• Title/Summary/Keyword: audio application

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Augmented Multimedia E-commerce System using Person Wide Web (Person Wide Web 기술을 활용한 증강형 멀티미디어 상거래)

  • Han, Sang-Sook;Kim, Byung-Ho;Eun, Seong-Bae
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.15 no.1
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    • pp.81-88
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    • 2011
  • Augmented multimedia is a technology to provide additional informations to mobile devices when multimedia contents like video, audio and images are being played. Person Wide Web, PWW, is a scheme for acquiring a link and browsing a corresponding web pages on mobile devices, in which the link is attached any object and space in real world. In this paper we proposed an augmented multimedia E-commerce application system based on PWW scheme which can browse additional informations from video play on public spaces, and implemented on Microsoft Silverlight platform. We showed that the proposed system can support effectively the augmented multimedia E-commerce.

The Study of an Entrance Experience into Major Subject of a Student Nurse (간호학생의 전공교육 입문경험에 관한 연구)

  • Jo, Kae-Hwa;Cho, Young-Suk
    • The Journal of Korean Academic Society of Nursing Education
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    • v.7 no.2
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    • pp.360-368
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    • 2001
  • This research was an attempt to find out an entrance experience into major subject of student nurse. The purpose of this study was to understand an lived learning experience of student nurse. Method of the study was applied a phenomenological approach used by Colaizzi's analytic technique. Participants were 33 sophomore student nurse from TC University. The investigator conducted participated observations and in-depth unstructured interviews which were audio taped under the permission of student nurse, formulating meanings, themes and theme clusters. Results of the study were revealed 6 categories, 14 theme clusters, and 31 themes. Six categories include 'feeling anxious about clinical application', 'making superficial class', 'evaluation for ability of memory', 'deficit of academic evaluation criteria', 'insufficiency of textbook' and 'deficit for uniqueness of nursing science'. This findings confirm the fact that the student nurse's entrance experience was complicated and diversely influenced by their learning experiences. In conclusion, this study will prove helpful not only in understanding student nurse's entrance experience about their specialities but also in developing curriculum for nursing education.

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Recent R&D Trends for 3D Deep Learning (3D 딥러닝 기술 동향)

  • Lee, S.W.;Hwang, B.W.;Lim, S.J.;Yoon, S.U.;Kim, T.J.;Choi, J.S.;Park, C.J.
    • Electronics and Telecommunications Trends
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    • v.33 no.5
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    • pp.103-110
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    • 2018
  • Studies on artificial intelligence have been developed for the past couple of decades. After a few periods of prosperity and recession, a new machine learning method, so-called Deep Learning, has been introduced. This is the result of high-quality big- data, an increase in computing power, and the development of new algorithms. The main targets for deep learning are 1D audio and 2D images. The application domain is being extended from a discriminative model, such as classification/segmentation, to a generative model. Currently, deep learning is used for processing 3D data. However, unlike 2D, it is not easy to acquire 3D learning data. Although low-cost 3D data acquisition sensors have become more popular owing to advances in 3D vision technology, the generation/acquisition of 3D data remains a very difficult problem. Moreover, it is not easy to directly apply an existing network model, such as a convolution network, owing to the variety of 3D data representations. In this paper, we summarize the 3D deep learning technology that have started to be developed within the last 2 years.

Adaptive Buffer Management Method for QoS of Internet Telephony (인터넷폰의 QoS를 위한 적응적인 버퍼관리 방식)

  • 류태욱;이현관;이용구;김주웅;엄기환
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2002.05a
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    • pp.384-387
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    • 2002
  • Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality if the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to confirm the validity of the suggested algorithm, comparisons of the performance have been made between the existing buffer management algorithms and this new algorithm in various network settings.

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Design Consideration of Session Manager for N-to-N Multicast Services (다자간 멀티캐스트 서비스 지원을 위한 세션 관리자 고려사항)

  • Jeong, Ok-Jo;Park, Ju-Young;Kang, Shin-Gak
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • v.9 no.2
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    • pp.525-527
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    • 2005
  • the multicast services with multiple senders is interested according to the internet services such as multiple audio/video conference services. With the N-plex multicast protocols, multicast session manager is also needed for membership control and session control. RMCP-3 is designed for N:N group communications which is standardizing in ITU-T and supports multiple senders who join sessions. In this paper, we consider requirements of multicast manager for multiplex multicast communication protocol which many a joiner has a role in sender such conference application.

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An Adaptive Multimedia Synchronization Scheme for Media Stream Delivery in Multimedia Communication (멀티미디어 통신에서 미디어스트림 전송을 위한 적응형 멀티미디어 동기화 기법)

  • Lee, Gi-Sung
    • The KIPS Transactions:PartC
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    • v.9C no.6
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    • pp.953-960
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    • 2002
  • Rel-time application programs have constraints which need to be met between media-data. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control and palyout control. It is the important factor for playback rate and QoS if the buffer level is normal or not. This paper, The method of maintenance buffer normal state transmits in multimedia server by appling feedback of filtering function. And synchronization method is processing adaptive playout time for smooth presentation without cut-off while media frame is skip. When audio frame which is master media is in upper threshold buffer level we decrease play out time gradually, low threshold buffer level increase it slowly.

Implementation of 16Kpbs ADPCM by DSK50 (DSK50을 이용한 16kbps ADPCM 구현)

  • Cho, Yun-Seok;Han, Kyong-Ho
    • Proceedings of the KIEE Conference
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    • 1996.07b
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    • pp.1295-1297
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    • 1996
  • CCITT G.721, G.723 standard ADPCM algorithm is implemented by using TI's fixed point DSP start kit (DSK). ADPCM can be implemented on a various rates, such as 16K, 24K, 32K and 40K. The ADPCM is sample based compression technique and its complexity is not so high as the other speech compression techniques such as CELP, VSELP and GSM, etc. ADPCM is widely applicable to most of the low cost speech compression application and they are tapeless answering machine, simultaneous voice and fax modem, digital phone, etc. TMS320C50 DSP is a low cost fixed point DSP chip and C50 DSK system has an AIC (analog interface chip) which operates as a single chip A/D and D/A converter with 14 bit resolution, C50 DSP chip with on-chip memory of 10K and RS232C interface module. ADPCM C code is compiled by TI C50 C-compiler and implemented on the DSK on-chip memory. Speech signal input is converted into 14 bit linear PCM data and encoded into ADPCM data and the data is sent to PC through RS232C. The ADPCM data on PC is received by the DSK through RS232C and then decoded to generate the 14 bit linear PCM data and converted into the speech signal. The DSK system has audio in/out jack and we can input and out the speech signal.

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Design of the New Third-Order Cascaded Sigma-Delta Modulator for Switched-Capacitor Application (스위치형 커패시터를 적용한 새로운 형태의 3차 직렬 접속형 시그마-델타 변조기의 설계)

  • Ryu Jee-Youl;Noh Seok-Ho
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2006.05a
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    • pp.906-909
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    • 2006
  • This paper proposes a new body-effect compensated switch configuration for low voltage and low distortion switched-capacitor (SC) applications. The proposed circuit allows rail-to-rail switching operation for low voltage SC circuits and has better total harmonic distortion than the conventional bootstrapped circuit by 19 dB. A 2-1 cascaded sigma-delta modulator is provided for performing the high-resolution analog-to-digital conversion on audio codec in a communication transceiver. An experimental prototype for a single-stage folded-cascode operational amplifier (opamp) and a 2-1 cascaded sigma-delta modulator has been implemented in a 0.25 micron double-poly, triple-metal standard CMOS process with 2.7 V of supply voltage.

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Remote Firmware Monitoring System based on UPnP Network Protocol (UPnP 네트워크 기반의 원격 펌웨어 모니터링 시스템)

  • Kim, Sang-Yoon;Kim, Gwan-Hyung;Shin, Dong-Suk;Kang, Sung-In;Cho, Hyun-Cheol
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2010.05a
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    • pp.214-215
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    • 2010
  • UPnP Protocol is protocol for home networking that extends PnP to IP based network. It is able to remote control, transfer video and audio. However, It is only used simple devices. In this paper, UPnP is applied to Non IP Network and Extended its function. also, it Implements the System that is capable of updating firmware and realtime monitoring, and therefore it suggestions application possibility of UPnP Protocols that for efficiently management In environment of network on IP or Non-IP.

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Education equipment for FPGA-based multimedia player design (FPGA 기반의 멀티미디어 재생기 설계 교육용 장비)

  • Yu, Yun Seop
    • Journal of Practical Engineering Education
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    • v.6 no.2
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    • pp.91-97
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    • 2014
  • Education equipment for field programmable gate array (FPGA) based multimedia player design is introduced. Using the education equipment, an example of hardware design for color detection and augment reality (AR) game is described, and an example of syllabus for "Digital system design using FPGA" course is introduced. Using the education equipment, students can develop the ability to design some hardware, and to train the ability for the creative capstone design through conceptual, partial-level, and detail designs. By controlling audio codec, system-on-chip (SOC) design skills combining a NIOS II soft microprocessor and digital hardware in one FPGA chip are improved. The ability to apply wireless communication and LabView to FPGA-based digital design is also increased.