• Knowledge Management Research
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• v.18 no.1
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• pp.83-116
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• 2017

The purpose of this study is to analyze empirically analyze how the characteristics of participants in educational and social learning forums and the quality of events influence expectations and satisfaction of forums. The study also aims to provide strategic implications for forum organizers and give them suggestions on how to set up target audience, manage forum contents, speakers, and services, improve attendee satisfaction, and ultimately maximize overall outcomes. As exchanges among individuals, enterprises, and organizations, as well as countries are growing rapidly, the convention industry has become a key player in the market. Conventions have also become a venue for people to discuss a specific agenda or topic, exchange information and learn knowledge and insights. Especially, the forum - as part of the convention industry - plays a vital role providing educational and social learning opportunities as scholars and expertise come together to share their knowledge and experience through a variety of discussions. With its role, many of forums are taking place in recent years; however, there have been few empirical studies upon the forum itself. Also, there have been few attempts to research how the quality of forums affect participants' satisfaction along with their characteristics and how much of practical knowledge is provided throughout the events. This study is meaningful in that it is the first practical study that takes a deep understanding of the forum and sees how the quality of the forums influences participants' satisfaction and whether the characteristics of participants have a moderating effect in increasing the level of satisfaction. Forum organizers could also take a strategic approach as their major concerns are to increase the number of participants and raise degree of satisfaction by providing significant information. There are four key elements that determine success or failure of a social learning forum. The four elements are contents, speakers, services, and participants. Content plays an important role in providing rich information and knowledge for participants. Speakers are the main knowledge providers who contribute to the forum's social learning role. Also, the services provided by forum organizers such as simultaneous interpretation services, program brochures, lunch and refreshments, and the overall design of event hall can also influence the level of participants' satisfaction. Lastly, the participants and their characteristics are important since they are the ones who receive knowledge from the providers. The results of this study show that the quality of forum (content, speaker, and services) has a decisive effect on the participants' satisfaction and there are some differences in expectation among the participants in the forum. Also, some groups of participants were more likely to be stimulated by the quality of forum when determining their satisfaction. The study is modeled after MBN Y Forum 2016 and its participants' characteristics. The forum is one of the most representative social learning forums of South Korea and its audiences are mostly young people. It has analyzed how the participants' characteristics influence their satisfaction by grouping them into ${\Delta}participants$ who have invited for free and those who paid for the entrance fee, ${\Delta}first-time$ participants and returning participants, ${\Delta}voluntary$ and involuntary participants, ${\Delta}participants$ who registered through web and those who did through mobile, and ${\Delta}participants$ who registered during pre-sale opens and those who registered during general opens.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.34-39
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• 2001

An adaptive multi-rate (AMR) speech coder was adopted as a standard of W-CDMA by 3GPP and ETSI. The AMR coder is based on the CELP algorithm operating at rates ranging from 12.2 kbps down to 4.75 kbps, and it is a source controlled codec according to the channel error conditions and the traffic loading. In this paper, we implement the DSP S/W of the AMR coder using OakDSPCore. The implementation is based on the CSD17C00A chip developed by C&S Technology, and it is tested using test vectors, for the AMR speech codec, provided by ETSI for the bit exact implementation. The DSP B/W requires 20.6 MIPS for the encoder and 2.7 MIPS for the decoder. Memories required by the Am coder were 21.97 kwords, 6.64 kwords and 15.1 kwords for code, data sections and data ROM, respectively. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.9
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• pp.1031-1041
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• 1999

기존의 한글에 대한 입술 애니메이션 방법은 음소의 입모양을 몇 개의 입모양으로 정의하고 이들을 보간하여 입술을 애니메이션하였다. 하지만 발음하는 동안의 실제 입술 움직임은 선형함수나 단순한 비선형함수가 아니기 때문에 보간방법에 의해 중간 움직임을 생성하는 방법으로는 음소의 입술 움직임을 효과적으로 생성할 수 없다. 또 이 방법은 동시조음도 고려하지 않아 음소들간에 변화하는 입술 움직임도 표현할 수 없었다. 본 논문에서는 동시조음을 고려하여 한글을 자연스럽게 발음하는 입술 애니메이션 방법을 제안한다. 비디오 카메라로 발음하는 동안의 음소의 움직임들을 측정하고 입술 움직임 제어 파라미터들을 추출한다. 각각의 제어 파라미터들은 L fqvist의 스피치 생성 제스처 이론(speech production gesture theory)을 이용하여 실제 음소의 입술 움직임에 근사한 움직임인 지배함수(dominance function)들로 정의되고 입술 움직임을 애니메이션할 때 사용된다. 또, 각 지배함수들은 혼합함수(blending function)와 반음절에 의한 한글 합성 규칙을 사용하여 결합하고 동시조음이 적용된 한글을 발음하게 된다. 따라서 스피치 생성 제스처 이론을 이용하여 입술 움직임 모델을 구현한 방법은 기존의 보간에 의해 중간 움직임을 생성한 방법보다 실제 움직임에 근사한 움직임을 생성하고 동시조음도 고려한 움직임을 보여준다.Abstract The existing lip animation method of Hangul classifies the shape of lips with a few shapes and implements the lip animation with interpolating them. However it doesn't represent natural lip animation because the function of the real motion of lips, during articulation, isn't linear or simple non-linear function. It doesn't also represent the motion of lips varying among phonemes because it doesn't consider coarticulation. In this paper we present a new coarticulation model for the natural lip animation of Hangul. Using two video cameras, we film the speaker's lips and extract the lip control parameters. Each lip control parameter is defined as dominance function by using L fqvist's speech production gesture theory. This dominance function approximates to the real lip animation of a phoneme during articulation of one and is used when lip animation is implemented. Each dominance function combines into blending function by using Hangul composition rule based on demi-syllable. Then the lip animation of our coarticulation model represents natural motion of lips. Therefore our coarticulation model approximates to real lip motion rather than the existing model and represents the natural lip motion considered coarticulation.

• Journal of Digital Contents Society
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• v.17 no.2
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• pp.71-79
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• 2016

This study is intended to compare the performances of sound source localization methods used for stable automobile control by improving voice recognition rate in automobile environment and suggest how to improve their performances. Generally, sound source location estimation methods employ the TDOA algorithm, and there are two ways for it; one is to use a cross correlation function in the time domain, and the other is GCC-PHAT calculated in the frequency domain. Among these ways, GCC-PHAT is known to have stronger characteristics against echo and noise than the cross correlation function. This study compared the performances of the two methods above in automobile environment full of echo and vibration noise and suggested the use of a median filter additionally. We found that median filter helps both estimation methods have good performances and variance values to be decreased. According to the experimental results, there is almost no difference in the two methods' performances in the experiment using voice; however, using the signal of a song, GCC-PHAT is 10% more excellent than the cross correlation function in terms of the recognition rate. Also, when the median filter was added, the cross correlation function's recognition rate could be improved up to 11%. And in regarding to variance values, both methods showed stable performances.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.292-294
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• 2018

Recently, damage caused by natural disasters such as fire, earthquake, heavy rains and heavy snow is increasing. In addition, traffic accidents due to freezing, fog and fire in tunnels and bridges are frequently occurring. In such a disaster situation, it is very important to take prompt action by the person in charge of managing the facility and area.To this end, a disaster broadcasting system is used, but in the existing system, the broadcasting room and the speaker are connected by a wired connection. Also, the person in charge has to be in the broadcasting room to broadcast, which has a problem of delaying the time. In this paper, we design a disaster broadcasting system using LTE modem. The designed system enables a broadcasting person to make a call to a broadcasting system from anywhere using a cellular phone and a public telephone. Broadcasting via telephone is possible only with the telephone number pre-registered in the system and can be registered / deleted by the administrator. The registered telephone number, incoming voice file, and announcement voice for automatic broadcasting are stored in the system internal SD memory for convenient management. This disaster broadcasting system is expected to contribute to quick and convenient disaster broadcasting.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.307-314
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• 2003

In the general multiple video object coder, more interested objects such as speaker or moving object is consistently coded with higher priority. Since the priority of each object may not be fixed in the whole sequence and be variable on frame basis, it must be adjusted in a frame. In this paper, we analyze the independent rate control algorithm and global algorithm that the QP value is controled by the static parameters, object importance or priority, target PSNR, weighted distortion. The priority among static parameters is analyzed and adjusted into dynamic parameters according to the visual interests or importance obtained by camera interface. Target PSNR and weighted distortion are proportionally derived by using magnitude, motion, and distortion. We apply those parameters for the weighted distortion control and the priority-based control resulting in the efficient bit-rate distribution. As results of this paper, we achieved that fewer bits are allocated for video objects which has less importance and more bits for those which has higher visual importance. The duration of stability in the visual quality is reduced to less than 15 frames of the coded sequence. In the aspect of PSNR, the proposed scheme shows higher quality of more than 2d13 against the conventional schemes. Thus the coding scheme interfaced to human- eye proves an efficient video coder dealing with the multiple number of video objects.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.3
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• pp.304-315
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• 2001

In this paper, we peformed the real-time implementation of AMR(Adaptive Multi-Rate) speech coding algorithm which is adopted for IMT-2000 service using TMS320C6201, i.e., a Texas Instrument´s fixed-point DSP. With the ANSI C source code released from ETSI, optimization is performed to make it run in real-time with memory as small as possible using the C compiler and assembly language. Implemented AMR speech codec has the size of 32.06 kWords program memory, 9.75 kWords data RAM memory, and 19.89 kWords data ROM memory. And, The time required for processing one frame of 20 ms length speech data is about 4.38 ms, and it is short enough for real-time operation. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.375-382
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• 2003

Lip-reading technique based on bimodal is to enhance speech recognition rate in noisy environment. It is most important to detect the correct lip-image. But it is hard to estimate stable performance in dynamic environment, because of many factors to deteriorate Lip-reading's performance. There are illumination change, speaker's pronunciation habit, versatility of lips shape and rotation or size change of lips etc. In this paper, we propose the IIR filtering in time-domain for the stable performance. It is very proper to remove the noise of speech, to enhance performance of recognition by digital filtering in time domain. While the lip-reading technique in whole lip image makes data massive, the Principal Component Analysis of pre-process allows to reduce the data quantify by detection of feature without loss of image information. For the observation performance of speech recognition using only image information, we made an experiment on recognition after choosing 22 words in available car service. We used Hidden Markov Model by speech recognition algorithm to compare this words' recognition performance. As a result, while the recognition rate of lip-reading using PCA is 64%, Time-domain filter applied to lip-reading enhances recognition rate of 72.4%.

• The Journal of the Acoustical Society of Korea
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• v.15 no.4
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• pp.24-31
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• 1996

In this paper, we present a speech translation system for hotel reservation, KT_STS(Korea Telecom Speech Translation System). KT-STS is a speech-to-speech translation system which translates a spoken utterance in Korean into one in Japanese. The system has been designed around the task of hotel reservation(dialogues between a Korean customer and a hotel reservation de나 in Japan). It consists of a Korean speech recognition system, a Korean-to-Japanese machine translation system and a korean speech synthesis system. The Korean speech recognition system is an HMM(Hidden Markov model)-based speaker-independent, continuous speech recognizer which can recognize about 300 word vocabularies. Bigram language model is used as a forward language model and dependency grammar is used for a backward language model. For machine translation, we use dependency grammar and direct transfer method. And Korean speech synthesizer uses the demiphones as a synthesis unit and the method of periodic waveform analysis and reallocation. KT-STS runs in nearly real time on the SPARC20 workstation with one TMS320C30 DSP board. We have achieved the word recognition rate of 94. 68% and the sentence recognition rate of 82.42% after the speech recognition tests. On Korean-to-Japanese translation tests, we achieved translation success rate of 100%. We had an international joint experiment in which our system was connected with another system developed by KDD in Japan using the leased line.

• Sijohaknonchong
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• v.19 no.1
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• pp.155-177
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• 2003

Jeong Geuk-in was a poet of the early Joseon period. He lived for 45 years before Hangeul was published and 35 years afterwards. So, he wrote poetry both in Chinese and Korean. He was a creative writer who wrote Korean poems and songs. There were only a few works written in Korean including and before him. His Korean poems are , and . He created Korean poems and songs by unifying three literary forms of Sijo, Gyeong-gi-che-ga and Gasa. This study was intended to examine written in Korean. For the study, the form of the Bul-woo-heon-ga was analyzed and it was considered as Saseolsijo (a form of sijo with no restrictions on the length of the first two verses) for genre classification. However, it is generally thought that the Saseolsijo appeared in the seventeenth century. Therefore, this study is to explain the reason why Bul-woo-heon-ga is included in Saseolsijo. Another problem is that the writer of Bul-woo-heon-ga is not Jeong Geul-in, because of the fact that the speaker who appears in Bul-woo-heon-ga admired Jeong Geuk-in. In general, people do not admire themselves. As Jeong Geuk-in is a subject to be admired in the book, it is thought that the writer of the book is considered as one of his pupils or friends.