• Journal of Information Management
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• v.35 no.2
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• pp.41-55
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• 2004

This study is expected to discuss some methods regarding the uplift of user's use convenience and the efficiency of material management by constructing the multimedia database through the digitalization of the especial audiotape-typed materials among accompanying materials. In order to do that, this paper dealt with the present management conditions of accompanying materials, the sorts of audio data formats, data format transformation, the methods of administration and utilization, etc. And this paper also presented the expected effect and problems caused by multimedia databases construction of accompanying materials.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10b
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• pp.493-495
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• 2003

최근의 휴대 전화는 단순한 음성 통신 기기 역할 뿐만 아니라 데이터 통신 기기로도 쓰이고 있으며, CDMA-2000 망 보급으로 인하여 데이터 통신 대역폭이 멀티미디어 데이터를 처리 할 수 있을 만큼 증대 되었다. 하지만 휴대 전화는 하드웨어 성능이 음성 통신 기기로 최적화 되어 있고 휴대성을 높이기 위해 저전력의 저 성능 프로세서를 탑재 하였기 때문에 소프트웨어 방식의 비디오/오디오 재생이 매우 힘들다. 특히. 널리 사용되는 비디오/오디오 표준(MPEG-x, H.26x, 등등)은 압축 최우선의 방식으로써 계산량이 매우 크기 때문에 휴대 전화에서 하드웨어 도움 없이 소프트웨어로만 재생하기에는 적합하지 않다. 본 논문에서는 이러한 문제를 해결하기 위해 먼저 일반 목적의 널리 사용되는 코덱의 문제점과 휴대전화의 하드웨어 자원에 관해 알아 보고, 연산량을 효과적으로 조절할 수 있는 웨이블릿 함수를 이용하여 휴대 전화 시스템에 적합한 비디오/오디오 코덱을 제안한다. 또한 비디오 디코딩에 필요한 연산을 측정하고 실제 휴대 전화에 적용하여 그 성능을 확인 한다.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.437-445
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• 2004

In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.6
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• pp.180-187
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• 2013

In this paper, we propose a LED communication based multi-hop audio data transmission network system. The main contribution and features of the proposed system are as follows. First, the contribution of this research is to develope the LED communication based multi-hop transmission network system which can transmit audio data signal with long distance via multi-hops. Second, the developed system has the following features: In transmitter, audio data is transmitted after encoding with S/PDIF format via a general LED. The relay receives digital audio signal by using photo diode and then transmits the signal to receiver after error checking and amplifying. The receiver receives the encoded audio data via photo diode and then converts to analog audio signal by using decoding and amplifying. The performance evaluation of the proposed system is conducted in the laboratory with fluorescent light source. The results of the performance evaluation confirm that the system can provide high quality audio transmission from transmiter to receiver via multi-hop relays in a long distance while we can see there are differences in the transmitted audio quality according to the used LED colors.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.19-22
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• 1997

음성이나 동화상 같은 실시간 정보를 인터넷상으로 전송하는 인터넷방송 시스템은 인터넷 서비스의 중요한 이슈중에 하나지만 구현시에 가장 커다란 문제가 되는 것은 패킷 손실이다. 인터넷라디오나 AOD에서 발생하는 손실된 패킷을 복구하기 위해 여러 가지 방법이 제안되고 있으나, 현재 인터넷폰 등을 위해서는 잉여 오디오 정보를 이용한 패킷 복구 알고리즘이 가장 효과적으로 사용되고 있으나 단방향 통신이면서 음질이 가장 중요한 요소인 방송시스템에서는 그대로 적용하기에 문제가 있다. 본 논문에서는 잉여 오디오 정보를 이용한 패킷 복구 알고리즘에 사용될 수 있는 인터리빙 방법을 제안하였다. 연구결과, 인터리빙 처리를 위하여 추가적인 지연이 발생하는 단점은 있지만 연집 에러를 효과적으로 분산시켜 패킷 손실 복구율을 크게 개선시킬 수 있다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2021.10a
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• pp.284-286
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• 2021

With the continuous development of the Internet and digital, platforms are emerging to store large amounts of media data and provide customized services to individuals through online. Companies that provide these services recommend movies that suit their personal tastes to promote media consumption. Each company is doing a lot of research on various algorithms to recommend media that users prefer. Movies are divided into genres such as action, melodrama, horror, and drama, and the film's audio (music, sound effect, voice) is an important production element that makes up the film. In this research, based on movie trailers, we extract audio for each genre, check the commonalities of audio for each genre, distinguish movie genres through supervised learning of artificial intelligence, and propose a utilization method for generating metadata in the future.

• Journal of Global Scholars of Marketing Science
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• v.16 no.1
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• pp.93-116
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• 2006

The main purpose of this study is to examine whether power-of-speech have an impact on persuasion in radio CM and to determine what is stronger variable either power-of-speech or sex of source. Previous research showed confused result about this issue. Some research supposed the effect of power-of-speech depend on sex of source/Carli, 1990) but the others supposed it is not(Holtgraves & Lasky, 1999; Erickson, Lind, Johnson, & O'Barr, 1978). So present research is designed to show empirical evidence about this issue examining the interaction between effect of power-of-speech and sex of source. Result revealed that power-of-speech affects strongly on persuasion and that effect have no interaction with sex of source. This result means that the effect of power-of-speech is stronger than the effect of sex of source. But in attitude toward product exceptionally, female source using powerful speech induce more favorable attitude toward product than female source using powerless speech whereas male source using powerful and powerless speech induce same attitude toward product.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.04a
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• pp.453-456
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• 2001

21세기에 새로운 분야로 대두되고 있는 가상현실은 많은 사람들의 흥미를 끌고 있다. 상상 속에서나 가능하던 일들을 현실감과 입체감을 통해 실제처럼 느낄 수 있게 해준다는 것이 가상현실의 가장 큰 매력일 것이다. 가상현실을 요하는 멀티미디어 기기들의 활발한 시장진출로 3D 효과를 가진 오디오/비디오의 하드웨어 구현이 불가피하다. 본 연구에서는 휴대용 기기들에서 실시간 3D 입체음향 효과를 얻을 수 있도록 하드웨어를 구성하였다. 기존의 입체음향 기술에서 사용되는 콘볼루션 방법은 계산량이 많기 때문에 실시간 구현이 어렵다. 그러나 제안된 방식은 FFT를 사용하여 주파수 영역에서 처리함으로써 계산량을 줄여 하나의 프로세서로도 실시간 처리가 가능하도록 하였다. 저가/저전력/소형화조건을 요구하는 휴대용 기기에서 3D 입체 음향 효과를 얻을 수 있는 것이다. DSP는 TI(Texas Instruments)사의 16비트 고정소수점(fixed-point) 프로세서인 TMS320C5416을 사용한다. 구현된 3D 입체음향 칩은 입체음향을 필요로 하는 휴대용 MP3 Player, 가전용 오디오/비디오 등에 응용될 수 있다.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2E
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• pp.91-97
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• 1995

This paper describes the implementation of a multi-channel audio codec for HETV. This codec has the features of the 3/2-stereo plus low frequency enhancement, downward compatibility with the smaller number of channels, backward compatibility with the existing 2/0-stereo system(MPEG-1 audio), and multilingual capability. The encoder of this codec consists of 6-channel analog audio input part with the sampling rate of 48 kHz, 4-channel digital audio input part and three TMS320C40 /DSPs. The encoder implements multi-channel audio compression using a human perceptual psychoacoustic model, and has the bit rate reduction to 384 kbit/s without impairment of subjective quality. The decoder consists of 6-channel analog audio output part, 4-channel digital audio output part, and two TMS320C40 DSPs for a decoding procedure. The decoder analyzes the bit stream received with bit rate of 384 kbit/s from the encoder and reproduces the multi-channel audio signals for analog and digital outputs. The multi-processing of this audio codec using multiple DSPs is ensured by high speed transfer of date between DSPs through coordinating communication port activities with DMA coprocessors. Finally, some technical considerations are suggested to realize the problem of real-time operation, which are found out through the implementation of this codec using the MPEG-2 layer II sudio coding algorithm and the use of the hardware architecture with commercial multiple DSPs.

• The Journal of the Acoustical Society of Korea
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• v.33 no.5
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• pp.326-334
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• 2014

This paper proposes a novel covariance matrix estimation scheme for minimum variance distortionless response (MVDR) beamforming. By accurately tracking direction-of-sound source arrival (DoA) information using audio-visual sensors, the covariance matrix is efficiently estimated by adopting a variable forgetting factor. The variable forgetting factor is determined by considering signal-to-interference ratio (SIR). Experimental results verify that the performance of the proposed method is superior to that of the conventional one in terms of interference/noise reduction and speech distortion.