• Journal of the Institute of Electronics Engineers of Korea SC
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• v.43 no.2 s.308
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• pp.28-32
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• 2006

In this paper, NTSC TV Transmitter Modulo will be designed and produced which make possible playing the motion picture not only on TV but also on portable TV. NTSC TV Transmitter Module modulates signals that received Video and Audio signals from a mobile on NTSC TV CH4 mechanism. so it has an advantage of convenience that watching the motion picture of mobile on TV without any other cable through transmitting signals by wireless. But it has some demerits of long size antenna and noise sensitiveness. In the future, if some problems like a size of antenna distortion of signal and noise can be solved through continuous researching about Radio Frequency part, it is possible to play mobile motion pictures on the more media like a camcorder, DVD player and so on.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.293-303
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• 2002

This paper implements optimized Time Scale Modification (TSM) system using ITU G.729 speech decoder and Dual SOLA algorithm. The proposed system assume 8 Kz sampling rate, 80 samples/frame input speech from the ITU G.729 speech Decoder and the TSM (Time Scale Modification) feature of Dual SOLA produces the high quality output speech that was slow-down or speed up as a user's choice. Especially, the proposed Optimized Dual SOLA base on various simulations and theoretical analysis, and the additional interpolation procedure of the speech makes it possible to setup high performance integrated TSM system at the maximum time scale modification rate. The system performance is analyzed and verified with various input speech and playback speed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.7A
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• pp.951-958
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• 2000

In wavelength-routed all-optical networks, signals are transmitted on a direct optical path, or a lightpath, in a single-hop manner without opto-electronic/electro-optic(OE/EO) conversion at intermediate nodes. However, due to the physical constraints of optical elements such as ASE noise and crosstalk signals can be degraded un a long path. To establish a connection under such impairments, the optical signal may need to be regenerated at intermediate nodes, dividing a lightpath into two or more fragments. However, since signal regeneration at intermediate nodes requires additional network resources, the selection of these nodes should be made carefully to minimize blocking of other lightpaths. In this paper, we deal with the problem of establishing a lightpath in a multihop manner under physical constraints. We provide both minimal-cost and heuristic algorithms for locating signal regeneration nodes(SRNs). For a minimal-cost algorithm, we formulate the problem using dynamic programming(DP) such that blocking of other lightpaths due to the lack of transmitters/receivers(TXs/RXs) and wavelengths is minimized throughout the network.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.9
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• pp.2164-2170
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• 2010

Digital Radio Frequency Memory, ( as DRFM ), is a device with the ability to restore output to the input RF signal in the required time after storing the incoming RF signals. Therefore DRFM is widely used in Jammer, EW Simulator, Target Echo Generator, and so on. This paper proposes its hardware implementation composed with the high frequency part and the digital processing part consisting of RF input/output module and local oscillator module. It is also proposed the replicated signal generation method which is consisted of the Analog-Digital conversion in the form of pulsed RF signal quantization, and FPGA to save and produce the playback signal, and RF signals to produce a Digital-Analog Conversion in the digital processing unit. This proposed scheme applied to test board and confirmed the validity of the proposed scheme through the test results obtained by the simulated input signals.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.1
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• pp.70-75
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• 2007

In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio with 44.1 kHz sampling rate, current existing digital audio is always restricted by sampling rate and bandwidth. This kind of restriction normally can be resolved by using low bit rate audio codec such as MP3, OGG, and AAC. However it suffers a major problem such as a loss of high frequency fidelity. This high frequency loss will reproduce only the band-limited low-frequency part of audio in the standard CD-quality audio. In general, the high frequency contents of audio have lots of information such as localization and ambient information, and bright nature of audio. The purpose of this paper is to implement on ARM platform system that can effectively estimate and compensate the missing high frequency contents of MP3 audio. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed algorithms for MP3 audio quality enhancement.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.599-609
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• 2009

An overview of the recently finalized multi-channel audio coding standard MPEG Surround is provided. This audio coding standard downmixes multi-channel signals to mono or stereo signals and, simultaneously, extracts spatial parameters for its encoding process. In its decoding process, it reconstructs multi-channel signals based on the downmix signals and spatial parameters. Since the downmix signals are coded in conventional audio coding format such as AAC and MP3 and the spatial parameters require a small amount of information MPEG Surround guarantees high sound quality multi-channel audio at low bit rates. Besides, it is backward-compatible to conventional audio coding techniques because the downmix signals can be played on portable audio devices ignoring the spatial parameter information. In this paper, Part 1 presents an overview of the basic structure of MPEG Surround and Part 2 describes various modes and tools including the binaural mode which supports the virtual 5.1-channel playback via headphones or earphones. The listening test results by various companies and organizations are also presented.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.11A
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• pp.1827-1840
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• 2001

In digital data transmission/storage systems, the compensation for channel distortion is conducted normally using a training sequence that is known a priori to both the sender and receiver. The use of the training sequences results in inefficient utilization of channel bandwidth. Sometimes, it is also impossible to send training sequences such as in the burst-mode communication. As such, a great deal of attention has been given to the approach requiring no training sequences, which has been called the blind equalization technique. On the other hand, to utilize the limited bandwidth effectively, the concept of partial response (PR) signaling has widely been adopted in both the high-speed transmission and high-density recording/playback systems such as digital microwave, digital subscriber loops, hard disk drives, digital VCRs and digital versatile recordable disks and so on. This paper is concerned with blind adaptive equalization of partial response channels whose transfer function zeros are located on the unit circle, thereby causing some problems in performance. Specifically we study how the problems of blind channel equalization associated with the PR channels can be improved. In doing so, we first discuss the existing methods and then propose new structures for blind PR channel equalization. Our structures have been extensively tested by computer simulation and found out to be encouraging in performance. The results seem very promising as well in terms of the implementation complexity compared to the previous approach reported in literature.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.775-778
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• 1998

근래에 들어 컴퓨터 기술은 멀티미디어 기수의 발달과 더불어 그에 따른 데이터량의 증가로 인해 데이터를 처리, 전송, 저장하는 모든 부문에서의 고속, 대용량화를 요구하고 있다. 이 중에서 특히 저장장치 부문은 응용 프로그램이 대형화되고 멀티미디어화에 따른 데이터량이 크게 증가하는 추세에 있기 때문에 지속적인 용량 증가가 요구되고 있다. 이런 상황에서 주목을 받고 있는 것이 신호처리 방식을 개선하여 저장장치의 기록 밀도를 향상시키는 기술의 하나인 partial response maximum likelihood (PRML) 기술이다. PRML 방식은 HDD 나 광 디스크로부터 데이터를 읽어낼때의 신호처리 기술 중의 ㅎ나로 신호간 간섭을 허용하여 데이터 속도를 증가시키고, 신호를 재생할 때 신호간 간섭을 보상하여 원래 신호를 복원해 내는 기술이다. 이를 이용하면 기존의 기록방식에 비해 기록밀도를 20-50% 정도 높일 수 있다.〔1〕〔2〕

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.4
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• pp.633-642
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• 2021

Growing demand for electricity, when combined with the need to limit carbon emissions, drives a huge increase in renewable energy industry. In the electric power system, electricity supply always needs to be balanced with electricity demand and network losses to maintain safe, dependable, and stable system operation. There are three broad challenges when it comes to a power system with a high penetration of renewable energy: transient stability, small signal stability, and frequency stability. Transient stability analyze the system response to disturbances such as the loss of generation, line-switching operations, faults, and sudden load changes in the first several seconds following the disturbance. Small signal stability refers to the system's ability to maintain synchronization between generators and steady voltages when it is subjected to small perturbations such as incremental changes in system load. Frequency stability refers to the ability of a power system to maintain steady frequency following a severe system upset resulting in significant imbalance between generation and load. In this paper, we discusses these stability using system simulation by renewable energy deployment plan, and also analyses the influence of the renewable energy sources to the grid stability.

• Electronics and Telecommunications Trends
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• v.10 no.2 s.36
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• pp.191-201
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• 1995

ISO/MPEG 오디오 그룹은 2채널 스테레오 오디오에 대한 압축 부호화 알고리즘을 ISO/MPEG-1 오디오 표준으로 정하고, 국제 표준 (ISO 11172-3)으로 제정하였다. ISO/MPEG-1 오디오의 주요 목적중 하나는 한정된 용량의 저장매체나 제한된 전송채널의 조건하에서 고품질의 오디오를 저장하거나 전송할 수 있는 저비트율의 오디오 압축 부호화 알고리즘의 표준을 개발하는 것이다. 이를 위해 ISO/MPEG 오디오에서는 심리음향 특성을 모델링한 지각 부호화 방식을 사용하여 원신호와 재생신호 사이에 정보의 객관 측정량의 차이는 있으나 주관 측정량의 차이가 없도록 한다. 이 글에서는 ISO/MPEG-1 오디오 부호화의 계층 I과 계층 II를 중심으로 하여 표준의 개요, 청각 특성을 이용한 심리 음향 모델, 부호화 알고리즘, 데이타 구조, 그리고 응용분야 등에 대해 기술하였다.