• 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.

• 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.

• Journal of Practical Engineering Education
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• v.7 no.1
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• pp.31-37
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• 2015

Terrestrial digital multimedia broadcasting (TDMB) is a method of bringing multimedia images, radio, internet, and television to portable devices through terrestrial digital radio transmissions. TDMB related educations being carried out in colleges are focusing on developing firmware which enables users to choose a wanted service. TDMB transmission frame is made up of synchronization channel (SC), fast information channel (FIC), and main service channel (MSC). Services such as video, audio and date are transmitted in the form of subchannel in the MSC. FIC carries information related to each services and subchannels. This paper presents a TDMB frame analysis simulator for analyzing and displaying FIC data on PC. TDMB frame analysis simulator contains functions such as controlling TDMB receiver through USB, establishing the frequency, bringing FIC to PC, displaying ensemble ID and levels, and displaying informations related to services and subchannels. In addition to that, this simulator has a function of being able to store FIC date and subchannel data. This simulator being developed with C++ is expected to be used to view those data visually so that it helps students to understand the TDMB system better and bring about the educational motivation.

• Smart Media Journal
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• v.7 no.3
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• pp.29-34
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• 2018

This paper presents a study on adapting optical communication technology using infrared ray for the enhancement of hearing aids in noisy environment. The transmitter module containing microphone and infrared ray-emitting diode converts audio signal to infrared optical signal and sends it out in the air. The receiver module located in a distance receives the infrared signal, converts it to electrical signal, and transfers it to an input of a digital hearing aid. Especially, the receiver module needs to be small, low voltage, and consume low power since it will be attached to hearing aids. Experiments with applying infrared communication technology of digital modulation method and analog non-modulation method show that the analog non-modulation method is adequate for infrared communication of approximately 5m distance indoor. Prototypes of transmitter module and receiver module were manufactured, and internal parameters of the digital hearing aid were adjusted to confirm normal transmit-receive operation of audio signals.

• The KIPS Transactions:PartB
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• v.12B no.6 s.102
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• pp.631-636
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• 2005

Recently, research started to focus on Digital Multimedia Broadcasting(DMB), while the DMB service is already being launched to provide high quality audio, video and data broadcasting in Korea. Therefore, the demand for multimedia contents is also rose very sharply. In this paper, we introduced an MPEG-4 contents authoring system with mobile device for Digital Multimedia Broadcasting. Our system allows both single-handed and collaborative authoring modo. Further more, to overcome the delaytime for decoding the BIFS information, we also introduced an efficient algorithm to generate dynamic and light BIFS information.

• Proceedings of the KIEE Conference
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• 2008.10b
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• pp.143-144
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• 2008

본 논문은 오디오 압축시 하모닉의 특성을 적용함으로써 신호의 압축률을 향상시킬 수 있도록 하는 내용을 기술하고 있다. 하모닉 코딩은 오디오 신호가 가지는 특징인 복합음(Complex tone)의 특성을 이용하는 것으로, 주파수 공간에서 정수배의 주파수가 존재하며, 정면파의 특성상 시간적으로 인접 신호들간의 유사성이 매우 높은 특징을 이용하여 압축효율을 향상시키는 방법이다. 하지만 실질적인 오디오 신호의 경우, 악기들의 harmonic stretch, 전달과정에서 발생하는 신호의 왜곡, 외부 잡음등의 특성으로 인하여 수집된 오디오 신호를 분석하는 과정에서 부정확한 하모닉의 판단이 이루어질 가능성이 높으며, 이는 압축과정에서 심각한 음질의 열화를 가져오게 된다. 따라서 본 논문에서는 프레림간의 변화 추이의 판단을 통하여 하모닉의 변화를 예측하고, 예측 오류에 대한 보상값을 전달함으로써 오디오 신호의 안정적인 압축/복원이 가능하도록 하는 신호처리 방법에 대한 내용을 기술하고있다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2013.06a
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• pp.342-343
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• 2013

본 논문에서는 유럽의 디지털 라디오 방송 규격인 DAB+(Digital Audio Broadcasting Plus)와 DRM+(Digital Radio Mondiale Plus) 시스템을 수신하기 위한 소프트웨어 기반의 수신 시스템의 설계 및 구현에 대한 연구를 수행하였다. 기존의 아날로그 FM 수신기를 대체할 수 있도록 소프트웨어 기반의 디지털 라디오 수신 시스템을 구현하였으며, 각 시스템의 USB 수신기로부터 방송을 입력받아 메모리 공유기법을 통해 일괄 수신 처리함으로써 시스템의 부하를 감소시킨다.

• Journal of Korea Multimedia Society
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• v.1 no.2
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• pp.131-141
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• 1998

Video data occurred in multimedia documents consist of a large scale of irregular data including audio-visual, spatial-temporal, and semantic information. In general, it is difficult to grasp the exact meaning of such a video information because video data apparently consist of unmeaningful symbols and numerics. In order to relieve these difficulties, it is necessary to develop an integrated manager for complex structures of video data and provide users of video digital libraries with easy, systematic access mechanisms to video informations. This paper proposes a generic integrated video information model(GIVIM) based on an extended Dublin Core metadata system to effectively store and retrieve video documents in digital libraries. The GIVIM is an integrated mo이 of a video metadata model(VMN) and a video architecture information model(VAIM). We also present design and implementation results of a video document management system(VDMS) based on the GIVIM.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.165-170
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• 1995

In this paper, it is attempted to apply the adaptive digital signal processing to compensate for a linear distortion of a loudspeaker and implement a real time hardware for that purpose. The real time system is implemented by using the DSP56001, a general purpose signal processor, as a host processor and the DSP56200, a cascadable adaptive FIR filter peripheral chip, as an adaptive digital filter. The system has 1000 taps at a 44.1kHz. After inverse modeling of under_compensation_speaker, the system reduces loudspeaker's linear distortions by pre-processing an input audio signal to loudspeaker. The experiment shows satisfactory results; after adaption with white noise as input signal for 60sec, the flat amplitude and linear phase frequency characteristics is found to lie over a wide frequency range of 100Hz to 20kHz.

• 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.