• 한국정보통신설비학회:학술대회논문집
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• 2006.08a
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• pp.249-254
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• 2006

As the concept of Ubiquitous City has become an important issue, we need to have a voice in the field of privacy. The Ubiquitous Computing revolution, it is achieved a physical space and an intelligent union of cyber space in the space where an off-line space was integrated with on-line. It is combined with a life space naturally, and Ubiquitous creates a new integrated space. The space is the opportunity space that is limitless as soon as it is the unknown world that mankind was not able to experience yet. According to the recent papers and studies, Privacy or Personal Information sector is mentioned as a major problem for Ubiquitous Computing Society. Therefore, this paper checks Ubiquitous City, the present privacy concept and situation of currently proceeding Ubiquitous computing environment. Also, The main purpose of this paper is to analyze the concepts privacy, personal information. Thus this paper treated the analysis of case, technological or social issue, problem and solution, competing values of privacy and so on.

• The Journal of Korea Robotics Society
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• v.13 no.1
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• pp.16-25
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• 2018

The talking head (TH) indicates an utterance face animation generated based on text and voice input. In this paper, we propose the generation method of TH with facial expression and intonation by speech input only. The problem of generating TH from speech can be regarded as a regression problem from the acoustic feature sequence to the facial code sequence which is a low dimensional vector representation that can efficiently encode and decode a face image. This regression was modeled by bidirectional RNN and trained by using SAVEE database of the front utterance face animation database as training data. The proposed method is able to generate TH with facial expression and intonation TH by using acoustic features such as MFCC, dynamic elements of MFCC, energy, and F0. According to the experiments, the configuration of the BLSTM layer of the first and second layers of bidirectional RNN was able to predict the face code best. For the evaluation, a questionnaire survey was conducted for 62 persons who watched TH animations, generated by the proposed method and the previous method. As a result, 77% of the respondents answered that the proposed method generated TH, which matches well with the speech.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.5
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• pp.691-708
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• 2010

An important issue of supporting multi-users with diverse quality-of-service (QoS) requirements over wireless networks is how to optimize the systematic scheduling by intelligently utilizing the available network resource while, at the same time, to meet each communication service QoS requirement. In this work, we study the problem of a variety of communication services over multi-media heterogeneous cognitive OFDM system. We first divide the communication services into two parts. Multimedia applications such as broadband voice transmission and real-time video streaming are very delay-sensitive (DS) and need guaranteed throughput. On the other side, services like file transmission and email service are relatively delay tolerant (DT) so varying-rate transmission is acceptable. Then, we formulate the scheduling as a convex optimization problem, and propose low complexity distributed solutions by jointly considering channel assignment, bit allocation, and power allocation. Unlike prior works that do not care computational complexity. Furthermore, we propose the FAASA (Fairness Assured Adaptive Sub-carrier Allocation) algorithm for both DS and DT users, which is a dynamic sub-carrier allocation algorithm in order to maximize throughput while taking into account fairness. We provide extensive simulation results which demonstrate the effectiveness of our proposed schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4B
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• pp.355-363
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• 2006

A rapidly increasing demand of voice and data through mobile telecommunication service and the development of technologies related with the service are sufficiently said to make the service become an indispensible communicating means, not just selective one. However, in the other side of the generalizing of mobile telecommunication service, there is said to be a worsening situation of unbalanced benefits due to the problem of information difference between the non. The more advanced mobile telecommunication service proceeds, the more deepens the problem, and it is generally known that the problem cannot be solved by the ability of free market itself as seen in the wired telecommunication market. Therefore, we need to set up a systematic policy of mobile telecommunication service for solving this problem, and also think over whether to include the service within the range of universal service. In this study, we examine the required factors for including the service within the range of universal service, and analyze the validity of the including in the view point of institutional aspects.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.11
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• pp.830-835
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• 2006

Because of its exponential growth of data and voice transmissions through wireless communications, efficient resource management became more important factor when we design wireless networks. One of those limited resources in the wireless communications is frequency bandwidth. As a solution of increasing reusability of resources, the efficient frequency assignment problems on wireless networks have been widely studied. One suitable approach to solve these frequency assignment problems is transforming the problem into traditional graph coloring problems in graph theory. However, most of frequency assignments on arbitrary network topology are NP-Complete problems. In this paper, we consider the Chromatic Bandwidth Problem on the cellular topology wireless networks. It is known that the lower bound of the necessary number of frequencies for this problem is $O(k^2)$. We prove that the lower bound of the necessary number of frequencies for the Chromatic Bandwidth Problem is $O(k^3)$ which is tighter lower bound than the previous known result.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.7
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• pp.1118-1130
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• 1999

In this paper, error performance of DS/FH CDMA system has been analyzed in a radio channel which is characterized by near-far problem and multi-path fading. The DS/FH CDMA system adopts Maximum Ratio Combining(MRC) diversity and BCH(Bose-Chau dhuri-Hocquenghem) coding techniques to enhance system performance. Using the derived error probability equation, the error performance of DS/FH CDMA system has been evaluated and shown in figures to discuss as a function of PN code length(N), hopping rate(q), number of diversity branch(M), coding rate($\gamma$) and bit energy per noise power ratio ${E_b/N_o}$. The results show that DS/FH system is more effective to restrain the affection of near-far problem and multi-path fading than DS system. And there is a substantial enhancement in performance by employing an MRC diversity or BCH coding techniques. Consequently, we expected that proposed system structure is reliable to the voice communication system in near-far problem and multi-path fading channel.

• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.8
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• pp.3728-3752
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• 2016

Wireless mesh networks (WMNs) have been increasingly applied in private and public networks during the last decade. In a different context, voice over IP (VoIP) has emerged as a new technology for making voice calls around the world over IP networks and is replacing traditional telecommunication systems. The popularity of the two technologies motivated the deployment of VoIP over WMNs. However, VoIP over WMNs suffers from inefficient bandwidth utilization because of two reasons: i) attaching 40-byte RTP/UDP/IP header to a small VoIP payload (e.g., 10 bytes) and ii) 841 μs delay overhead of each packet in WMNs. Among several solutions, VoIP packet multiplexing is the most prominent one. This technique combines several VoIP packets in one header. In this study, we will survey all the VoIP multiplexing methods over WMNs. This study provides a clear understanding of the VoIP bandwidth utilization problem over WMNs, discusses the general approaches in which packet multiplexing methods could be performed, provides a detailed study of present multiplexing techniques, shows the aspects that hinder the VoIP multiplexing methods, discusses the factors affected by VoIP multiplexing schemes, shows the merits and demerits of different multiplexing approaches, provides guidelines for designing a new improved multiplexing technique, and provides directions for future research. This study contributes by providing guidance for designing a suitable and robust method to multiplex VoIP packets over WMNs.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.74-82
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• 1995

This paper is study on the voice dialing using HMM and post processing of the connected digits. HMM algorithm is widely used in the speech recognition with a good result. But, the maximum likelihood estimation of HMM(Hidden Markov Model) training in the speech recognition does not lead to values which maximize recognition rate. To solve the problem, we applied the post processing to segmental K-means procedure are in the recognition experiment. Korea connected digits are influenced by the prolongation more than English connected digits. To decrease the segmentation error in the level building algorithm some word models which can be produced by the prolongation are added. Some rules for the added models are applied to the recognition result and it is updated. The recognition system was implemented with DSP board having a TMS320C30 processor and IBM PC. The reference patterns were made by 3 male speakers in the noisy laboratory. The recognition experiment was performed for 21 sort of telephone number, 252 data. The recognition rate was $6\%$ in the speaker dependent, and $80.5\%$ in the speaker independent recognition test.

• The Journal of the Korea Contents Association
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• v.6 no.3
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• pp.1-11
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• 2006

With the improvement of information telecommunication technology, the various service in broadband integrated services digital networks have a wide range of delay, delay jitter and cell loss probability requirements according to traffic specification. Therefore, the design of appropriate control schemes that can satisfy the cell loss, delay requirements with various traffic specification for B-ISDN is an extremely important challenging problem. In this paper, we propose a priority control scheme using a window counter and a cell counter per each type of class. In the proposed priority control scheme, for satisfying required service quality, we performed the priority control scheme using the delay/loss factors obtained by comparing window counter with cell counter. The performance of proposed control scheme is estimated by computer simulation. In the results of simulation, we verified that the proposed method satisfied per class requirements as the results showed that cell loss probability has a order of video, data, voice and delay time has a order of video, voice and data.