• Proceedings of the Acoustical Society of Korea Conference
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• 1984.12a
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• pp.91-95
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• 1984

Because of layer and scattering in the ocean, there are some problem in algorithm currently used for the recognition of targets. Those are time delay of processing and circuit design. The simple method of detecting direct sound wave in noise caused by time delay is proposed-recognized, estimated, and then direcxt sound wave is reconstructed by the AMDF and $\mu$-processor. 2KHz, 4KHz, 8KHz, 12KHz, 16KHz sound waves are used in experiment. To obtain a reference signal, anechoic water tank is used is processing and aluminium water tank used instead of real ocean. As a result, there are a few errors which caused by anechoic water tank error, decreasing of frequency make errors. Possibility of application to Sonar Signal Processing is proved.

• Journal of KSNVE
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• v.9 no.4
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• pp.687-692
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• 1999

Digital holographic reconstruction method [W. A. Veronesi, et al, 1989, JASA, Vol. 85, pp. 588∼598] reconstructs a whole sound field by using measured sound pressure. This paper reports numerical errors of the method that occur at specific frequencies. The errors occur due to the truncation errors included in the calculation of transfer matrixes. The frequencies of the errors depend on the size of boundary element surfaces. Moreover, a modified calculation technique is proposed in the paper. The technique prevents the truncation errors by employing an indirect calculation procedure.

• Journal of the Korean Institute of Navigation
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• v.25 no.4
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• pp.371-379
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• 2001

We have reported the new measurement system which was substituted digital filter for the analog filter in order to develop the optimal system that could find the time delay between each sensors with high accuracy. And also we have confirmed through the experiments that the accuracy of measurements were differentiated by the methods what kind of digital filter had been adopted. This paper suggests two algorithms which approximate the sound source's bearing and distance. One is that sound source's relative bearing can be approximately regarded as the gradient of hyperbolic asymptote, the other is that the source's range can be approximated under the condition of a long range source relative to the sensor's interval. And a series of experiments were carried out with the source's distance 22.42meters and the random bearing interval within the limits of $-90^{\circ}$~$+90^{\circ}$. As a result, we have recognized that the approximation methods could measure the bearing and distance with higher accuracy than the method using trigonometric relation could do.

• International journal of advanced smart convergence
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• v.13 no.1
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• pp.77-84
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• 2024

This study delves into the subtle sounds emanating from beneath the earth's surface to unveil hidden messages and the movements of life. It transforms these acoustic phenomena into digital data and reimagines them as visual elements. By employing Sismophone microphones and utilizing the FFT function in p5.js, it analyzes the intricate frequency components of subterranean sounds and translates them into various visual elements, including 3D geometric shapes, flowing lines, and moving particles. This project is grounded in the sounds recorded in diverse 'spaces of death,' ranging from the tombs of Joseon Dynasty officials to abandoned areas in modern cities. We leverage the power of sound to transcend space and time, conveying the concealed narratives and messages of forgotten places .Through the visualization of these sounds, this research blurs the boundaries between 'death' and 'life,' 'past' and 'present,' aiming to explore new forms of artistic expression and broaden perceptions through the sensory connection between sound and vision.

• Transactions on Electrical and Electronic Materials
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• v.10 no.6
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• pp.222-226
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• 2009

The objective of this study is to share multimedia contents included in existing digital devices and to solve the problems of an increase in installation fees and non-environmentally friendly interiors. This study designed a new digital signal transmitter and receiver using power line transmission and HDMI in order to solve the problems in the existing systems. The transmitter and receiver designed in this study used an AD9867BCPZ PLC chip in which the transmission came from digital signals originating in a PC, and the system architecture was configured so that the outputs signals were connected to a TV from the receiver. The experiment was implemented by adding a Video Test Generator, a USBPre external sound card, and Smaart Live 6 for analyzing the characteristics of the configured system. In the video test results, it was verified that communication was actively implemented, and the image quality showed a constant level from the measurement of the captured video. In the case of the sound, it was verified that more than 90% of the sound signals were normally transmitted and received from the examination of their phases and magnitudes. Thus, the performance of the system designed in this study was verified, which leads to the resolution of some of the problems found in current digital devices.

• Journal of the Korea Computer Industry Society
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• v.3 no.1
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• pp.27-34
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• 2002

There is much interest on information communications owing to the rapid development of network and IT(Information Technology). Analog signals are converted into digital signals for information communications. However, it is very difficult to completely erase the distortion induced during the conversion of analog signals such as voices and images into digital signals. Existing audio graphic equalizer requires very complex processes to calculate the gain and coefficients of the higher-order filter which is required to generate natural sound and to satisfy the need of each person. Unfortunately it is uneconomical and very difficult to embed the existing digital audio equalizer in the system because of the complexity of the existing digital audio equalizer for high quality sound. This paper discusses the design of a new digital audio graphic equalizer(DAGEQ) which can improve system performance and the quality of audio sound, and can be embedded in the system. This new DAGEQ is designed so that the gain can be controlled automatically. The automatic control of coefficients and gain empowers real time processing and the improvement of audio quality.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.435-438
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• 2005

In this paper, we implemented a system that is called sound engine in musical synthesizer and synthesized a sound of Gayageum using TMS320C6713 DSK. Sound engine consists of two parts: synthesis algorithm and processor. We improved physical modeling using digital waveguide as a synthesis algorithm and we used TMS320C6713 as a processor. The excitation signals that make timbre are stored in memory. When we input parameters, sound engine synthesizes sound of Gayageum. The experimental result shows that synthesized sounds are very similar to real sounds.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2003.05b
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• pp.96-102
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• 2003

Today, stage technology is developing highly by application of digital computer. Performance is composed of audio/video and acoustic technology takes very important position in field of stage technology. Generally speaking, four factors of sound are loudness, pitch, sound timbre and duration. Loudness depends on sound pressure level, yet partly related with spectrum and dulation. Pitch depends mainly on frequence and have a relation with sound pressure and duration. sound timbre depends strongly on spectrum and have a relation with frequence. In this paper, I designed a multi-microphone system which can used in broadcasting and performance stage with vicboss 200MHz-VHF wireless microphone and vicboss 900MHz-VHF wireless microphone. I also studied about multi-microphone which can use conveniently in the super play that needs many microphones. If this multi-microphone is prodused, we could expect better sound quality and a big progress in stereo recording technology.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.3
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• pp.55-59
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• 1984

This paper describes a method of Hangout recognition using 2 - D digital filtering. The 170 patterns classified by the positions of the initial sound (consonant), middle sound (vowel) and terminal sound (consonant) of the 1,659 characters were established and models formed by using 2 - D digital filtering for each patterns were obtained. Based on these models we proposed an algorithm that can recognize KOREAN combinational characters by separating patterns from them with superpostion principles. As a result of simulation, 100% of recognition rate is obtained in the case of the print letter.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.49 no.3
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• pp.69-74
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• 2012

This paper shows the results about field operating digital hearing aids which protect the ears of the battle field soldiers from explosive sound and minimize the difficulty of mutual communication during the battle. The essence of the hearing aid is in its signal compression technology in which soft sound is amplified while rapidly increased explosive sound is attenuated. This nonlinear compression technology can be applied for the protection of the ears of the battle field soldiers. As a core part of the hearing aid, when a new DSP IC chip is launched, the modified firmware and fitting software is developed for adaption. Ezairo 5910 which was recently launched by DSP factory in Canada was used for the development of the firmware of the hearing aid.