• Journal of KIISE:Computing Practices and Letters
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• v.6 no.6
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• pp.580-587
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• 2000

In this paper, we present a design method of a digital filter converting humming voice melody into MIDI data and a method of adapting it to a distributed multimedia I/O server. MuX uses device-independent DLMs(Dynamic Linking Module) for the inteliace with various I/O devices, and has wave-form audio DLM and MIDI DLM for audio interfaces. In order to expand the audio device interfacing ability of MuX system, we have designed and implemented a filter transforming human voice into MIDI messages. As the methods to input MIDI data are expanded to human voice in addition to MIDI files and MIDI instrument, someone who is not good at playing instruments can also generate the MIDI data, which enables our media interfaces to be used in various applications.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2009.11a
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• pp.285-285
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• 2009

Due to a development of a modem technology as Power Line Communication(PLC) over 200 Mbps, the high-speed multi-media data trasmission could be currently possible. In This paper we develop a high quality media transmitter-receiver based on merging the HomePlug AV, which is 200 Mbps class PLC technology and HDMI Interface technology. Smart Live 6 software were used for the assessment of audio property. As the result of measurement of the HD class images by capturing from the receiver of the PLC, the quality of images couldn't be confirm any deterioration, which has compared with original reflections. In case of audio part as the result of confirmation of the Phase, Magnitude, it has been confirmed that over 90% of nomal transmition and receiving of acoustic signal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.873-892
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• 1998

In this paper, we propose an architecture of multimedia communication terminal that can e used in broadband ISDN environments. To design and implement the multimedia communication terminal, we analyzed the structure of multimedia terminals and the types of services which is recommended by public and private standard or ganization, such as ITU-T, Digital Audio-Visual Council(davic) and ATM Forum. The multifunctional communication terminal designed in this paper could allow inter-working between existing communication terminals on the heterogeneousnetwork and accept current and advanced multimedia communication application flexible. An implemented terminal is consisted of the multimedia processing board and the ATm interface board that is installed in PCI bus on personal computer. The integrated service multimedia communication terminal that is implemented supports retrieval, distributive and conversational communication service simultaneously. And then we performaned functional module test according to the individual communication services.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1998.06b
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• pp.116-121
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• 1998

As digital audio video data compression and transmission techniques are matured, huge amount of digital multimedia material is produced and delivered via broadcasting, digital storage media and world-wide web(WWW). Thus it became very important to provide a standardized way of multimedia data content description, so that efficient and effective access and reuse of valuable multimedia information can be possible. In this paper, enabling core technologies and our research directions on this are presented with brief introduction on the scope of the multimedia content description interface, called MPEG-7, in terms of objective, application and requirements.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.11-20
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• 2005

In this paper, we deal with the research about a S/PDIF (Sony Philips Digital Interface) receiver which can operate without PLL (Phase Locked Loop) circuits. Although a S/PDIF receiver is used in most audio devices and audio processors in these days. yet there are only few domestic researches about S/PDIF. Currently used commercial DACs (Digital-to-Analog Converters) which can decode S/PDIF signals, have a PLL circuit inside them. The PLL makes it possible to extract clock information from S/PDIF digital signal and to synchronize a clock signal with input signals. But the PLL circuit makes many diffculties in designing the SOC (System On Chips) of VLSIs (Vew Large Scale Integrated Ciruits) because it is an "analog circuit". We proposed a S/PDIF receiver which doesn't have PLL circuits and only has Pure digital circuits. The key idea of the proposed S/PDIF receiver. is to use the ratio between a 16 MHz basic input clock and S/PDIF signals. After having decoded hundreds thousands S/PDIF inputs, it went to prove that a S/PDIF receiver can be designed with pure digital circuits and without any analog circuits such as PLL circuits. We have confidence that the proposed S/PDIF receiver can be used as an IP (Intellectual Property) for the SOC design of the digital circuits.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.562-566
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• 2004

In this paper, we propose a method for constructing terrestrial/cable PSIP converting system, so-called a PSIP converter, which is converting terrestrial PSIP into cable PSIP for a data broadcasting service in the interoperable network of terrestrial and cable, and define an interface between the PSIP converter and the OOB SI generator by using PMCP messages compliant to ATSC T3/S1. The exiting PSIP converter just converts terrestrial PSIP into cable PSIP compliant to ATSC and OCAP standard and transmits by a MPEG-2 TS format. That is to say, it is not for the digital data broadcasting but for the digital broadcasting. Say in other word, the proposed PSIP converter is designed to convert and transmit PSIP including DET information which is necessary data event information for data broadcasting service. In addition, the PSIP converter can support various types of PSIP information to the OOB SI generator by using PMCP messages defined by a hierarchical structure as per each channel, audio/video event, data event and so on.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.48 no.11
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• pp.27-33
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• 2011

Recently many display devices support the digital display interface as display market growth. DisplayPort is a next generation display interface at the PC, projector and high definition content applications in more widely used connection solution development. This paper implements multiple streams based on the behavior of the main link that is suitable for the display port v1.1a standard. The limit point of Displayport, interface between the Sink Device and Sink Device is also implemented. And two or more differential image data are enable to output the result through four Lanes stated in display port v1.1a, of two or more display devices without the addition of a separate Lane. The Multiple Video Streaming Controller is implemented with 6,222 ALUTs and 6,686 register, 999,424 of block memory bits synthesized using Quartus II at Altera Audio/Video Development board (Stratix II GX FPGA Chip).

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.354-359
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• 2008

This Audio-Visual Interactive Installation is composed of a video projection of a video Projection and digital Interface technology combining with the viewer's voice recognition. The Viewer can interact with the computer generated moving images growing on the screen by blowing his/her breathing or making sound. This symbiotic audio and visual installation environment allows the viewers to experience an illusionistic spacephysically as well as psychologically. The main programming technologies used to generate moving water waves which can interact with the viewer in this installation are visual C++ and DirectX SDK For making water waves, full-3D rendering technology and particle system were used.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.06a
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• pp.258-259
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• 2016

일반적으로 UHD 방송 편집 시스템에서 UHD 영상의 데이터양이 막대하기 때문에 실시간 전송을 위해 코덱과 함께 압축하여 편집 서버로 혹은 편집 서버로부터 스트림 형태로 전송한다. BT.1120 형태로 전송 송출된 SDI (Serial Digital Interface) 내장 음향 데이터는 영상과 달리 보조 데이터 영역에 다른 메타 데이터들과 함께 합성되어 전송 송출되기 때문에 추출 및 합성이 상대적으로 어렵다. 특히 재생을 위해서는 영상 코덱으로부터의 출력 영상과의 동기를 고려해야 하고 음향 데이터를 BT.1120 표준에 맞춰 보조 데이터 영역에 합성해야하기 때문에 개발에 어려움이 있다. 이에 본 논문은 UHD 영상/음향 데이터의 실시간 획득/재생 시스템에서의 SDI 내장 음향 데이터의 추출 및 합성을 위한 FPGA (Field Programmable Gate Array) 기반 하드웨어 플랫폼을 제안하였다. 또한, 이를 위한 음향 데이터 추출 로직과 합성 로직을 HDL(Hardware Design Language) 설계하여 FPGA 내에 탑재하고 카메라/디스플레이/편집 서버와 통합하였다. 시험 결과 4K 60fps 데이터에서 정상적으로 영상과 음향을 분리/획득 및 합성/재생하였다.

• 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.