0.7dB for male voice respectively. Compared to the MPC, SNRseg of AC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.40-50
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• 1994

In this paper, the sensitivity compensation method for three-dimensional acoustic intensity probe in the higher frequency range has been studied. The measurement error in the higher frequency range is generated from the phase mismatch between microphone's signals of the probe. If the wavelength of sound signal measured is less than those of the distance between microphones of the probe, that is, the higher frequency of the sound signal, the bigger measurement error is generated. In this study, we proposed the compensation methods for one-dimensional acoustic intensity probe with two-microphones, and the efficiency of those methods were investigated by numerical calculation of computer. It was most effective method to compensate the phase mismatch between microphone for the acoustic intensity probe was investigated for the sound estimated. and the efficiency of this method in a three-dimensional probe was investigated for the sound wave travelling in the arbitrary direction by numerical calculation of computer. In this result, the efficiency was proved that, for the measurement error of 1dB or less with the three-dimensional probe of 60mm space, the frequency should be less than 1.2kHz without the error compensation method, but the frequency increased up to 2.8kHz with the error compensation method.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.21 no.5
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• pp.69-76
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• 2020

The sound signal radiated from an underwater source is received by the hydrophone of the system, including multi-path time-delay and multi-path signal by sea surface and bottom reflection. The system using a time-delay between received signals for the source localization shows performance degradation due to incoherence by the multi-path propagation environment and the disturbance of a marine environment. Various types of array and signal processing have been used for robust source range and bearing estimation in this environment. In this paper, we use a line array composed of doublet array and an estimated time-delay correction method for robust localization performance in a multi-path propagation environment. Three doublet arrays are located on the same line, and the time-delay between signals received on each doublet array is estimated in a two-step procedure. The estimated time-delay value is obtained by the cross-correlation function and corrected by the interaction formula between the center-frequency of received signal and the geometry of the array with respect to aperture. By this proposed procedure, the range and bearing of source from array were calculated. In order to confirm the validity of the proposed method and array, we simulated localization and estimation using the Monte-Carlo method.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.541-544
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• 2004

본 논문에서는 서라운드 패닝알고리즘에서의 음색보정필터 설계기법에 대한 것으로 기존 패닝알고리즘에 음색보정필터를 추가하여 가상음원의 음색왜곡을 보정하는 알고리즘을 제안한 다. 가상음상과 실제 라우드 스피커의 머리전달함수 분석을 통해 기존 일정파워패닝알고리즘의 음색왜곡 문제점을 지적하고 이를 완화시키기 위한 새로운 패닝알고리즘을 제안한다.

• Geophysics and Geophysical Exploration
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• v.9 no.1
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• pp.102-107
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• 2006

Recently, rapid developments in computer hardware have enabled reverse-time migration to be applied to various production imaging problems. As a wave-equation technique using the two-way wave equation, reverse-time migration can handle not only multi-path arrivals but also steep dips and overturned reflections. However, reverse-time migration causes unwanted artefacts, which arise from the two-way characteristics of the hyperbolic wave equation. Zero-lag cross correlation with diving waves, head waves and back-scattered waves result in spurious artefacts. These strong artefacts have the common feature that the correlating forward and backward wavefields propagate in almost the opposite direction to each other at each correlation point. This is because the ray paths of the forward and backward wavefields are almost identical. In this paper, we present several tactics to avoid artefacts in shot-domain reverse-time migration. Simple muting of a shot gather before migration, or wavefront migration which performs correlation only within a time window following first arriving travel times, are useful in suppressing artefacts. Calculating the wave propagation direction from the Poynting vector gives rise to a new imaging condition, which can eliminate strong artefacts and can produce common image gathers in the reflection angle domain.