• Title/Summary/Keyword: Digital Voice

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Coast Evaluation Techniques for Mode Selection in Video Coding (동영상에서 모드 선택을 위한 코스트 평가 방법)

  • Song, Dae-Geon
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.13 no.6
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    • pp.275-280
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    • 2013
  • Recently, access networking BroadBand the high performance of the video equipment to the Internet via voice, video, multimedia services, such as dealing with the media information dissemination is becoming increasingly attracting attention. More video devices and network environments in the future to keep pace with the high-quality video using the form dealing with an increasingly diversified and shall utilization is expected. Among them, video encoding technology, image compression encoding technology of information technology is one of the central role. Video coding technology that requires a vast amount of information contained in the video signal and the appropriate amount of information to eliminate redundancy as the efficiency of the digital code representing video signal is developed as a technology is going. Therefore, this study applied to video coding mode selection in the cost evaluation methods to examine and to maximize the coding efficiency and the proposed method compared to the conventional method was confirmed excellence.

Dynamic Channel Allocation and Channel Access Mechanism of Multimedia Traffic in the UTRA TDD Systems (UTRA TDD 시스템에서 멀티미디어 트래픽의 동적 채널 할당 및 채널 액세스 매커니즘)

  • 주용원;윤찬영;오영환
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.27 no.8C
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    • pp.819-827
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    • 2002
  • Personal mobile communication has been developed up to IMT-2000 which is called the third generation mobile communication. The first generation of personal mobile communication was analog cellular, the second was digital cellular, and the 2.5 generation was PCS. Before the third generation had been developed, the personal mobile communication service was focused mainly on the voice-oriented service. But, we can expect that multimedia service after the third generation system will be the core of the subject. T In this thesis, we propose a dynamic resource allocation algorithm in the UTRA TDD systems which can support the asymmetric traffic propensity and multimedia traffic. The proposed algorithm consists two sub-algorithms. One is the dynamic channel allocation method that determines the amount of bandwidth assigned between uplink and downlink according to resource allocation status list through asymmetric traffic propensity. The other is the channel access mechanism that assigns RU to bearer service generated in a cell. By simulation, the proposed dynamic resource allocation algorithm is proved to support asymmetric propensity of traffic and shows a better throughput for multimedia traffic.

The First Formant Characteristics in Vocalize of One Soprano (소프라노 1인의 모음곡 발성 시 제 1 포먼트의 변화양상)

  • Song, Yun-Kyung;Jin, Sung-Min
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.16 no.1
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    • pp.10-14
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    • 2005
  • Background and Objectives : Vowels are characterized on the basis of formant patterns. The first formant(F1) is determined by high-low placement of the tongue, and the second formant (F2) by front-back placement of the tongue. The fundamental frequency(F0) of a soprano often exceed the normal frequency of the first formant. And the vocal intensity is boosted when F0 is high and a harmonic coincides with a formant. This is called a formant tuning. Experienced singers thus learned how to tune their formants over a resonable range by lowering the tongue to maximize their vocal intensity. So, the current study aimed to identify the formant tuning in one experienced soprano by comparing the first formants of vowel [i] in three different voice production : speech, ascending scale, and vocalize. Materials and Method : All voices recordings of vowel [i] in speech, ascending scale (from F4 note to A4 note), and vocalize(:Ridente la calam") were made with digital audio tape-corder in a sound treated room. And the captured data were analyzed by the long term average(LTA) power spectrum using the FFT algorithm of the Computerized Speech Lab(CSL, Kay elementrics, Model, 4300B). Results : Although the first formant of vowel [i] in speech was 238Hz, those of ascending scale [i] were 377Hz, 405Hz, 453Hz respectively in F4(349z), G4(392Hz), A4(440Hz) note, and 722Hz, 820Hz, 918Hz respectively in F5 (698Hz), G5(784Hz), A5(880Hz) note. In vocalize, first formants of [i] were 380Hz, 398Hz, 453Hz respectively in F4, G4, A4 note, and 720Hz, 821Hz, 890Hz respectively in F5, G5, A5 note. Conclusion : These results showed that the first formant of ascending scale and vocalize sustained higher frequency than fundamental frequency in high pitch. This finding implicates that the formant tuning of vowel [i] in ascending scale was also noted in vocalize.

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Evolution of Next Generation Mobile Network Based on CDMA2000-1X Network (CDMA 2000-1X를 기반으로한 차세대 이동망의 진화)

  • Son, Dong Chul;Kim, J.W.;Ryu, C.S.
    • The Journal of the Korea institute of electronic communication sciences
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    • v.1 no.1
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    • pp.70-80
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    • 2006
  • The large portion of communication service areas move from a legacy wire-line voice service to mobile data service. For the purpose of satisfaction on market need, many communication systems should be installed and upgraded based on a mobile wide-band transmission facility. Recently, large part of communication service is based on internet protocol by packet switch techniques and required new technologies such as multimedia processing, QoS achievement, and mobility managememobile communication network such as IS-95A/B and CDMA2000-1X. In this paper, I analyzed the network architecture and service provision methods. in CDMA2000-1X nt. In korea, a CDMA communication technique is standardized for digital mobile communication systems. By using the analysed results, I will extract an efficient method for network evolution and a core technique for next generation mobile communication network.

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The Similarity of the Image Comparison System utilizing OpenCV (OpenCV를 활용한 이미지 유사성 비교 시스템)

  • Ban, Tae-Hak;Bang, Jin-Suk;Yuk, Jung-Soo
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2016.05a
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    • pp.834-835
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    • 2016
  • In recent years, advances in technology, IT is rapidly growing. Accordingly, real time image processing and multiple platforms, providing compatibility with OpenCV for image processing technology research on actively in progress. At present, different, comparing the images to determine the similarity is low, the system will match the rate of people using the analogue figures to determine the system is for the most part. In this paper, Template Matching of OpenCV and Feature Matching utilizing different images to determine the similarity between digital values for the system. A comparison of the features of a specific point on the screen the image to extract the same feature in a different size, you can compare the features of the target image recognized as compared to three historic castle in comparison, verification. This is the voice and image recognition and analysis, check the matching rate readings than in Zhengzhou treatment techniques are available. The future of forensic and other image processing technologies for OpenCV studies will be needed to feed.

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Design of User Access Authentication and Authorization System for VoIP Service (사용자 접근권한 인증을 이용한 안전한 VoIP 시스템 설계)

  • Yang, Ho-Kyung;Kim, Jin-Mook;Ryou, Hwang-Bin;Park, Choon-Sik
    • Convergence Security Journal
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    • v.8 no.4
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    • pp.41-49
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    • 2008
  • VoIP is a service that changes the analogue audio signal into a digital signal and then transfers the audio information to the users after configuring it as a packet; and it has an advantage of lower price than the existing voice call service and better extensibility. However, VoIP service has a system structure that, compared to the existing PSTN (Public Switched Telephone Network), has poor call quality and is vulnerable in the security aspect. To make up these problems, TLS service was introduced to enhance the security. In practical system, however, since QoS problem occurs, it is necessary to develop the VoIP security system that can satisfy QoS at the same time in the security aspect. In this paper, a user authentication VoIP system that can provide a service according to the security and the user through providing a differential service according to the approach of the users by adding AA server at the step of configuring the existing VoIP session is suggested. It was found that the proposed system of this study provides a quicker QoS than the TLS-added system at a similar level of security. Also, it is able to provide a variety of additional services by the different users.

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A Design of Turbo Decoder using MAP Algorithm (MAP 알고리즘을 이용한 터보 복호화기 설계)

  • 권순녀;이윤현
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.7 no.8
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    • pp.1854-1863
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    • 2003
  • In the recent digital communication systems, the performance of Turbo Code using the mr correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the huh decoder. However, performance depends on the interleaver depth that needs many delays over the reception process. Moreover, turbo code has been blown as the robust coding methods with the confidence over the fading channel. International Telecommunication Union(ITU) has recently adopted it as the standardization of the channel coding over the third generation mobile communications(IMT­2000). Therefore, in this paper, we preposed the interleaver that has the better performance than existing block interleaver, and modified turbo decoder that has the parallel concatenated structure using MAP algorithm. In the real­time voice and video service over third generation mobile communications, the performance of the proposed two methods was analyzed and compared with the existing methods by computer simulation in terms of reduced decoding delay using the variable decoding method over AWGN and fading channels for CDMA environments.

A Study on the Removal of Impulse Noiseusing Wavelet Transform Pair and Adaptive-Length Median filter (웨이브렛 변환쌍과 적응-길이 메디안 필터를 이용한 임펄스 노이즈 제거에 관한 연구)

  • 배상범;김남호
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.7 no.7
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    • pp.1575-1581
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    • 2003
  • As a society has progressed rapidly toward a highly advanced digital information age, a multimedia communication service for acquisition, transmission and storage of image data as well as voice has being commercialized externally and internally. However, in the process of digitalization or transmission of data, noise is generated by several causes, and researches for eliminating those noises have been continued until now. There were the existing FFT(fast fourier transform) and STFT(short time fourier transform) for removing noise but it's impossible to know information about time and time-frequency localization capabilities has conflictive relationship. Therefore, for overcoming these limits, wavelet transform which is presented as a new technique of signal processing field is being applied in many fields recently. Because it has time-frequency localization capabilities it's Possible for multiresolution analysis as well as easy to analyze various signal. And when two wavelet base were designed to form Hilbert transform pair, wavelet pair provide superior performance than the existing DWT(discrete wavelet transform) in data characteristic detection. Therefore in this parer, we removed impulse noise by using adaptive-length median filter and two dyadic wavelet base which is designed by truncated coefficient vector.

An Enhanced DAP-NAD Scheme for Multi-Hop Transmission in Combat Net Radio Networks (전투 무선망에서 다중 홉 전송을 위한 향상된 DAP-NAD 기법)

  • Jung, Jong-Kwan;Kim, Jong-Yon;Roh, Byeong-Hee
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.37C no.10
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    • pp.977-985
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    • 2012
  • Recently, many countries have been developing new protocols to improve the performance of tactical ad hoc networks for implementing NCW (Network Centric Warfare). Combat net radio (CNR) networks are the most important communication infra for the ground forces such as infantry of Army. U.S. Army had developed MIL-STD-188-220D that is the Interoperability Standard for DMTDs (Digital Messages Transfer Device Subsystems) for voice and data communication in CNR. MIL-STD-188-220D is a candidate for MAC protocol of TMMR which is next radio and has a few constraints to used in TMMR. NAD (Network Access Delay) defined in MIL-STD-188-220D needs time synchronization to avoid collision. However, it is difficult for time synchronization to fit in multi-hop environment. We suggest the enhanced DAP (Deterministic Adaptable Priority)-NAD to prevent conflicts and decrease delays in multi-hop CNR. Simulation results show that the proposed scheme improves the performance in multi-hop CNR networks.

Efficiency Analysis of Schisandra Tea Using Image & Acoustic Signal Processing (영상과 음성 처리를 이용한 오미자차의 효능 분석)

  • Kim, Bong-Hyun;Cho, Dong-Uk;Han, Kil-Sung;J.Bae, Young-Lae
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.12 no.7
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    • pp.2975-2981
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    • 2011
  • We propose an efficiency analysis method to test whether Schisandra tea is beneficial to the health of internal organs, and, if so, to find which organ is mostly benefited. Firstly, for this, the color change in the right cheek region of face images before and after drinking Schisandra tea is analyzed using image processing techniques. Also, acoustic analysis experiments on both of the vocal cords vibration and the intensity value of acoustic signal are performed to investigate further Schisandra tea's effectiveness. The results of lung function smoothly by eating omijacha become. Vocal cords vibration and voice energy intensity of the right cheek region has changed a steady increase of the b values were observed.