• Title/Summary/Keyword: Voice Quality

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A Study on the Design of Cyber lecture Component (가상강의 Component 설계에 관한 연구)

  • 강정배;김선경
    • Proceedings of the Korea Society of Information Technology Applications Conference
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    • 2002.11a
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    • pp.171-177
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    • 2002
  • E-Loaming is a modem main teaching method starting from the concept of remote education. This research is aimed for proposing cyber education library system, and designing a cyber education component that becomes a basis for e-Learning system. Cyber education library is a storage system of cyber lectures that can supply high quality data to the needed developers. Cyber education component consists of 5 categories and those are text, voice, image, animation, and flash. By using this system, the developers can save the necessary time and effort in education development. This system also helps students. The students can access various lecture data on a given subject and select the best fit for them.

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A Study on the Realization of Echo Canceller in CDMA Mobile Communication Networks (CDMA 이동통신 망에서의 반향제거기 구현에 관한 연구)

  • 유태훈;박광철;이윤희;김기두
    • Journal of the Institute of Electronics Engineers of Korea TE
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    • v.37 no.5
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    • pp.36-47
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    • 2000
  • The CDMA digital cellular systems provide better voice Quality than analog systems, however there exists inherent delays due to speech coding and transmission processing, which brings echoes returned by the BSC and PSTN interface. In this paper, we show the performance improvement of a proposed echo canceller by real time implementation, where Block Update NLMS algorithm is applied into the TMS320C54X DSP. By applying the proposed method into the practical mobile phone, we verify that various types of echoes (LE, ESE, AE) may be removed more precisely. We also cope with echo path change resulting from change of delay length after taking VAD to find echo path delay.

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Performance Analysis of Mobile IP Network Based on MPLS/WLAN for providing QoS (MPLS/WLAN기반 Mobile IP망에서 QoS 제공을 위한 성능분석)

  • Kim, Jin-Hae;Ye, Hwi-Jin;Cho, Sung-Joon
    • Journal of Advanced Navigation Technology
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    • v.12 no.6
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    • pp.591-597
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    • 2008
  • With the established voice service, the data service is increasing in the field of mobile telecommunications in these days. Also, the data service develops into the form of multimedia service which combines various media. However, it is limited to provide QoS(Quality of Service) for the multimedia using the established network which manages all the data in sam: rule. So, this paper suggests the network architecture which assures QoS of multimedia data by applying MPLS(Multi-Protocol Label Switching) to the core network. Especially, we analyze the efficiency of the network combining Mobile IP network based on the WLAN(Wireless LAN) and core network based on MPLS through by network simulations using QualNet.

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A Video Conference System using Multimedia Data and Screen Sharing Methods (멀티미디어 데이터 및 화면 공유 기법을 활용한 화상 회의 시스템)

  • Ko Kwang-San;Jang Jung-Soo;Jung Hoe-Kyung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.9 no.5
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    • pp.1012-1018
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    • 2005
  • According as computer industry develops and the Internet is spread information interchange between remote users video conference system that can exchange multimedia data such as voice as well as simplicity text data or burn by real-time appeared and is used in various field. However, most systems developed present have controversial points of that need expensive multi-point connection equipment or network resource of High band width. Hereupon, in this paper, designed and embody video conference system that can be stable and supplies multimedia information and screen public ownership environment of high-quality as software using minimum network resource without using expensive connection equipment.

A VoIP Traffic Generator for Simulating Call Processing in an IP Contact Center (IP 컨택 센터에서 통화 처리 모의 실험을 위한 VoIP 트래픽 생성기)

  • Jung, In-Hwan
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.6B
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    • pp.575-584
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    • 2009
  • In this paper, we design and implement a VoIP traffic generator for simulating call processing in IP contact center systems. Creating a VoIP call based on H.323 and SIP and generating RTP traffic which uses G.711 codec, the generator lets many users simulate situations on which they call each other. With this tool, which is named VoIPTG, users can combine H.323 or SIP session control protocol, the number of users, time variation, and voice codecs and then direct various situations for simulation. This traffic generator can be used for testing functions of an IP contact center and especially it is necessary for testing the quality of IP based call recording systems.

Design and Analysis of Mobile-IPv6 Multicasting Algorithm Supporting Smooth Handoff in the All-IP Network (All-IP망에서 Smooth Handoff를 지원하는 Mobile-IP v6 멀티캐스팅 알고리즘의 설계 및 분석)

  • 박병섭
    • The Journal of the Korea Contents Association
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    • v.2 no.3
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    • pp.119-126
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    • 2002
  • The QoS(Quality of Service) guarantee mechanism is one of critical issues in the wireless network. Real-time applications like VoIP(Voice over IP) in All-IP networks need smooth handoffs in order to minimize or eliminate packet loss as a Mobile Host(MH) transitions between network links. In this paper, we design a new multicasting algorithm using DB(Dynamic Buffering) mechanism for Mobile-IPv6. A key feature of the new protocol is the concepts of the DB and MRA(Multicast Routing Agent) to reduce delivery path length of the multicast datagram. Particularly, the number of tunneling and average routing length of datagram are reduced relatively, the multicast traffic load is also decreased.

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Characteristics of Phoniatrics in Patients with Spastic Dysarthria (경직형 마비말장애의 음성언어의학적 특성)

  • Kim, Sook-Hee;Kim, Hyun-Gi
    • Speech Sciences
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    • v.15 no.4
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    • pp.159-170
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    • 2008
  • The purpose of this study was to find out the ability of coordination of the articulatory motor and the ability of control of the respiration and laryngeal for spastic dysarthria by acoustic analysis. The sustained of vowel /a/ and repetition of syllable /pa/ in 15 normal and 10 spastic dysarthria were measured. Multi-Speech, MDVP, and MSP were used for data recording and analysis. As a result, the mean DDK rate in the spastic group was significantly slower than in the normal. The maximum phonation time in the spastic group ($4.80{\pm}1.94$) was shorter than in the normal ($11.20{\pm}3.72$). The DDKjit in the spastic group was significantly higher than in the normal. The DDKsla was reduced in the spastic group. The mean syllable duration in the spastic group (146.2ms) was significantly longer than in the normal (75.8ms). The mean energy was reduced in the spastic group. The range of Fo was greater than in the normal. The frequency perturbation (jitter, vFo) and amplitude perturbation (shimmer, vAm) were higher than in the normal group. The NHR was higher than in the normal group. The parameters of this were significantly difference between the spastic dysarthria and the normal (p<0.05). Finally, the spastic dysarthria has short respiration, slow speech rate, and voice quality problem. The these results will help to establish a plan and the intervention of treatment.

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A Novel Resource Allocation Algorithm in Multi-media Heterogeneous Cognitive OFDM System

  • Sun, Dawei;Zheng, Baoyu
    • 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.

A Study on the Design of Cyber lecture Component (가상강의 Component 설계에 관한 연구)

  • 강정배;김선경
    • Proceedings of the Korea Society for Industrial Systems Conference
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    • 2002.11a
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    • pp.171-177
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    • 2002
  • E-Learning is a modem main teaching method starting from the concept of remote education. This research is aimed for proposing cyber education library system, and designing a cyber education component that becomes a basis for e-Learning system. Cyber education library is a storage system of cyber lectures that can supply high quality data to the needed developers. Cyber education component consists of 5 categories and those are text, voice, image, animation, and flash. By using this system, the developers can save the necessary time and effort in education development. This system also helps students. The students can access various lecture data on a given subject and select the best fit for them.

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A Study of Speech Control Tags Based on Semantic Information of a Text (텍스트의 의미 정보에 기반을 둔 음성컨트롤 태그에 관한 연구)

  • Chang, Moon-Soo;Chung, Kyeong-Chae;Kang, Sun-Mee
    • Speech Sciences
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    • v.13 no.4
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    • pp.187-200
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    • 2006
  • The speech synthesis technology is widely used and its application area is also being broadened to an automatic response service, a learning system for handicapped person, etc. However, the sound quality of the speech synthesizer has not yet reached to the satisfactory level of users. To make a synthesized speech, the existing synthesizer generates rhythms only by the interval information such as space and comma or by several punctuation marks such as a question mark and an exclamation mark so that it is not easy to generate natural rhythms of people even though it is based on mass speech database. To make up for the problem, there is a way to select rhythms after processing language from a higher level information. This paper proposes a method for generating tags for controling rhythms by analyzing the meaning of sentence with speech situation information. We use the Systemic Functional Grammar (SFG) [4] which analyzes the meaning of sentence with speech situation information considering the sentence prior to the given one, the situation of a conversation, the relationship among people in the conversation, etc. In this study, we generate Semantic Speech Control Tag (SSCT) by the result of SFG's meaning analysis and the voice wave analysis.

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