• Journal of Soil and Groundwater Environment
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• v.26 no.6
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• pp.176-185
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• 2021

This paper aims to develop a new paradigm for groundwater management which is compatible with integrated water management policies in Korea. Three key roles of groundwater are defined for addressing water cycle distortion, high water stress, water quality degradation, aquatic ecosystems deterioration, and water-related hazards. Firstly, groundwater plays an important role in contributing soundness of water cycle as a component of water cycle. Secondly, it is a local water resource to ensure water supply sustainability. Thirdly, groundwater is an essential water resource for drought and emergencies. In order to support the groundwater roles, we propose a paradigm shift for groundwater management and policy directions towards integrated water management. The new paradigm consists of managements for sound water cycle on a watershed scale and groundwater environment(quantity, quality, and groundwater dependent ecosystems) managements for both human and nature. A prospective management also constitutes the new paradigm. In addition, this paper proposes four policy directions in groundwater management. The policies emphasize the integrated management of groundwater and surface water, management of groundwater environment(quantity, quality, and groundwater dependent ecosystems), management of groundwater uses for water sustainability and security, and enhancement of groundwater publicity.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.4 no.2
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• pp.77-82
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• 2011

In this paper, we investigate source separation algorithms for stereo audio mixed using amplitude panning method. This source separation algorithms can be used in various applications such as up-mixing, speech enhancement, and high quality sound source separation. The methods in this paper estimate the panning angles of individual signals using the principal component analysis being applied in time-frequency tiles of the input signal and independently extract each signal through directional filtering. Performances of the methods were evaluated through computer simulations.

• Journal of Korean Society of Archives and Records Management
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• v.11 no.1
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• pp.183-206
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• 2011

The research and laboratory records management is the prerequisite for quality research. Quality research assurance system can work out on the sound basis of quality research records management. It is an important task that establishing a proper research and laboratory records system for obtaining quality of research according to the rapid growth of Research and Development area in Korea and the trend of electronic laboratory notebooks. The purpose of this study is to identify issues and find a direction for solutions related to the research and laboratory records management systems. For this after analysing previous studies and current status related to the research records management and GLP(Good Laboratory Practice) is benchmarked as a best practice for quality research, some suggestions for enhancement of research records management are given as a result.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.20 no.6
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• pp.55-62
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• 2019

We examined the Audit Quality on the Socially Responsible Investment(SRI) Corporate. We used 1,497 sample data from 2014 to 2016. In short, the result of this paper's is as followed. Socially Responsible Investment(SRI) has a positive relevance with Audit Quality. Socially Responsible Investment(SRI) has a positive relevance with Audit Fee, Audit Time and Audit Size specifically. Therefore we can support that a firm has a high level of Socially Responsible Investment(SRI) will have the better the Audit Quality according to this study. This study contributes as follow. We can verify that the more Socially Responsible Investment(SRI) the better Quality of Accounting Information. We expect that this study can be helped positive image enhancement of Socially Responsible Investment(SRI) Corporate. So we hope that our paper can contribute sound capital market's development.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.45 no.5
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• pp.13-20
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• 2008

Although digital hearing aid algorithms have been developed to compensate hearing loss and to help hearing impaired people to communicate with others, digital hearing aid user still complain about difficulty of hearing the speech. The reason could be the quality of speech through digital hearing aid is insufficient to understand the speech caused by feedback, residual noise and etc. And another thing is masking effect among formants that makes sound quality low. In this study, we measured the masking characteristics of normal listeners and hearing impaired listeners having presbyacusis to confirm masking effect in speech itself. The experiment is composed of 5 tests; pure tone test, speech reception threshold (SRT) test, word recognition score (WRS) test, puretone masking test and speech masking test. In speech masking test, there are 25 speeches in each speech set. And log likelihood ratio (LLR) is introduced to evaluate the distortion of each speech objectively. As a result, the speech perception became lower by increasing the quantity of formant enhancement. And each enhanced speech in a speech set has statistically similar LLR, however speech perception is not. It means that acoustic masking effect rather than distortion influences speech perception. In actuality, according to the result of frequency analysis of the speech that people can not answer correctly, level difference between first formant and second formant is about 35dB, and it is similar to result of pure tone masking test(normal hearing subject:36.36dB, hearing impaired subject:32.86dB). Characteristics of masking effect is not similar between normal listeners and hearing impaired listeners. So it is required to check the characteristics of masking effect before wearing a hearing aid and to apply this characteristics to fitting.

• Journal of the Korean Society of Radiology
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• v.17 no.1
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• pp.157-165
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• 2023

Ultrasound imaging uses sound waves of frequencies to cause physical actions such as reflection, absorption, refraction, and transmission at the edge between different tissues. Improvement is needed because there is a lot of noise due to the characteristics of the data generated from the ultrasound equipment, and it is difficult to grasp the shape of the tissue to be actually observed because the edge is vague. The edge enhancement method is used as a method to solve the case where the edge surface looks clumped due to a decrease in image quality. In this paper, as a method to strengthen the interface, the quality improvement was confirmed by strengthening the interface, which is the high-frequency part, in each image using an unsharpening mask and high boost. The mask filtering used for each image was evaluated by measuring PSNR and SNR. Abdominal, head, heart, liver, kidney, breast, and fetal images were obtained from Philips epiq5g and affiniti70g and Alpinion E-cube 15 ultrasound equipment. The program used to implement the algorithm was implemented with MATLAB R2022a of MathWorks. The unsharpening and high-boost mask array size was set to 3*3, and the laplacian filter, a spatial filter used to create outline-enhanced images, was applied equally to both masks. ImageJ program was used for quantitative evaluation of image quality. As a result of applying the mask filter to various ultrasound images, the subjective image quality showed that the overall contour lines of the image were clearly visible when unsharpening and high-boost mask were applied to the original image. When comparing the quantitative image quality, the image quality of the image to which the unsharpening mask and the high boost mask were applied was evaluated higher than that of the original image. In the portal vein, head, gallbladder, and kidney images, the SNR, PSNR, RMSE and MAE of the image to which the high-boost mask was applied were measured to be high. Conversely, for images of the heart, breast, and fetus, SNR, PSNR, RMSE and MAE values were measured as images with the unsharpening mask applied. It is thought that using the optimal mask according to the image will help to improve the image quality, and the contour information was provided to improve the image quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1573-1580
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• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• Management & Information Systems Review
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• v.28 no.3
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• pp.191-210
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• 2009

Internet was first designed only for sending data, but as the time passed, internet started to evolve into a broadband multi-media web that is capable of transmitting sound, video, high-capacity data and more due to the demands of internet users and the rapid changing internet-communication technology. Domestically, in January, 2000 Saerom C&T, launched a free VoIP, but due to limited ways of conversation(PC to PC) and absence of a revenue model, and bad speech quality, it had hit it's growth limit. This research studied VoIP based on technological enhancement in super-speed internet. According to IDC, domestic internet market's size was 80,800 million in 2008, and it formed a percentage of 12.5% out of the whole sound-communication market. in case of VoIP, it is able to maximize it's profit by connecting cable and wireless network, also it has a chance of becoming firm-concentrated monopoly market by fusing with IPTV. Considering the fact that our country is insignificant in MVNO revitalization, regulating organizations will play a significant roll on regulating profit between large and small businesses. Further research should be done to give VoIP a secure footing to prosper and become popularized.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.