• Title/Summary/Keyword: Voice problem

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Simulation Analysis for Verifying an Implementation Method of Higher-performed Packet Routing

  • Park, Jaewoo;Lim, Seong-Yong;Lee, Kyou-Ho
    • Proceedings of the Korea Society for Simulation Conference
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    • 2001.10a
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    • pp.440-443
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    • 2001
  • As inter-network traffics grows rapidly, the router systems as a network component becomes to be capable of not only wire-speed packet processing but also plentiful programmability for quality services. A network processor technology is widely used to achieve such capabilities in the high-end router. Although providing two such capabilities, the network processor can't support a deep packet processing at nominal wire-speed. Considering QoS may result in performance degradation of processing packet. In order to achieve foster processing, one chipset of network processor is occasionally not enough. Using more than one urges to consider a problem that is, for instance, an out-of-order delivery of packets. This problem can be serious in some applications such as voice over IP and video services, which assume that packets arrive in order. It is required to develop an effective packet processing mechanism leer using more than one network processors in parallel in one linecard unit of the router system. Simulation analysis is also needed for verifying the mechanism. We propose the packet processing mechanism consisting of more than two NPs in parallel. In this mechanism, we use a load-balancing algorithm that distributes the packet traffic load evenly and keeps the sequence, and then verify the algorithm with simulation analysis. As a simulation tool, we use DEVSim++, which is a DEVS formalism-based hierarchical discrete-event simulation environment developed by KAIST. In this paper, we are going to show not only applicability of the DEVS formalism to hardware modeling and simulation but also predictability of performance of the load balancer when implemented with FPGA.

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Implementation of Hands-Free Phone in a Car Using DSP (DSP를 이용한 차량용 핸즈프리 전화기의 구현)

  • Hong, Ki-Jun;Roh, Yi-Ju;Jeong, Kyung-Hoon;Kang, Dong-Wook;Yun, Kee-Bang;Kim, Ki-Doo
    • 전자공학회논문지 IE
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    • v.44 no.4
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    • pp.1-10
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    • 2007
  • In this thesis, we study the implementation of hands-free phone in a car, taking acoustic echo canceller, in order to remove acoustic echo effectively. Conventional coustic echo canceller used for only adaptive filtering has much difficulty to solve both echo and double-talk problem. To tackle this problem, we propose acoustic echo canceller consisting of adaptive filter using a modified NLMS, VAD to catch exact voice activity duration using two independent forgetting factors, double-talk detector to detect fast and precise double talk duration using cross-correlation between microphone signal and residual echo, and output controller using VAD and double-talk detector. The proposed hands-free phone taking acoustic echo canceller shows the performance that has not acoustic echo and guarantees full duplex.

Streaming Service Scheduling Scheme in Mobile Networks (모바일환경에서 실시간 데이타서비스를 위한 스케줄링 정책)

  • Min Seung-Hyun;Kim Myung-Jun;Bang Kee-Chun
    • Journal of Digital Contents Society
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    • v.3 no.1
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    • pp.47-57
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    • 2002
  • Recently, wireless networks have been pursuing multimedia data service as voice, data, image, video and various form of data according to development of information communication technology. It guarantees cell delivery delay of real time data in efficient real time multimedia data transfer. Also, it minimizes cell loss rate of non-real time multimedia data. In the wireless ATM, there are based on Asynchronous Transfer Mode(ATM). It implies that there are various service with difficult transmission rates and qualities in the wireless communication network. As a result, it is important to find out the ways to guarantee the Quality of Service(QoS) for each kind of traffic in wireless network. In this thesis, we propose an improved TCRM scheduling algorithms for transmission real-time multimedia data service in wireless ATM Networks. We appear real time multimedia scheduling policy that apply each different method to uplink and downlik to wireless ATM network. It can guarantee QoS requirements for each real time data and non-real time data. It also deals the fairness problem for sharing the scarce wireless resources. We solve fault of TCRM as inefficient problem of non-real data by using arbitrary transmission speed and RB(Reservation Buffer) through VC(Virtual Control) and BS(Base Station).

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A Study on Wireless PKI Technology Standard (무선 PKI 기술 표준에 관한 연구)

  • Sung, Yeon-Guk;Kim, Hyun-Chul;Jung, Jin-Wook;Kim, Soon-Chul;Ryu, Won
    • Convergence Security Journal
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    • v.2 no.2
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    • pp.29-38
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    • 2002
  • Everyday demand of wireless internet is increasing. Security problem is certainly resolved for wireless internet activation. Especially problem is only wiretap in mobile communication with voice, but problems, user authentication, data integrity guarantee etc., are resolved in data-services that have commercial transaction over simple data information service such bill, banking. Necessity of wireless PKI that can offer security service likely in wired environment is requested for offering security service in wireless environment. For offering security services, confidentiality, integrity, non-repudiation etc, that of offered in wired environment in wireless environment, first it must construct wireless PKI infrastructure and do service activity. This paper analyze various wireless internet technology for offering safe wireless internet service and wireless PH standards. Performance Result of this paper expect activity of safe wireless PH service and activity of electronic commercial transaction used wireless internet such banking service, bill transaction, online shopping.

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A Unified Method for Vocal Source Separation From Stereophonic Music Signals (스테레오 음악 신호에서의 보컬 음원 분리를 위한 통합 알고리즘)

  • Kim, Min-Je;Jang, In-Seon;Kang, Kyeong-Ok
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.47 no.5
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    • pp.89-99
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    • 2010
  • A unified method for separating musical sources, singing voice for example, from stereophonic mixtures is provided. We usually have two observed signals in stereophonic music contents, where more than two instruments are played together. If we regard each instrument as source, this problem becomes an underdetermined source separation problem and cannot be solved by conventional methods, which infers the spatial environment of the downmixing process happens. Instead, source-specific information has been exploited to recover a particular instrumental source. This paper provides a unifying structure consists of heterogenious ad-hoc separate algorithms, which are designed for separating vocal sources using stereophonic channel information and dominant pitch information of the sources, respectively. Experiments on real world music contents show that the proposed unification can neutralize the drawbacks of the two ad-hoc separation algorithms and finally enhance the separation results.

A Study of Enemy Aptitude of Pistol Sound Source for Space Estimation (공간평가를 위한 피스톨음원의 적정성에 관한 연구)

  • Shon, Jang-Ryul;Kim, Jung-Joong
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.15 no.3 s.96
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    • pp.320-328
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    • 2005
  • Last target of architectural acoustics is that people wish to convey voice effectively from the space adaptively in use purpose in building. But, how exactly through space sound (sound source) that wish to deliver from indoor can be passed method to do quantification and evaluate quantity of sound by method to serve indoor architectural acoustics estimation summer period and methods to estimate definition propose. This Study searches special quality of sound source about MLS signal that is occurred short-answer sound source (pistol sound source) and nondirectional speaker among indoor sound estimation method, and measure and analyzed reverberation time (RT60), definition (C80, D50) by regulation of each ISO 3382 in age place (classroom, hall, gymnasium). Analysis result and sound factor among could know that d of two sound sources converges in measurement error extent about reverberation time (RT60) of analysis incidental and sound factors and value shows change irregularly about sound factor of D50, C80, pistol sound source judged there is problem. Also, could know that problem is happened in deflection except reverberation time is in deflection analysis with wave that measure each in fixed distance in branch. Finally, when differ size of sound source and measure about change of sound pressure level in case measure sound pressure level giving difference about 10 dB, sound factor could know that there is no different effect.

A Study on Protection of Iris and fingerprint Data Based on Digital Watermarking in Mid-Frequency Band (중간 주파수 영역에서의 디지털 워터마킹 기법에 의한 홍채 및 지문 데이터 보호 연구)

  • Jeong, Dae-Sik;Park, Kang-Ryoung
    • Journal of Korea Multimedia Society
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    • v.8 no.9
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    • pp.1227-1238
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    • 2005
  • Recently, with the advance of network and internet technologies, it is appeared the Problem that the digital contents such as image, voice and video are illegally pirated and distributed. To protect the copyright of the digital contents, the digital watermarking technology of inserting the provider's information into the contents has been widely used. In this paper, we propose the method of applying the digital watermarking into biometric information such as fingerprint and iris in order to prevent the problem caused by steal and misuse. For that, we propose the method of inserting watermark in frequency domain, compare the recognition performance before and aster watermark inserting. Also, we experiment the robustness of proposed method against blurring attack, which is conventionally taken on biometrics data. Experimental results show that our proposed method can be used for protecting iris and fingerprint data, efficiently.

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Anti-Spam for VoIP based on Turing Test (튜링 테스트 기반으로 한 VoIP 스팸방지)

  • Kim, Myung-Won;Kwak, Hu-Keun;Chung, Kyu-Sik
    • Journal of KIISE:Computing Practices and Letters
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    • v.14 no.3
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    • pp.261-265
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    • 2008
  • As increasing the user of VoIP service using ITSP(Internet Telephony Service Provider), the VoIP spam becomes a big problem. The spam used in the existing public telephone is detected by using the pattern inspection of call behavior because it is difficult to filter contents for the characteristic of real-time voice communication. However there is a false-positive problem. The threat on spam remains where spam with low threshold can't be detected or users share one number. In this paper, we propose anti-spam for VoIP based on luring test. The proposed method gives a user luring test and he/she can connect to a receiver if passing turing test. A ticket is given to a user that pass luring test and it reduces overhead of luring test in re-dial. The proposed method is implemented on ASUS WL-500G wireless router and Asterisk IP-PBX. Experimental results show the effectiveness of the proposed method.

A Study on Kidney Diseases Diagnosis System for Sensation Type Using Physiological Signal Analysis (생체 신호 분석을 이용한 감각형 신장 질환 진단 시스템 연구)

  • Cho, Dong-Uk;Kim, Bong-Hyun;Lee, Se-Hwan
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.10C
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    • pp.964-972
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    • 2006
  • The kidney keeps with close relationship in the internal organs, that the kidney function filtering eliminate the wastes to the urine on the processing to replace the old with the new blood. In case of these problem in the kidney, there is no way to catch out with self-awakening symptom except for serious illness. This problem can solve with keeping the systematic diagnosis method in the kidney trouble shooting. Under the circumstances, the importance of the diagnosis for the kidney disease is growing day after day. In this paper, among the great four diagnosises, using the way of ocular inspection & auscultation, we would like to propose rouble shooting in the way of the kidney. To do this, through the assistance of the input image, extract the value of the color with appropriate output, analysing the color of the face with related to the kidney, using the results we would like to get the accurate symptoms on the kidney's problems. Also, through analysing and comparing with the relationship the kidney and the signal of voice, we would like to realize the proof system of human health. Finally, we'd like to make proof of the usefulness for proposed method from this study.

Transmission performance Improvement of Wireless ATM Cell in Fading Channel (페이딩 채널에서 무선 ATM 셀전송 성능개선에 관한 연구)

  • Chung, Boo-Young;Park, Ki-Sik;Kang, Young-Heung;Cho, Sung-Joon;Lee, Jin
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.9 no.2
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    • pp.179-190
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    • 1998
  • One of the most important topics of recent years is that a multimedia technology such as wireless ATM has been introduced in radio systems for video, voice and data transmission. A major problem in such systems is the occurrence of errors due to multipath fading or interference. To combat this problem, forward error correction (FEC) coding is commonly used as a means of eliminating these errors. This paper has analyzed theoretically the effect of fading to design effective wireless ATM system. The BER, CLR, and CER of wireless ATM system adopting FEC coding and diversity reception techniques have been derived in fading environment. Also, simulations of BER and CLR have been carried out and compared with the above theoretical results.

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