• International Journal of Contents
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• v.7 no.2
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• pp.11-16
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• 2011

In this paper, we develop a mobile MAP open API using HTML5 local storage and the W3C geolocation API. The mobile MAP open API consists of the basic JavaScript MAP API, offline navigation API, and multimedia POI (mPOI) API. The basic JavaScript MAP API creates a map and controls, rotates, and overlays data on the map. The offline navigation API is developed using HTML5 local storage and web storage. The mobile web application downloads and stores mPOIs of works of art to local storage or web storage from a web server. The mPOI API is developed using HTML5 video and audio APIs. We develop a mobile art park guidance application using the developed mobile MAP open API.

• MALSORI
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• no.43
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• pp.45-55
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• 2002

High reliability of automatic speaker recognition regardless of voice conditions is necessary for forensic application. Audio recordings in real cases are not consistent in voice conditions, such as duration, time interval of recording, given text or conversational speech, transmission channel, etc. In this study the variation of verification error rate of ASR system with the voice conditions was investigated. As a result in order to decrease both false rejection rate and false acception rate, the various voices should be used for training and the duration of train voices should be longer than the test voices.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.131-135
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• 2009

This paper is the introduction of our work on distributed load scheduling in single-level tree network. In this paper, we derive a new calculation model in single-level tree network and show a closed-form formulation of the time for computation system. There are so many examples of the application of this technology such as distributed database, biology computation on genus, grid computing, numerical computing, video and audio signal processing, etc.

• Journal of the KSME
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• v.33 no.2
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• pp.145-151
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• 1993

전자산업이 고도화되어감에 따라 전자기술은 소재 . 기계분야와 복합된 기술의 발전이 가속화 되어 가고 있다. 특히 이 글에서는 응력해석 CAE 기술이 전자부품산업에 적용되는 사례를 소 개하고자 한다. 그 중 하나는 하드디스크의 정밀부품 중의 하나인 스핀들 모터용 Hub의 절삭 가공시 고려하여야 하는 탄성 변형량을 해석하고 이를 적용하는 사례이고 또 하나는 Audio Deck의 Capatan Shaft를 지지하는 플리스틱 Base가 신뢰성 시험도중 또는 고온 부하시 발생하는 크리프현상을 해석하여 적용하는 사례이다. 이 글에서 소개하는 두 가지 사례는 전자부품산업에 응력해석 CAE를 적용할 수 있는 예들 중 빙산의 일각에 불과할 뿐이고, 이와 유사한 전자부품에 응력해석 CAE를 적용할 수 있다.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.1011-1012
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• 2006

In this paper, implementation process of standard platform for T-DMB Receiver in low-cost and small-size are following: First, implement SoC for 32 bit RISC CPU and 16 bit DSP, Hardware H.264 CODEC, Post Processor or Video Display, Audio Processor, I/O Device. Second, implement Real Time OS for flexible application. Third, propose simple architecture for interface with peripheral devices using one-chip processor.

• Proceedings of the Korea Information Processing Society Conference
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• 2009.04a
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• pp.71-74
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• 2009

The data-flow paradigm has been employed in various application areas. It is particularly useful where large data-streams must be processed, for example in video and audio processing, or for scientific visualization. A video-based 3D reconstruction system should process multiple synchronized video streams. The system exhibits many properties that can be targeted using a data-flow approach that is naturally divided into a sequence of processing tasks. In this paper we introduce our concept to apply the data-flow approach to a multi-video 3D reconstruction system.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.6 s.312
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• pp.110-116
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• 2006

In this paper, we implement a speech recognition system to support ubiquitous sensor network application services such as switch control, authentication, etc. using wireless audio sensors. The proposed system is consist of the wireless audio sensor, the speech recognition algorithm using psychoacoustic model and LDPC(low density parity check) for correcting errors. The proposed speech recognition system is inserted in a HOST PC to use the sensor energy effectively mil to improve the accuracy of speech recognition, a FEC(Forward Error Correction) system is used. Also, we optimized the simulation coefficient and test environment to effectively remove the wireless channel noises and correcting wireless channel errors. As a result, when the distance between sensor and the source of voice is less then 1.0m FAR and FRR are 0.126% and 7.5% respectively.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.12
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• pp.51-60
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• 2004

DAB(Digital Audio Broadcasting) is a new media service that can provide CD quality audio, various data service, interactive and high quality mobile communications through popular media such as terrestrial broadcasting, satellite, cable TV, and internet. In this paper, a new EPG(Electronic Program Guide) application model is proposed. The model is based on DAB and combines a DAB receiver and PCs so that it can take advantages of using various multimedia services and plenty of internet contents. The developed EPSD(EPG Provider System on DAB) has Web-based Server/Client structure and povides EPG functionalities to client PCs over internet. Therefore, the DAB receiver can be smaller and cheaper, and can develop abundant data services on internet. It can also provide high quality video services and be expected to become an important component in future home network systems.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.11
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• pp.141-148
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• 2012

A design of a 32-bit fourth-stage decimation filter decimation filter used in sigma-delta analog-to-digital (A/D) converter is proposed in this work. A four-stage decimation filter with down-sampling factor of 512 and 32-bit output is developed. A multi-stage cascaded integrator-comb (CIC) filter, which reduces the sampling rate by 64, is used in the first stage. Three half-band FIR filters are used after the CIC filter, each of which reduces the sampling rate by two. The pipeline structure is applied in the CIC filter to reduce the power consumption of the CIC. The Canonic Signed Digit (CSD) arithmetic is used to optimize the multiplier structure of the FIR filter. This filter is implemented based on a semi-custom design flow and a 130nm CMOS standard cell library. This decimation filter operates at 98.304 MHz and provides 32-bit output data at an audio frequency of 192 kHz with power consumption of $697{\mu}W$. In comparison to the previous work, this design shows a higher performance in resolution, operation frequency and decimation factor with lower power consumption and small logic utilization.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.44 no.4 s.316
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• pp.36-44
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• 2007

In this paper, we implement the speech & face recognition system to support various ubiquitous sensor network application services such as switch control, authentication, etc. using wireless audio and image interface. The proposed system is consist of the H/W with audio and image sensor and S/W such as speech recognition algorithm using psychoacoustic model, face recognition algorithm using PCA (Principal Components Analysis) and LDPC (Low Density Parity Check). The proposed speech and face recognition systems are inserted in a HOST PC to use the sensor energy effectively. And improve the accuracy of speech and face recognition, we implement a FEC (Forward Error Correction) system Also, we optimized the simulation coefficient and test environment to effectively remove the wireless channel noises and correcting wireless channel errors. As a result, when the distance that between audio sensor and the source of voice is less then 1.5m FAR and FRR are 0.126% and 7.5% respectively. The face recognition algorithm step is limited 2 times, GAR and FAR are 98.5% and 0.036%.