• Journal of KIISE:Databases
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• v.27 no.2
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• pp.209-215
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• 2000

In the on-demand services like AOD(Audio On Demand) over the internet, existing operating systems cannot serve repeatedly requested data efficiently. This paper proposes a web cache architecture. It predicts the songs to be requested in near future, based on the intervals between the requests in the past on the same song and keeps the songs in the web cache. For the replacement strategy of the web cache, LFRR(Least Frequently Requested Recently) is proposed. LFRR replaces the song that has less probability to be requested in near future. The average of the intervals between the requests in the past and the new request is used as the probability of the requests. It is more likely to be requested in near future as the average is less. The web cache decreases the number of disk access extremely, and support to serve more users with restricted resources. From the simulation result based on the data at the AOD site currently operating, it is shown that the high performance enhancement is achieved.

• Proceedings of the Korea Contents Association Conference
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• 2009.05a
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• pp.127-134
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• 2009

In spite of controversial debates on aesthetic issues of computer-generated electronic music, rapid advancement of music technologies in the past decade have resulted proliferation of using virtual software synthesizers and samplers in music composition. Computer-based music production platform has become not only a norm among some of contemporary music composers but also vital apparatus for their compositional process. There are two imperative parts of this compositional process involving sampling in computer-based music production, which are commercially available sample libraries that include pre-recorded audio samples, and music production software that processes them. The purpose of this study is to investigate the process and effectiveness of reconstructive compositional process utilizing distinctive features of sampling on computer music production software. This study addresses issues such as: the definition of audio sampling, how sampling is incorporated in compositional process, and what features of music production software are particularly effective in various musical expressions. The result of this study will hopefully accommodate and fulfill the needs of electronic and acoustic musicians' creativeness.

• Journal of Korean Elementary Science Education
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• v.41 no.2
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• pp.432-442
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• 2022

This study aims to apply a maker education program in an online class for a unit in elementary science entitled "Life of a Plant." Furthermore, it aims to investigate the effect of the program on the interest of students in science and creative problem-solving. The theme of the proposed maker education program is "Making a fairy tale audio book with the theme: the life cycle of a plant." The program consists of activities intended to produce creative audiobooks using concepts learned from the TMI maker education model stage. The result points to the positive effect of maker education on the interest of elementary school students in science and creative problem-solving. Therefore, applying the maker education program to online classes for elementary science is effective in improving the interest of students. In addition, further research is required to effectively apply the maker education program to online classes in science.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.446-453
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• 2004

In this paper, a new adaptive perceptual filter using variable threshold to enhance audio signals degraded by additively nonstationary noise is proposed. The adaptive perceptual filter updates variable threshold each time according to the power of signal and the effect of noise variation. So the noisy audio signal is enhanced by the method which controls a residual noise effectively. The proposed algorithm uses the perceptual filter which transforms a time domain signal into frequency domain and calculates an intensity energy and an excitation energy in bark domain. In this method. the stage updated the response of filter is decided by threshold. The proposed algorithm using vairable threshold effectively controls a residual noise using the energy difference of audio signals degraded by the additive nonstationary noise. The proposed method is tested with the noisy audio signals degraded by nonstationary noise at various signal -to-noise ratios (SNR). We carry out NMR and MOS test when the input SNR is 15dB. 20dB. 25dB and 30dB. An approximate improvement of 17.4dB. 15.3dB, 12.8dB. 9.8dB in NMR and enhancement of 2.9, 2.5, 2.3, 1.7 in MOS test is achieved with the input signals. respectively.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.34 no.1
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• pp.115-127
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• 2024

The recent increase in digital audio media has greatly expanded the size and diversity of sound data, which has increased the importance of sound data analysis in the digital forensics process. However, the lack of standardized procedures and guidelines for sound data analysis has caused problems with the consistency and reliability of analysis results. The digital environment includes a wide variety of audio formats and recording conditions, but current audio forensic methodologies do not adequately reflect this diversity. Therefore, this study identifies Life-Cycle-based sound data elemental technologies and provides overall guidelines for sound data analysis so that effective analysis can be performed in all situations. Furthermore, the identified elemental technologies were analyzed for use in the development of digital forensic techniques for sound data. To demonstrate the effectiveness of the life-cycle-based sound data elemental technology identification system presented in this study, a case study on the process of developing an emergency retrieval technology based on sound data is presented. Through this case study, we confirmed that the elemental technologies identified based on the Life-Cycle in the process of developing digital forensic technology for sound data ensure the quality and consistency of data analysis and enable efficient sound data analysis.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.16 no.4 s.95
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• pp.427-437
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• 2005

In this paper, the perceived audio level metering algorithm of digital audio sound to be able to operate in real-time is proposed. Through analyzing a conventional recommendation ITU-RBS1387-I for objective audio quality analysis, FFT-based loudness metering algorithm is implemented and the real-time method of that algorithm was advised and proved. The proposed method is based on look-up table. In order to prove the proved method, using 23 pure tones and 30 preselected digital audio samples, its performance and operation time is evaluated. Its performance, compared with an original algorithm's, have a good figure of less than $2\;\%$ error even if look-up table related with spectral spreading have large level resolution of $10\;\cal{dB}$. The proposed algorithm take only 1/21 of original algorithm's measuring time. Also, in the proposed algorithm auditory pitch group energy calculation take 1/450 of original algorithm's and excitation calculation take 1/3.57. In conclusion, the proposed algorithm is expected to be implemented into DSP-based real-time loudness meter.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.367-372
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• 2004

디지탈 보청기의 DSP 칩 파라미터들은 보청기 사용자에게 가장 최적의 파라미터들로 구성되어져야 한다. Gennum GB3211과 같은 디지탈 보청기용 DSP 칩은 전용 전자 칩으로써 약 90 여개의 제한된 파라미터들을 조합하여 약 47억여개 이상의 거의 무한한 정도의 다양한 경우 수를 대응하도록 제작되었다. 보청기 사용자의 전기-음향 효과를 극대화하기 위해 가장 최적화된 파라미터 피팅 프로그램을 개발하였다. 컴퓨터 입력이 가능한 오디오그램 청력 역치로부터 여러 다른 Formula를 사용할 뿐 만 아니라, 마이크로폰과 리시버의 보정 효과를 모두 포함하는 최적 보청기 피팅 프로그램을 개발하였으며 몇 가지 사례를 적용해 보였다.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04b
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• pp.529-531
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• 2002

생방송 비디오 프록시는 사용자에게 최신의 비디오 데이터를 전송해야 한다. 그러나 큰 파일을 나누어 미리 캐싱하는 기존의 방법으로는 최신의 데이터를 보여줘야 하는 생방송 VOD의 제약을 만족시킬 수 없다. 본 논문에서는 생방송 비디오 스트림을 대상으로 효과적으로 캐싱할 수 있도록 클라이언트의 요청을 받는 큐와, 생방송 비디오 데이터를 저장하는 버퍼풀을 구성하였고, 이들이 제대로 동작하기 위한 동기화 과정을 설계하였다. 이 구조의 특징은 프록시가 클라이언트들의 요청을 일정시간 동안 모아서, 미디어 서버에게 한번만 요청한다는 데에 있다 미디어 서버는 프록시에게만 요청을 받고 데이터를 전달하게 되며, 모든 부하는 프록시 쪽으로 옮겨간다 결국 프록시 서버의 개수를 늘리거나 성능을 확장하면 서비스 가능한 클라이언트의 수를 확장시진 수 있는, 이른바 확장성의 개선을 기대할 수 있다. 실제로 구현해본 절과 지연시간 개선, 확장성 효과 등 프록시로서 만족할 만한 성능이 나옴을 확인하였다. 비연속적인 비디오 데이터를 대상으로 한 본 논문은, 오디오와 같이 연속적인 성질의 데이터에 대한 처리를 보강한다면 수요가 늘고있는 생방송 VOD서비스에 대비한 프록시 개발에 도움을 줄 것이다.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.61-63
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• 2003

멀티미디어 기술의 발전은 단일 CD 에 동영상을 저장하기 위한 목적에서 개발되었던 MPEG-1을 비롯하여 최근에는 JVT등의 오디오, 비디오 압축 기술로 이어졌다. 뿐만 아니라 이러한 미디어 데이터를 효과적으로 전달하기 위한 스트리밍 기법, 미디어 데이터 간의 동기화 기법 등 다양한 방면에서 연구되고 있다. 이에 맞춰, 최근에는 사용자와의 상호작용을 위한 User Interaction 에 관한 연구가 활발히 진행되고 있으나 대부분의 연구가 Interaction 자체에 대한 정의 및 이에 대한 기본적인 지원에 관한 연구에 국한되고 있다. 따라서 본 연구에서는 구체적인 성능 향상의 관점에서 interaction 을 효과적으로 제공하기 위한 여러 기법들에 관해 알아보고, 보다 나은 이동 단말에서의 interaction 스트리밍 기법을 제안한다.

• 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.