• 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.

• 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.

• 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.