• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.8-13
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• 2013

Persons with sensorineural hearing impairment have troubles in hearing at noisy environments because of their deteriorated hearing levels and low-spectral resolution of the auditory system and therefore, they use hearing aids to compensate weakened hearing abilities. Various algorithms for hearing loss compensation and environmental noise reduction have been implemented in the hearing aid; however, the performance of these algorithms vary in accordance with external sound situations and therefore, it is important to tune the operation of the hearing aid appropriately in accordance with a wide variety of sound situations. In this study, a sound classification algorithm that can be applied to the hearing aid was suggested. The proposed algorithm can classify the different types of speech situations into four categories: 1) speech-only, 2) noise-only, 3) speech-in-noise, and 4) music-only. The proposed classification algorithm consists of two sub-parts: a feature extractor and a speech situation classifier. The former extracts seven characteristic features - short time energy and zero crossing rate in the time domain; spectral centroid, spectral flux and spectral roll-off in the frequency domain; mel frequency cepstral coefficients and power values of mel bands - from the recent input signals of two microphones, and the latter classifies the current speech situation. The experimental results showed that the proposed algorithm could classify the kinds of speech situations with an accuracy of over 94.4%. Based on these results, we believe that the proposed algorithm can be applied to the hearing aid to improve speech intelligibility in noisy environments.

• Journal of Korea Water Resources Association
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• v.51 no.4
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• pp.281-291
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• 2018

Harmony search algorithm (HSA), developed by Hydrosystem lab. in Korea University in 2001, was a new meta-heuristic optimization algorithm inspired by the iterative improvision process of Jazz music players where the best harmony is eventually produced. HSA is now one of the most well-known meta-heuristic algorithms (as proven by its cited number of the first published paper more than 3,600 times as of January 11th 2018 based on Google Scholar citation) and has been applied to diverse research domains such as not only water resources and civil engineering but also in medical science, business, and humanities. This paper is a review article written with the wish for wider application of HSA and other optimization algorithms, especially in the domain of water resources engineering. Therefore, this paper first briefly introduces the mechanism and operators of HSA and then reviews its application area and citation frequency per research domain. In addition, recent globalization of HSA will be investigated and summarized by checking the current status of related international conferences and on-going research projects. After reviewing previous domestic papers with optimization algorithms specifically published in the water resources domain, this paper is finalized by delivering some suggestions to encourage the application of optimization algorithms including HSA.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.25 no.6
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• pp.1353-1360
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• 2015

With the extreme development of Internet, recently most users refer the sites with the various Recommendation Systems (RSs) when they want to buy some stuff, movie and music. However, the possibilities of the Sybils with the malicious behaviors may exists in these RSs sites in which Sybils intentionally increase or decrease the rating values. The RSs cannot play an accurate role of the proper recommendations to the general normal users. In this paper, we divide the given rating values into the stable or unstable states and propose a system information based recommendation algorithm that minimizes the malicious user's influence. To evaluate the performance of the proposed scheme, we directly crawl the real trace data from the famous movie site and analyze the performance. After that, we showed proposed scheme performs well compared to existing algorithms.

• The KIPS Transactions:PartD
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• v.9D no.5
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• pp.925-930
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• 2002

This paper discusses techniques for embedding data into 3D polygonal models of geometry. Much researches of Watermarking had been gone as element technology of DRM (digital rights management). But, few research had gone to 3D polygonal model. Most research is limited at text document, 2D image, animation, music etc. RP system is suitable a few production in various goods species, and it is used much in industry to possible reason that produce prototype and find error or incongruent factor at early stage on design in product development childhood. This paper is research about method that insert watermark in STL ( stereolithography) file that have 3D shape model. Proposed algorithm inserts watermark in normal vector region and facet's interior region of 3D shape data. For this reason, 3D shape does not produce some flexure and fulfill invisibility of watermark. Experiment results that insert and extract watermark in normal netter region and facet's Interior region of 3D shape data by proposed algorithm do not influence entirely in 3D shape and show that insertion and extraction of watermark are possible.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.7
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• pp.308-316
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• 2002

In this paper, we propose a robust digital copyright-protection technique based on the concept of human auditory system. First, we propose a watermarking technique that accepts the various attacks such as, time scaling, pitch shift, add noise and a lot of lossy compression such as MP3, AAC WMA. Second, we implement audio PD(portable device) for copyright protection using proposed method. The proposed watermarking technique is developed using digital filtering technique. Being designed according to critical band of HAS(human auditory system), the digital filers embed watermark without nearly affecting audio quality. Before processing of digital filtering, wavelet transform decomposes the input audio signal into several signals that are composed of specific frequencies. Then, we embed watermark in the decomposed signal (0kHz~11kHz) by designed band-stop digital filer. Watermarking detection algorithm is implemented on audio PD(portable device). Proposed watermarking technology embeds 2bits information per 15 seconds. If PD detects watermark '11', which means illegal song. PD displays "Illegal Song" message on LCD, skips the song and plays the next song, The implemented detection algorithm in PD requires 19 MHz computational power, 7.9kBytes ROM and 10kBytes RAM. The suggested technique satisfies SDMI(secure digital music initiative) requirements of platform3 based on ARM9E core.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.23 no.6
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• pp.1069-1086
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• 2013

Nowadays, web defacement becomes the utmost threat which can harm the target organization's image and reputation. These defacement activities reflect the hacker's political motivation or his tendency. Therefore, the analysis of the hacker's activities can give the decisive clue to pursue criminals. A specific message or photo or music on the defaced web site and the outcome of analysis will be supplying some decisive clues to track down criminals. The encoding method or used fonts of the remained hacker's messages, and hacker's SNS ID such as Twitter or Facebook ID also can help for tracking hackers information. In this paper, we implemented the web defacement analysis system by applying CBR algorithm. The implemented system extracts the features from the web defacement cases on zone-h.org. This paper will be useful to understand the hacker's purpose and to plan countermeasures as a IDSS(Investigation Detection Support System).

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.4
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• pp.136-141
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• 2011

In this paper, we propose an approach to improve the data detection performance of the acoustic data transmission system based on the modulated complex lapped transform (MCLT). We first present an effective analysis of the features and the detection method of data in the acoustic data transmission system. And then feature vectors which are applied to the Gaussian mixture model (GMM) are selected from relevant parameters of the previous system for the efficient data detection. For the purpose of evaluating the performance of the proposed algorithm, Bit error rate (BER) of the received data was measured at different environments (music genres (rock, pop, classic, jazz) and different distances (1m∼5m) from the loudspeaker to the microphone in a office room) and yields better results compared with the conventional scheme of the acoustic data transmission system based on the MCLT.

• The Journal of the Acoustical Society of Korea
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• v.35 no.1
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• pp.49-54
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• 2016

In this paper, we propose a vocal separation method using weighted ${\beta}$-order minimum mean wquare error estimation (WbE) based on kernel back-fitting algorithm. In spoken speech enhancement, it is well-known that the WbE outperforms the existing Bayesian estimators such as the minimum mean square error (MMSE) of the short-time spectral amplitude (STSA) and the MMSE of the logarithm of the STSA (LSA), in terms of both objective and subjective measures. In the proposed method, WbE is applied to a basic iterative kernel back-fitting algorithm for improving the vocal separation performance from monaural music signal. The experimental results show that the proposed method achieves better separation performance than other existing methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.2C
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• pp.145-152
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• 2009

Ultra-wideband communication systems for impulse radio have merits that are possible for either high resolution ranging system or radio determination. Conventionally, in order to accomplish these functions, the rapid analog to digital convertor (ADC) is necessary to apply radio determination system operating in time domain. However, considering that low rate - wireless personal area network (LR-WPAN) aims to low-cost hardware implementation, the expensive ADC converting GHz sampling per second is not appropriate. So, this paper introduces the high resolution ranging system operating in frequency domain with using low sampling rate ADC, and a new non-coherent ranging scheme utilizing analog Frequency Modulation (FM) mode for the frequency domain transformation. To verify the superiority of the proposed ranging algorithm working in frequency domain, the suggested IEEE 802.15.4a TG channel model is used to exploit affirmative features of the proposed algorithm with conducting the simulation results.

• International journal of advanced smart convergence
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• v.5 no.4
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• pp.15-20
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• 2016

When a virtual sound source for 3D auditory system is reproduced by a linear loudspeaker array, listeners can perceive not only the direction of the source, but also its distance. Control over perceived distance has often been implemented via the adjustment of various acoustic parameters, such as loudness, spectrum change, and the direct-to-reverberant energy ratio; however, there is a neglected yet powerful cue to the distance of a nearby virtual sound source that can be manipulated for sources that are positioned away from the listener's median plane. This paper address the problem of generating binaural signals for moving sources in closed or in open environments. The proposed perceptual enhancement algorithm composed of three main parts is developed: propagation, reverberation and the effect of the head, torso and pinna. For propagation the effect of attenuation due to distance and molecular air-absorption is considered. Related to the interaction of sounds with the environment, especially in closed environments is reverberation. The effects of the head, torso and pinna on signals that arrive at the listener are also objectives of the consideration. The set of HRTF that have been used to simulate the virtual sound source environment for 3D auditory system. Special attention has been given to the modelling and interpolation of HRTFs for the generation of new transfer functions and definition of trajectories, definition of closed environment, etc. also be considered for their inclusion in the program to achieve realistic binaural renderings. The evaluation is implemented in MATLAB.