So-Yeon Min

A Study on the protection of personal information using a Virtual IDs in an anonymous bulletin board

The argument related to the use of real and anonymous names on the Internet bulletin board has recently become a main issue. When using real names, it is possible to violate free discussion and privacy. Also, when using anonymous names, it is possible to have the reverse function of the Internet in regard to the use of malicious replies or the distribution of false ideas. Therefore, this paper has made it possible to prevent the spread of the users personal information and execute the single log-in process by using the XML-token method which is one of the SSO technologies. Also, by issuing virtual IDs and forming the path when establishing tokens, the anonymous bulletin board which provides anonymity with a conditional tracing process has been suggested. After analyzing the performance of visitor numbers at authentication time, the anonymous bulletin board based on the group signature method showed the average response rate of 0.72 seconds, 0.18 seconds, which was suggested scheme. In the authentication time 4-5 times faster response speed, respectively. Also, since the suggested system does not have to provide a single authentication process or make the user provide his or her signature, the level of users convenience seems to be much higher. Such a result shows that the system suggested on the anonymous bulletin board has a more appropriate level of users convenience.

A Study on Implementation of Emotional Speech Synthesis System using Variable Prosody Model

Abstract This paper is related to the method of adding a emotional speech corpus to a high-quality large corpus based speech synthesizer, and generating various synthesized speech. We made the emotional speech corpus as a form which can be used in waveform concatenated speech synthesizer, and have implemented the speech synthesizer that can be generated various synthesized speech through the same synthetic unit selection process of normal speech synthesizer. We used a markup language for emotional input text. Emotional speech is generated when the input text is matched as much as the length of intonation phrase in emotional speech corpus, but in the other case normal speech is generated. The BIs(Break Index) of emotional speech is more irregular than normal speech. Therefore, it becomes difficult to use the BIs generated in a synthesizer as it is. In order to solve this problem we applied the Variable Break[3] modeling. We used the Japanese speech synthesizer for experiment. As a result we obtained the natural emotional synthesized speech using the break prediction module for normal speech synthesize.

An Algorithm on Improving a Pitch Searching by Energy Compensation in a Frame for Vocoder

Abstract It is important to search a pitch for vocoder. The major drawback to vocoders is their large computational requirements in searching a pitch and a codebook. In this paper, a simple method is proposed to improve the pitch searching process in the pitch filter almost without degradation of quality. The period of speech signal is emphasized by using Dual Pulse technique, the same type of autocorrelation method, in pitch search. Sometimes the incorrect pitch can be obtained by halving, doubling and trifling, To solve it, before searching a pitch, we estimate energy rate in a frame and compensate envelop of signal with it. By using the proposed algorithm in pitch search, its required computation are reduced and searching pitch is improved Key Words : Pitch search, Autocorrelation, Dual pulse technique * Corresponding Author : Myung-Jin Bae Tel: +82-2-824-0902 email: mjbae@ssu.ac.kr 접수일 12년 05월 29일 수정일 (1차 12년 07월 03일, 2차 12년 07월 11일) 게재확정일 12년 07월 12일 1. 서론 음성신호를 메모리에 저장하거나 전송하기 위한 음성 부호화법에는 크게 파형부호화법, 신호원부호화법, 혼성부호화법등 세 가지로 나눌 수 있다. 파형부호화법은 음성의 성분 분리 없이 파형 자체의 잉여 성분만을 제거한 후 부호화하여 전송하고 다시 합성하는 방식으로 고음질과 화자의 개성이 유지되는 반면 전송파형을 유지하기 위한 데이터 량이 많기 때문에 전송률이 높으며 대용량의 메모리가 필요하게 된다[1]. 이에 비해 신호원 부호화법은 음성의 발성모델에 근거하여 음성 신호의 여기 성분과 여파기 성분을 분석하여 각각을 독립적으로 분리시켜 부호화하는 방법을 사용하기 때문에 전송 대역폭이 작고 메모리 사용량이 작다. 그러나 분석 및 합성 시에

A study on the reduction of LSP (line spectrum pair) computation using adjustment of search band sequence and interval

We develop a computation reduction method for the real root method that is mainly used in the CELP (code excited linear prediction) vocoder. The real root method is that if polynomial equations have real roots, we are able to find those and transform them into LSP (line spectrum pairs). However, this method takes much time to compute, because the root searching is processed sequentially in the frequency region. But the important characteristic of LSP is that most coefficients occur in a specific frequency region. We suggest a method for reducing the LSP transformation time using voice characteristics. In the case of the silence period, the distribution of LSP parameters is uniform. In the case of the voice activity period, distribution of LSP parameters is not sequential. The proposed method applies search order and interval differently according to the distribution of LSP parameters. Also in the case of vowels, the search band can be controlled by the relation between the first formant and the second formant. When we compare this proposed method with the conventional real root method, the experimental result is that the searching time was reduced about 46% on average.

The high-speed LSF transformation algorithm in CELP vocoder

We propose a computation reduction method of the real root method that is mainly used in the CELP (code excited linear prediction) vocoder. The real root method is that if polynomial equations have real roots, we are able to find those and transform them into the LSF. However, this method takes much time to compute, because the root searching is processed sequentially in the frequency region. But, the important characteristic of the LSF is that most of the coefficients occur in a specific frequency region. So, the searching frequency region is ordered by each coefficients distribution, and the coefficients are searched in the ordered frequency region. The transformation time can be reduced further by this method than by the sequential searching method in the frequency region. When we compare this proposed method with the conventional real root method, the experimental result is that the searching time was reduced by about 46% in average.

A Design of Authority Management Protocol for Secure Storage Access Control in Cloud Environment

With the enhancements in existing major industries, cloud computing-based converging services have been created, as well as value-added industries. A variety of converging services are now provided, from personalized services up to industrial services. In Korea, they have become the driving force behind existing industries throughout the whole economy, but mainly in finance, mobile systems, social computing, and home services, based on cloud services. However, not only denial of service (DOS) and distributed DOS (DDOS) attacks have occurred, but also attack techniques targeting core data in storage servers. Even security threats that are hardly detected, such as multiple attacks on a certain target, APT, and backdoor penetration have also occurred. To supplement defenses against these, in this article, a protocol for authority management is designed to provide users with safe storage services. This protocol was studied in cases of integration between a cloud environment and big data-based technology, security threats, and their requirements. Also studied were amalgamation examples and their requirements in technology-based cloud environments and big data. With the protocol suggested, based on this, security was analyzed for attack techniques that occur in the existing cloud environment, as well as big data-based techniques, in order to find improvements in session key development of approximately 55%.

A Design of Key Generation and Communication for Device Access Control based on Smart Health Care

Smart healthcare systems, a convergent industry based on information and communications technologies (ICT), has emerged from personal health management to remote medical treatment as a distinguished industry. The smart healthcare environment provides technology to deliver vital information, such as pulse rate, body temperature, health status, and so on, from wearable devices to the hospital network where the physician is located. However, since it deals with the patients personal medical information, there is a security issue for personal information management, and the system may be vulnerable to cyber-attacks in wireless networks. Therefore, this study focuses on a key-development and device-management system to generate keys in the smart environment to safely manage devices. The protocol is designed to provide safe communications with the generated key and to manage the devices, as well as the generated key. The security level is analyzed against attack methods that may occur in a healthcare environment, and it was compared with existing key methods and coding capabilities. In the performance evaluation, we analyze the security against attacks occurring in a smart healthcare environment, and the security and efficiency of the existing key encryption method, and we confirmed an improvement of about 15%, compared to the existing cipher systems.

A Implement of Integrated Management Systems for User Fraud Protection and Malware Infection Prevention

The Internet continues to grow and develop, but there are going to generate a variety of Internet attacks that exploit it. In the initial Internet environment, the attackers maliciously exploited Internet environments for ostentations and hobbies. but these days many malicious attempts purpose the financial gain so systematic and sophisticated attacks that are associated with various crimes are occurred. The structures, such as viruses and worms were present in the form of one source multi-target before. but recently, APT(Advanced Persistent Threat, intelligent continuous attacks) in the form of multi-source single target is dealing massive damage. The performance evaluation analyzed whether to generate audit data and detect integrity infringement, and false positives for normal traffic, process detecting and blocking functions, and Agent policy capabilities with respect to the application availability.

Analysis of Voice Color Similarity for the development of HMM Based Emotional Text to Speech Synthesis

Maintaining a voice color is important when compounding both the normal voice because an emotion is not expressed with various emotional voices in a single synthesizer. When a synthesizer is developed using the recording data of too many expressed emotions, a voice color cannot be maintained and each synthetic speech is can be heard like the voice of different speakers. In this paper, the speech data was recorded and the change in the voice color was analyzed to develop an emotional HMM-based speech synthesizer. To realize a speech synthesizer, a voice was recorded, and a database was built. On the other hand, a recording process is very important, particularly when realizing an emotional speech synthesizer. Monitoring is needed because it is quite difficult to define emotion and maintain a particular level. In the realized synthesizer, a normal voice and three emotional voice (Happiness, Sadness, Anger) were used, and each emotional voice consists of two levels, High/Low. To analyze the voice color of the normal voice and emotional voice, the average spectrum, which was the measured accumulated spectrum of vowels, was used and the F1(first formant) calculated by the average spectrum was compared. The voice similarity of Low-level emotional data was higher than High-level emotional data, and the proposed method can be monitored by the change in voice similarity.

A Study on the Improvements of the Speech Quality by using Distribution Characteristics of LSP parameters in the EVRC(Enhanced Variable Rate Codec)

To improve the efficiency of the channel spectrum and to reduce the power consumption of the system in EVRC, the voice signal is compressed and transmitted only when the user speaks to. In addition to this, voice frames are divided into three rates 1, 1/2 and 1/8 and each frame is handled differently. For example, we assumed that the input is silence region if the 1/8 rate is used. In this paper, the sections are firstly separated into the voiced speech signal region, unvoiced speech signal region, and silence region by using distribution characteristics of LSP parameters. Then the paper suggested to encode 1 rate for the voiced speech signal, 1/2 rate for the unvoiced speech signal region, 1/8 rate for the silence region. In other words, traditional way of transmission is used when sending full rate in the EVRC. However, when sending half rate, the voice is firstly distinguished between voiced and unvoiced. If the voice is distinguished as voiced, voice is converted into full rate before the transmission. If it is distinguished as silence, EVRC`s basic rate is applied. In the experimental results with SNR, ASDM, transmission bit rate measurement, we have demonstrated that voice quality was improved by using the proposed algorithm.