• Journal of Communications and Networks
• /
• v.10 no.3
• /
• pp.287-296
• /
• 2008

The IEEE 802.11 wireless standard uses the carrier sense multiple access with collision avoidance (CSMA/CA) as its MAC protocol (during the distributed coordination function period). This protocol is an adaptation of the CSMA/CD of the wired networks. CSMA/CA mechanism cannot guarantee quality of service (QoS) required by the application because orits random access method. In this study, we propose a new MAC protocol that considers different types of traffic (e.g., voice and data) and for each traffic type different priority levels are assigned. To improve the QoS of IEEE 802.11 MAC protocols over a multi-channel CSMA/CA, we have developed a new admission policy for both voice and data traffics. This protocol can be performed in direct sequence spread spectrum (DSSS) or frequency hopping spread spectrum (FHSS). For voice traffic we reserve a channel, while for data traffic the access is random using a CSMA/CA mechanism, and in this case a selective reject and push-out mechanism is added to meet the quality of service required by data traffic. To study the performance of the proposed protocol and to show the benefits of our design, a mathematical model is built based on Markov chains. The system could be represented by a Markov chain which is difficult to solve as the state-space is too large. This is due to the resource management and user mobility. Thus, we propose to build an aggregated Markov chain with a smaller state-space that allows performance measures to be computed easily. We have used stochastic comparisons of Markov chains to prove that the proposed access protocol (with selective reject and push-out mechanisms) gives less loss rates of high priority connections (data and voices) than the traditional one (without admission policy and selective reject and push-out mechanisms). We give numerical results to confirm mathematical proofs.

• Proceedings of the IEEK Conference
• /
• 2003.11c
• /
• pp.353-356
• /
• 2003

As the wireless mobile telecommunication system has been developed with astonishment, its offering service has also widely been expanded including various data service. Currently, the wireless mobile telecommunication network presents voice service that covers for the most part of the whole service areas. For this reason, the availability of the switching capacity in the mobile switching center(MSC) is manipulated by the required volume of voice service. However, considering the increase of data service, it is desirable for the current switching method to be modified for more efficiency. In this Paper, we analyze the data traffic caused by providing data service in the wireless mobile telecommunication network. For this, we are firstly going to review the result of the analysis in the feature of the data traffic. Secondly, based on the review, we are also going to perform analyzing the other feature of the data traffic normally generated in the wireless mobile telecommunication network. We expect that this paper would be utilized as an elementary source for the feature of the SMS data .traffic and it will be an honour for ourselves to work on it.

• Journal of the Korea Institute of Military Science and Technology
• /
• v.13 no.4
• /
• pp.577-585
• /
• 2010

The Military Tactical Communication technology for effective network-centric warfare is developing. Targeting broadband wireless transmission, core technology for connection, and Transmission technology that secure survivability under High-speed Movement environment. On the one hand, Tactical data communication system that reflects military characteristic is developing on the base of Legacy communication equipment which is used in the field. Because almost every military units in the field have used voice to communicate which lower efficiency of operation, they have made effort to Substitute voice communication which delays military Operation Tempo to digital communication. The Communications environment of troops in Forward edge of battle field area is very poor. Especially in terms of limited frequency allocation and bandwidth. Therefore, improving the efficiency of frequency is essential for Military Tactical Communication. This paper is about The Data Compression Method for increase of Efficiency in Tactical Data Communication over Legacy Radios which are UHF, VHF, HF Radio. I proposed and proved the most efficient Data Compression Method that reflects military characteristic, after analyzing the experimentation, which simulate CAS(Close Air Support mission) data transmission between Pilot and TACP.

• Cartoon and Animation Studies
• /
• s.16
• /
• pp.165-178
• /
• 2009

It seems like that the main stream of animation for theaters transferred from cells to 30 animations. It can be easily seen if current situation is compared to the time when al! works were Cell animations. Although earlier papers were mostly about studies on Cell animation works, future studies will publish papers related to 30. Accordingly, this study studied the association of the researcher's paper with 30 works while It had been applied to 20. Then, based on the data, this study analyzed how major/supporting characters were transformed to animation characters in the scenes that were animated at the beginning and end of film that were reieased in 2007. This result was compared with the character made using a voice actor in the Pixar's long piece of animation for theater that had been already screened. The data were made into values to some extent in an attempt to increase satisfaction in the result of analysis comparing the character with the voice actor. When making characters using actual persons, it is hoped that grasping the modeling factors of the images making 20/30 characters as indicated in this study will be helpful in producing animation characters.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.12 no.1
• /
• pp.225-230
• /
• 2012

The opinion mining is that to use the existing data mining technology also uploaded blog to web, to use product comment, the opinion mining can extract the author's opinion therefore it not judge text's subject, only judge subject's emotion. In this paper, published opinion mining algorithms and the text using speech recognition API for non-voice data to judge the emotions suggested. The system is open and the Subject associated with Google Voice Recognition API sunwihwa algorithm, the algorithm determines the polarity through improved design, based on this interview, speech recognition, which implements the model.

• The KIPS Transactions:PartD
• /
• v.15D no.1
• /
• pp.131-138
• /
• 2008

This paper describes the implementations of the first application service based on embedded Linux; Mobile Teller which uses WiBro network for data communications and CDMA network for voice communications. Currently, with the appearance of WiBro service, dual-mode terminals which support two heterogeneous networks are available. But, the development of applications which effectively use these networks for providing better service to user is rarely prepared. At Mobile Teller, when a sender on a dual-mode terminal types texts, the texts are transmitted to a TTS server located in the Internet through WiBro network. Subsequently, the TTS server converts the texts into voices and transmits the voice data to the dual-mode terminal. At last the dual-mode terminal sends the voice to the receiver through the CDMA network. In case of noisy environment or when a user has difficulty in speaking, Mobile Teller makes voice communication possible

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.16 no.6
• /
• pp.4164-4169
• /
• 2015

In this paper planed Real-time Shopping System using Multipurpose Visual Language System(MVLS) with voice recognition remote controller. This system has a merit that using existing on-line & off-line shopping system with addition MVLS data. This can realization little modification existing shopping system. Also, customer's a point of view that has a merit to using easy device for shopping. That is no more using difficult device like that keyboard or mouse, and approach to easy device that voice recognition remote controller or smart phone. Especially, aspect of the old and the infirm and disabled persons that information minority group, can easy buy the product using this system. And, the sellers can more easily collection customer's data and using that future sales strategy.

• Korean Journal of Otorhinolaryngology-Head and Neck Surgery
• /
• v.55 no.8
• /
• pp.498-507
• /
• 2012

Background and Objectives We aimed to develop a Korean version of the velopharyngeal insufficiency (VPI) speech corpus system. Subjects and Method After developing a 3-channel simultaneous speech recording device capable of recording nasal/oral and normal compound speech separately, voice data were collected from VPI patients aged more than 10 years with/without the history of operation or prior speech therapy. This was compared to a control group for which VPI was simulated by using a french-3 nelaton tube inserted via both nostril through nasopharynx and pulling the soft palate anteriorly in varying degrees. The study consisted of three transcriptors: a speech therapist transcribed the voice file into text, a second transcriptor graded speech intelligibility and severity and the third tagged the types and onset times of misarticulation. The database were composed of three main tables regarding (1) speaker's demographics, (2) condition of the recording system and (3) transcripts. All of these were interfaced with the Praat voice analysis program, which enables the user to extract exact transcribed phrases for analysis. Results In the simulated VPI group, the higher the severity of VPI, the higher the nasalance score was obtained. In addition, we could verify the vocal energy that characterizes hypernasality and compensation in nasal/oral and compound sounds spoken by VPI patients as opposed to that characgerizes the normal control group. Conclusion With the Korean version of VPI speech corpus system, patients' common difficulties and speech tendencies in articulation can be objectively evaluated. Comparing these data with those of the normal voice, mispronunciation and dysarticulation of patients with VPI can be corrected.

• Journal of Internet Computing and Services
• /
• v.4 no.5
• /
• pp.61-68
• /
• 2003

In this paper, we propose a wireless MAC protocol named APRMA, which is capable of supporting the ABR type data service and Maximizing channel utilization. Data terminals with random data packets are not provided slot reservation with PRMA protocol. That is, slot reservation is applicable to the time constraint voice packet exclusively. But the reservation scheme have to be performed for loss sensitive data packet, and contended their quality of service, Therefore, in wireless MAC, reservation technique has to be used for both voice and data services. So the terminal which wants to request for ABR type service, is allocated a minimum bandwidth from system for the first time, If the system have some extra available bandwidth, ABR terminals would acquire additional bandwidth slot by slot, As a result, APRMA protocol can support the data service with loss sensitivity and maintain their channel utilization high.

• KIPS Transactions on Software and Data Engineering
• /
• v.12 no.7
• /
• pp.303-314
• /
• 2023

Voice conversion, a technology that allows an individual's speech data to be regenerated with the acoustic properties(tone, cadence, gender) of another, has countless applications in education, communication, and entertainment. This paper proposes an approach based on the StarGAN-VC model that generates realistic-sounding speech without requiring parallel utterances. To overcome the constraints of the existing StarGAN-VC model that utilizes one-hot vectors of original and target speaker information, this paper extracts feature vectors of target speakers using a pre-trained version of Rawnet3. This results in a latent space where voice conversion can be performed without direct speaker-to-speaker mappings, enabling an any-to-any structure. In addition to the loss terms used in the original StarGAN-VC model, Wasserstein distance is used as a loss term to ensure that generated voice segments match the acoustic properties of the target voice. Two Time-Scale Update Rule (TTUR) is also used to facilitate stable training. Experimental results show that the proposed method outperforms previous methods, including the StarGAN-VC network on which it was based.