• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.2
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• pp.24-29
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• 2009

Sound source localization system is used in a robot, a video conference and CCTV(Closed-circuit television) systems. In this Sound source localization systems are applied to human and they can receive a number of sound data frames during speaking. In this paper, we propose methods which is reducing angle estimation error by selecting sound data frame which can more precisely compute the angles from inputted sound data frame. After selected data converted to angle, the error of sound source localization recognition system can be reduced by applying to medium filter. By the experiment using proposed system it is shown that the average error of angle estimation in sound source recognition system can be reduced up to 31 %.

• Journal of the Korean Institute of Landscape Architecture
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• v.29 no.5
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• pp.28-36
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• 2001

The purpose of this study is to research the influence of soundscape in the preference of landscape. Specifically, standards type of communication are applied to the landscape such as positive scenery and negative scenery. The spatial image was analyzed by the variables of Kaplan\`s information process model. The level of visual preferences was measured by a type of acoustic information and visual information in the landscape, and this data was analyzed by multiple regression. The results of this study can be summarized as follows; The value of landscape preference was maximum value in Type I and minimum value in Type II from all fluent of coherence, mystery, and legibility to, except complexity, and it was not different from preference. These results clearly show the influence of sounds effecting decision of landscape preference. It was different by the type of acoustic information and visual information in landscape. The results of ANOVA among types of acoustic information were differences of mean between positive sound, no sound and negative sound from coherence, mystery, and legibility to, except complexity. These variables may be the major factors which must be considered in planning and designing as the functional basis for quantitative analysis.

• Journal of the Korea Society of Computer and Information
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• v.28 no.9
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• pp.55-62
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• 2023

In this paper, we present a method that utilizes geometry-based sound generation techniques to efficiently handle multiple sound sources, wind turbulence, and temperature-dependent interactions. Recently, a method based on reduced raytracing has been proposed to update the sound position and efficiently calculate sound propagation and diffraction without recursive reflection/refraction of many rays, but this approach only considers the propagation characteristics of sound and does not consider the interaction of multiple sound sources, wind currents, and temperature. These limitations make it difficult to create sound scenes in a variety of virtual environments because they only generate static sounds. In this paper, we propose a method for efficiently constructing a sound map in a situation where multiple sounds are placed, and a method for efficiently controlling the movement of an agent through it. In addition, we propose a method for controlling sound propagation by considering wind currents and temperature. The method proposed in this paper can be utilized in various fields such as metaverse environment design and crowd simulation, as well as games that can improve content immersion based on sound.

• The Journal of the Acoustical Society of Korea
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• v.19 no.1E
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• pp.13-21
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• 2000

In this paper, we present the description and results of experimental study of the sound power measurements based on International Standards ISO 3741 and 3745. The sound power emitted by a calibrated reference sound source was measured in a reverberation room and a free field over a reflecting plane, using the precision methods of International Standards ISO 3741 direct method and ISO 3745, respectively. The sound power measurements carried out in this study give accurate estimation and also show that both methods for determining the sound power levels of a sound source in a reverberation room and a free field over a reflecting plane, according to the ISO 3741 and 3745, respectively, have proved equally good.

• International Journal of Advanced Culture Technology
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• v.8 no.4
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• pp.287-293
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• 2020

More and more people are paying attention to the psychological pleasure and relaxation that sound hearing brings. In most cases, humans seem to have a special preference for natural sounds. Natural sounds are mainly white noise and pink noise such as wind, rain, waves, waterfall sounds, etc. All of these are often considered to be beneficial to human health, but in reality the same category of natural sounds is no different. It will be very different due to space, time and other factors. Each sound can be unique, so people's hearing experience is also different. This paper quantitatively analyzes the spectrum and brain waves to analyze the feeling of hearing the natural Broken Falls sound. In particular, we aim to objectively analyze the objective feeling of Broken Falls sound falling on the human auditory system through sound spectrum and brain waves.

• Journal of the Korea Society of Computer and Information
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• v.27 no.11
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• pp.81-87
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• 2022

In this paper, we propose diffraction-based sound control method to improve sound immersion in a virtual environment. The proposed technique can express the wave and flow of sound in a physical environment and a pattern similar to diffraction in real-time. Our approach determines whether there is an obstacle from the location of the sound source and then calculates the position of the new sound reflected and diffracted by the obstacle. Based on ray tracing, it determines whether or not it collides with an obstacle, and predicts the sound level of the agent behind the obstacle by using the vector reflected and refraction by the collision. In this process, the sound attenuation according to the distance/material is modeled by attenuating the size of the sound according to the number of reflected/refracted rays. As a result, the diffraction pattern expressed in the physics-based approach was expressed in real time, and it shows that the diffraction pattern also changes as the position of the obstacle is changed, thereby showing the result of naturally spreading the size of the sound. The proposed method restores the diffusion and diffraction characteristics of sound expressed in real life almost similarly.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.2
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• pp.962-977
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• 2019

The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.702-705
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• 2002

This paper describes effects of visual information influencing sound localization in the peripheral visual Held under dynamic conditions. Presentation experiments of an audio-visual stimulus were carried out using a movie of a moving patrol car and its siren sound. The tallowing results were obtained: first, the sound image on the timing at the beginning of the presentation was more strongly captured by the visual image than that at the end, i.e., the "beginning effect" was occurred; second, in the peripheral regions, the "beginning effect" was strongly appeared in near the fixation point of eyes.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.574-585
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• 2008

3D sound is of central importance for the virtual reality system, and is becoming increasingly important for the auditory displays and for the human-computer interaction. In this paper, we propose a novel real-time 3D sound representation system for virtual reality. At first, we propose a calculation method of the impulse response for virtual space. To transmit the information of the virtual space, we propose an enhanced DXF file type that contains the material information. And then, we implement the multi-channel sound panning system. we perform the experiment based on computer simulation and prove the utility of the proposed method.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10a
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• pp.4-6
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• 2003

A holistic representation, such as sparse ceding or independent component analysis (ICA), was successfully applied to explain early auditory processing and sound classification. In contrast, Part-based representation is an alternative way of understanding object recognition in brain. In this paper. we employ the non-negative matrix factorization (NMF)［1］which learns parts-based representation for sound classification. Feature extraction methods from spectrogram using NMF are explained. Experimental results show that NMF-based features improve the performance of sound classification over ICA-based features.