• Journal of Internet Computing and Services
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• v.9 no.2
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• pp.131-138
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• 2008

The goal of our research is to develop a smart location-based self guided lour assistant as a context-aware real world application. As a context-aware tourist guide application, we hove been developing a PDA-based location-aware tourist guide application for the old palace Deoksugung in the center of Seoul. It will guide visitors to the palace with information about: current location, attractions nearby, and details about specific buildings. Rich multimedia support has been incorporated into the system to provide extra features to enhance the self-guided tour.

• The Journal of the Acoustical Society of Korea
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• v.39 no.5
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• pp.447-453
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• 2020

In this paper, we propose a system to extract effective speaker representations from a speech signal using a deep learning method. Based on the fact that speech signal contains identity unrelated information such as text content, emotion, background noise, and so on, we perform a training such that the extracted features only represent speaker-related information but do not represent speaker-unrelated information. Specifically, we propose an auto-encoder based disentanglement method that outputs both speaker-related and speaker-unrelated embeddings using effective loss functions. To further improve the reconstruction performance in the decoding process, we also introduce a discriminator popularly used in Generative Adversarial Network (GAN) structure. Since improving the decoding capability is helpful for preserving speaker information and disentanglement, it results in the improvement of speaker verification performance. Experimental results demonstrate the effectiveness of our proposed method by improving Equal Error Rate (EER) on benchmark dataset, Voxceleb1.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.8 no.4
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• pp.952-960
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• 2007

Most Websites perform login process for authentication. But simple features like ID and Password have no trust because most people worry about appropriation. So the youth can easily access illegal media sites using other's ID and Password. Therefore this paper examine features be adaptable to authentication system, and propose a design of authentication system using multiple features. A proposed authentication system has two categories, such as low-level and high-level method. Low-level method consists of grant of authentication number through mobile phone from server and certificate from authority. High-level method combines ID/Password and features of fingerprint, character, voice, face recognition systems. For this, this paper surveys six recognition systems such as fingerprint, face, iris, character, vein, voice recognition system. Among these, fingerprint, character, voice, face recognition systems can be easily implemented in personal computer with low cost accessories. Usage of multiple features can improve reliability of authentication.

• Journal of the Korean Chemical Society
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• v.54 no.1
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• pp.93-98
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• 2010

In this work, the electrochemical properties of surface treated multi-walled carbon nanotubes (MWNTs) were studied. Nitrogen and oxygen functional groups of the MWNTs were introduced by urea and acidic treatment, respectively. The surface functional groups of the MWNTs were confirmed by X-ray photoelectron spectroscopy (XPS) measurements and zeta-potential method. The characteristics of $N_2$ adsorption isotherm at 77 K, specific surface area, and total pore volumes were investigated by BET eqaution, BJH method and t-plot method. Electrochemical properties of the functionalized MWNTs were accumulated by cyclic voltammetry at the scan rates of 50 $mVs^{-1}$ and 100 $mVs^{-1}$ in 1M $H_2SO_4$ as electrolytes. As a result, the functionalized MWNTs led to an increase of capacitance as compared with pristine MWNTs. It was found that the increase of capacitance for urea treated MWNTs was attributed to the increase in density of surface functional groups, resulting in improving the wettability between electrode materials and charge species.

• The Journal of the Acoustical Society of Korea
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• v.20 no.8
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• pp.3-11
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• 2001

This paper describes an implementation of inverse filter using SVD in order to recover the input in multi-channel system. The matrix formulation in SISO system is extended to MIMO system. In time and frequency domain we investigates the inversion of minimum phase system and non-minimum phase system. To execute an effective inversion of non-minimum phase system, SVD is introduced. First of all we computes singular values of system matrix and then investigates the phase property of system. In case of overall system is non-minimum phase, system matrix has one (or more) very small singular value (s). The very small singular value (s) carries information about phase properties of system. Using this property, approximate inverse filter of overall system is founded. The numerical simulation shows potentials in use of the inverse filter.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.10
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• pp.2263-2268
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• 2015

This paper describes packet scheduling techniques that can be used to alleviate the impact of interference. The mechanism is consisted of interference estimation and master delay police. Proposed scheduling police is effective in reducing packet loss and delay. Another advantage worth mentioning, are the additional saving s in the transmitter power since packet are not transmitted when channel is bad. This paper gives that scheduling policy works only with data traffic since voice packets need to be sent at fixed intervals. However, if the delay variance is constant and the delay can be limited to a slot, it may be worthwhile to use DM packet for voice.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1219-1222
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• 1998

This paper analyzes the dynamic spars algebraic codebook used to model a residual signal and proposes a new algebraic codebook structure as well as a searching process with improved performance. The proposed algorithm improves the disadvantage of algebraic codebook without increased computation. First, this paper makes it possibel to select various pulse amplitudes differently from the conventional method which looks up the sign bit simply. In addition, two pulses are made to be selected on the same track. For speech quality on the telephone line 5.6kbps speech coder using the proposed algorithm was equivalent to the 6.3kbps MP-MLQ in the viewpoint of subjective speech quality. However, speech degradation was caused a little compared to the MP-MLQ where MNRU 1=15dB.

• Information and Communications Magazine
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• v.17 no.3
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• pp.123-136
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• 2000

데이터(Data), 영상(Video) 그리고 음성(Audio)을 특정 사용자 그룹에게만 전송하는 데이터 전송기술인 멀티캐스트(Multicast)는 효과적인 그룹 접근 제어의 결려와 단일 유니캐스트 통신보다 많은 통신 링크 때문에, 부당한 공격자에게 많은 공격 기회를 제공하고 있다. 이것은 그룹의 수신자에게만 영향을 미치는 것이 아니라 잠재적으로 대부분의 네트워크에 연결된 사용자에게 영향을 미친다. 특히 참가자의 가입과 탈퇴 시에 신분위장, 재전송, 부인공격에 노출되어있기 때문에 이러한 부단한 공격 위협에 대한 보안대책이 필요하다. 본 논문에서는 안전한 멀티캐스트 트래픽 전송을 위한 보안 메커니즘 설계를 위한 기반 기술로서 멀티캐스트에서 보안의 필요성과 멀티캐스트에서 고려해야할 보안 서비스를 분석하고 단일 송신자와 다중 그룹에서 구성될 수 있는 보안 구조 및 보안 구성요소를 연구하였다.

• Proceedings of the Korean Society of Marine Engineers Conference
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• 2005.11a
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• pp.57-58
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• 2005

Recently the telematics system is used widely. Users want to high quality communications. Since the primary advantage of using an array is to enhance a desired signal and reject jamming interferences, array signal processing is essential to satisfy unmet demand of user. In general, beamforming is a spatial filtering operation performed on the data received by an array of sensors. So we propose the beamformer design that use FPGA for real time processing. And we use zero-padding interpolation for high resolution data.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.4
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• pp.57-65
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• 1997

The purpose of this paper is to propose the HMM (hidden markov model) based on multi-observation sequence for the isolated word recognition. The proosed model generates the codebook of MSVQ by dividing each word into several sections followed by dividing training data into several sections. Then, we are to obtain the sequential value of multi-observation per each section by weighting the vectors of distance form lower values to higher ones. Thereafter, this the sequential with high probability value while in recognition. 146 DDD area names are selected as the vocabularies for the target recognition, and 10LPC cepstrum coefficients are used as the feature parameters. Besides the speech recognition experiments by way of the proposed model, for the comparison with it, the experiments by DP, MSVQ, and genral HMM are made with the same data under the same condition. The experiment results have shown that HMM based on multi-observation sequence proposed in this paper is proved superior to any other methods such as the ones using DP, MSVQ and general HMM models in recognition rate and time.