• Proceedings of the KIEE Conference
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• 1999.07g
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• pp.3212-3214
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• 1999

RS(Reed-Solomon)부호는 오류 정정을 위한 채널 코딩기법중의 하나로 특히 연집 오류에 대해 강한 특성을 갖고 있으며, CD-P(Compact Disc Player), DAT(Digital Audio Tape). VTR, DVD(Digital Video Disc), 디지탈 TV 디코더등에서 사용되고 있다. 본 논문은 Galois Field, GF[$2^8$]상에서 (204. 188. 8)의 규격을 갖는 디지탈 TV용 RS 복호기의 구현에 관한 연구로 8개의 심볼 오류까지 정정 가능하다. 오증 계산은 16개의 오증 계산셀로 구성되어 지며, 오류 위치 다항식을 계산하는데 있어서는 Berlekamp-Massey 알고리즘을 사용한다. VHDL로 설계되어 Synopsys를 이용하여 검증 및 합성하였다.

• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.451-454
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• 2011

COSMOS (COntentS Making Omnipotent System)는 컴퓨터 게임이나 3차원 애니메이션 제작이 가능하도록 그래픽 랜더링, 특수효과, 물리엔진, 인공지능 엔진 등의 기능을 갖춘 범용성 3차원 콘텐츠 저작 시스템이며, 무엇보다도 직관적인 인터페이스 기능을 통해 사용자의 편리성을 제공해 준다. 본 논문은 COSMOS에서 음향 정보를 자동으로 3D 콘텐츠 구성 요소에 배합될 수 있도록 하는 기술에 대한 내용이다. 본 기술의 도입을 통해 COSMOS에서는 사용자의 의성어 소리를 인식하여, 그 의미에 적합한 디지털 사운드를 검색한 후에 사용자의 의도에 맞추어 변환하여 이와 관련된 콘텐츠 구성 요소와 일치 시켜줌으로써 보다 직관적으로 콘텐츠 저작 기능을 제공할 수 있다.

• Phonetics and Speech Sciences
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• v.15 no.2
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• pp.13-20
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• 2023

This study explores the potential of automated speech recognition (ASR) in assessing English learners' pronunciation. We employed ASR technology, acknowledged for its impartiality and consistent results, to analyze speech audio files, including synthesized speech, both native-like English and Korean-accented English, and speech recordings from a native English speaker. Through this analysis, we establish baseline values for the word error rate (WER). These were then compared with those obtained for human raters in perception experiments that assessed the speech productions of 30 first-year college students before and after taking a pronunciation course. Our sub-group analyses revealed positive training effects for Whisper, an ASR tool, and human raters, and identified distinct human rater strategies in different assessment aspects, such as proficiency, intelligibility, accuracy, and comprehensibility, that were not observed in ASR. Despite such challenges as recognizing accented speech traits, our findings suggest that digital tools such as ASR can streamline the pronunciation assessment process. With ongoing advancements in ASR technology, its potential as not only an assessment aid but also a self-directed learning tool for pronunciation feedback merits further exploration.

• Journal of Broadcast Engineering
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• v.11 no.3 s.32
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• pp.363-378
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• 2006

The Digital Multimedia Broadcasting(DMB) system is developed to offer high quality audio-visual multimedia contents to the mobile environment. The system adopts MPEG-4 standard for the main video, audio and other media format. It also adopts the MPEG-4 scene description for interactive multimedia contents. The animated and interactive contents can be actualized by BIFS(Binary Format for Scene), the binary format for scene description that refers to the spatio-temporal specifications and behaviors of the individual objects. As more interactive contents are, the scene description is also needed more high bitrate. However, the bandwidth for allocating meta data such as scene description is restrictive in mobile environment. On one hand, the DMB terminal starts demultiplexing content and decodes individual media by its own decoder. After decoding each media, rendering module presents each media stream according to the scene description. Thus the BIFS stream corresponding to the scene description should be decoded and parsed in advance of presenting media data. With these reason, the transmission delay of BIFS stream causes the delay of whole audio-visual scene presentation although the audio or video streams are encoded in very low bitrate. This paper presents the effective optimization technique for adapting BIFS stream into expected MPEG-2 TS bitrate without any bandwidth waste and avoiding the transmission delay of the initial scene description for interactive DMB contents.

• The Journal of the Korea Contents Association
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• v.11 no.12
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• pp.28-39
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• 2011

While the broadband multimedia technologies have been developing, the commercial market of digital contents has also been widely spreading. Most of all, digital cartoon market like internet cartoon has been rapidly large so video cartooning continuously has been researched because of lack and variety of cartoon. Until now, video cartooning system has been focused in non-photorealistic rendering and word balloon. But the meaningful frame extraction must take priority for cartooning system when applying in service. In this paper, we propose new automatic frame extraction method for video cartooning system. At frist, we separate video and audio from movie and extract features parameter like MFCC and ZCR from audio data. Audio signal is classified to speech, music and speech+music comparing with already trained audio data using GMM distributor. So we can set speech area. In the video case, we extract frame using general scene change detection method like histogram method and extract meaningful frames in the cartoon using face detection among the already extracted frames. After that, first of all existent face within speech area image transition frame extract automatically. Suitable frame about movie cartooning automatically extract that extraction image transition frame at continuable period of time domain.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.2
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• pp.34-45
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• 2007

Digital decimation filter is inevitable in oversampled sigma-delta A/D converters for the sake of reducing the oversampled rate to Nyquist rate. This paper presented a Verilog-HDL design and implementation of an area-efficient digital decimation filter that provides time-to-market advantage for sigma-delta analog-to-digital converters. The digital decimation filter consists of CIC(cascaded integrator-comb) filter and two cascaded half-band FIR filters. A CSD(canonical signed digit) representation of filter coefficients is used to minimize area and reduce in hardware complexity of multiplication arithmetic. Coefficient multiplications are implemented by using shifters and adders. This three-stage decimation filter is fabricated in $0.25-{\mu}m$ CMOS technology and incorporates $1.36mm^2$ of active area, shows 4.4 mW power consumption at clock rate of 2.8224 MHz. Measured results show that this digital decimation filter is suitable for digital audio decimation filters.

• Journal of Navigation and Port Research
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• v.28 no.3
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• pp.213-219
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• 2004

In this paper, a LNA(Low Noise Amplifier) has been developed, which is operating at L-band i.e., 1452∼1492 MHz for satellite DAB(Digital Audio Brcadcasting) receiver. The LNA is designed to improve input and output reflection coefficient and VSWR(Voltage Standing Wave Ratio) by balanced amplifier. The LNA consists of low noise amplification stage and gain amplification stage, which make a using of GaAs FET ATF-10136 and VNA-25 respectively, and is fabricated by hybrid method. To supply most suitable voltage and current, active bias circuit is designed Active biasing offers the advantage that variations in $V_P$ and $I_{DSS}$ will not necessitate a change in either the source or drain resistor value for a given bias condition. The active bias network automatically sets $V_{gs}$ for the desired drain voltage and drain current. The LNA is fabricated on FR-4 substrate with RF circuit and bias circuit, and integrated in aluminum housing. As a reults, the characteristics of the LNA implemented more than 32 dB in gain. 0.2 dB in gain flatness. lower than 0.95 dB in noise figure, 1.28 and 1.43 each input and output VSWR, and -13 dBm in $P_{1dB}$.

• ETRI Journal
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• v.28 no.5
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• pp.583-592
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• 2006

Digital broadcasting technology has developed focusing on multi-channel/multi-media, high-definition quality, and mobility-support. Recently, there has been a clear trend toward bidirectional service with the convergence between broadcasting and communication. The broadcasting viewer is no longer simply a passive receptor but has also become an information generator. Currently, the digital multimedia broadcasting (DMB) specifications are the major standard for portable digital broadcasting and have been establishing the overall guidelines for bidirectional service using the MPEG-4 system. While detailed specifications for DMB systems are not well-established for bidirectional service yet, they share the basic concepts underlying the European Eureka-147 Digital Audio Broadcasting (DAB) system. This paper develops key scenarios for bidirectional service in DMB, describes the signal transaction of broadcasting and return channels, and demonstrates typical scenarios using binary format for scenes (BIFS) in the MPEG-4 system.

• Electronics and Telecommunications Trends
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• v.10 no.3 s.37
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• pp.29-45
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• 1995

대화형 멀티미디어 서비스중에서 대표적인 VOD(Video on demand) 서비스 일반에 관한 표준제정이 DAVIC(Digital Audio-Visual Council)을 중심으로 한창 진행되고 있다. 본 고에서는 DAVIC에서 정하고 있는 시스템 기준모델을 각 구성요소별로 살펴보고 디지털 음성-영상 응용 및 서비스들간의 상호 운용성을 위해 현재 표준제정 중인 인터페이스와 프로토콜을 1995년 1월에 만들어진 초안과 3월의 그 수정판 (DAVIC 1.0)을 중심으로 상세히 소개하려고 한다.