• Journal of Korea Game Society
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• v.2 no.2
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• pp.72-77
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• 2002

Sound contents of these days are in the trend of 3D surround audio which has much higher quality then CD audio with the aid of DVD, Development environments of sound contents should be advanced and the efficient application of this system is quite necessary. In this paper, we proposed an improving method for utilizing sound/audio card systems which are necessory for developing game and multimedia sound contents as well as relate know-hows of producing 3D sounds with sound production tools for their efficient usages and applications.

• Proceedings of the KIEE Conference
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• 1989.07a
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• pp.486-490
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• 1989

In this paper, a program controlling an auto-audio media - cassette deck - by a 16 bit personal computer is studied in order to execute audio and visual learning in CAI. The results of this study are as follows. 1. Audio and visual learning is executed efficiently in CAI. 2. Access rate of voice information to text/image information is about 98% and 60% in "play" and "fast forward" respectively. 3. In "fast forward", quality of a cassette tape affects voice information access rate in propotion to motor driving speed. 4. Synchronizing signal may be mistaken by defects of tape itself.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.872-875
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• 2002

A digital watermarking technique applying psychoacoustic model for audio signal is proposed in this paper. In the watermarking scheme, the pseudo-random bit stream used as a watermark signal is embedded into the audio signal in both speech and music. The strength of the embedded signal is subject to the human auditory system in such a way that the disturbances on host audio signal are beyond the sensing of human ears. The experimental results show that the quality of the watermarked audio signal, in term of signal to noise ratio, can be improved up to 3.2 dB.

• Journal of Korea Society of Industrial Information Systems
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• v.17 no.5
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• pp.37-42
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• 2012

Audio steganography is quite similar to the procedure of modifying the least significant bit(LSB) of image media files. The most widely used technique today is hiding of secret messages into a digitized audio signal. In this paper, I propose a new method for hiding messages from attackers, high data inserting rate is achieved. In other words, based on the LSB hiding method and digitized to change the bit position of a secret message, an encrypted stego medium sent to the destination in safe way.

• Journal of Korea Multimedia Society
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• v.20 no.9
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• pp.1551-1558
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• 2017

Audio power amplifiers have been designed based on the premise that the impedance of loudspeakers is fixed at nominal 4 ohms or 8 ohms. However, it is known that the impedance varies with frequency and takes on the nominal value at some limited frequencies. The principle of the loudspeaker operation reveals that the sound pressure produced by the loudspeaker is proportional to the current flowing in the voice coil, not the voltage between the two terminals. We take the characteristics of the loudspeaker into account and compare the frequency responses of the loudspeaker in voltage-drive mode and current-drive mode via computer simulations, to conclude that the audio amplifier drive mode should be re-considered in an effort to improve the sound quality.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.1450-1451
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• 2013

본 논문에서는 음향 확성 환경과 음향신호의 입 출력 조건에 최적화된 음향 시스템을 실시간으로 생성 및 적용 가능한 음향 신호처리용 시스템으로서, 가변형 필터뱅크 기술 및 상황 적응적 필터 재조합 재배열 기술을 음향 신호처리용 DSP에 적용함으로서 Pro-Audio 장비, 방송 음향장비, 산업 음향장비와 같은 다양한 음향관련 장비에서 고품질 음향 서비스를 제공하기위한 핵심 기술인 가변형 필터뱅크 기반 고품질 음향 DSP 시스템 개발을 제안한다.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.580-588
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• 2009

In this paper, we describe technical trends, standard activities, and upcoming issues relating on UHDTV audio, which requires high quality realistic sound. We also propose a proper solution to it for domestic broadcasting and telecommunication environments.

• The Journal of the Acoustical Society of Korea
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• v.27 no.5
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• pp.229-238
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• 2008

Audio tuning improves not only the sound quality of the car audio but also the quality of the completed car itself. However without the subjective assessment on the users' preferences, it is hard to tune the car audio satisfying them. Even though there are lots of factors to be considered to assess the preferences, only a restricted number of factors should be included in the experiment because the total number of experiments increases rapidly as the number of factors in the experiment increases. A large number of factors make it hard to explore the relationship between the sound quality and the sound characteristics and also makes the panels exhausted. In this paper, 8 sound characteristics, each with 2 levels, are considered for the experiment. An orthogonal design of experiment is suggested to reduce the number of experiments from 256 to 16. The analysis of variance is applied to show that Treble is the most significant characteristic of the reproduced sound of the given pop music. Also Deep Bass, SAD, and the interaction between Treble and SAD are found to be significant. For the given classic music, SAD is the only characteristic which turns out to be significant.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.770-776
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• 2004

The goal of perceptual audio coder is to reduce redundancy and irrelevancy of audio signal based on the concept of masking. Several studies on masking effect reveal that the masking threshold varies as a function of the noise-like or tone-like nature of audio signals. Therefore, tonality of audio signal influences significantly the quality and efficiency of perceptual audio coder In this paper, we propose a new effective algorithm for tonality measure using spectrum energy. Since the proposed algorithm consists of a few transcendental functions and simple operations, it has lower complexity than MPEG psychoacoustic model-II. The proposed algorithm was tested with some audio signals, and DSP implementation showed that the proposed algorithm could be implemented with 3 MIPS. These results illustrate the efficiency of proposed algorithm in both performance and complexity.

• 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.