• Proceedings of the IEEK Conference
• /
• 2007.07a
• /
• pp.427-428
• /
• 2007

In this paper, we propose the method of generating smooth moving sound in a two-channel based 3D sound technique. HRTF is generally used to make the moving sound effect in frequency domain. But, the moving sound using the interpolation has many problems to be resolved as there are only discrete measured point of HRTF database. Therefore, HRTF of the moving sound channel makes effective moving sound by the improved grouping. Informal listening tests show that the proposed improves the effective moving sound much better than the conventional methods.

• Journal of the Korean Society for information Management
• /
• v.36 no.3
• /
• pp.131-148
• /
• 2019

This study constructed a cognitive model of information processing to understand the topic of a sound material and its characteristics. It then proposed methods to generate sound summaries, by incorporating anterior-posterior N400/P600 components of event-related potential (ERP) response, into the language representation of the cognitive model of information processing. For this end, research hypotheses were established and verified them through ERP experiments, finding that P600 is crucial in screening topic-relevant shots from topic-irrelevant shots. The results of this study can be applied to the design of classification algorithm, which can then be used to generate the content-based metadata, such as generic or personalized sound summaries and video skims.

• Journal of the Korean Institute of Telematics and Electronics B
• /
• v.32B no.7
• /
• pp.1025-1035
• /
• 1995

In this paper, we realize a real time digital FM synthesizer based on genetic algorithm using a general purpose digital signal processor. Especially, we synthesize diverse music sounds nicely using a synthesis model consisting of a single modulator and multiple carriers. Also we present genetic algorithm-based technique which determines optimal parameters for reconstruction through FM synthesis of a sound after analyzing the spectrum of PCM data as a standard music sound using FFT. Using the suggested parameter extractiuon algorithm, we extract parameters of several instruments and then synthesize digital FM sounds. To verify the validity of the parameter extraction algorithm as well as realization of a real time digital music synthesizer, the evaluation is first done by listening the sound directly as subjective test. Secondly, to evaluate the synthesized sound objectively with an engineering sense, we compare the synthesized sound with an original one in a time domain and a frequency domain.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2012.04a
• /
• pp.961-963
• /
• 2012

The sound source localization system is used to some area such as robotic system, object localization system, guarding system and medicine. So time delay estimation and angle estimation of sound direction are studied until now. These days time delay estimation is described in LabVIEW which is used to create innovative computer-based product and deploy measurement and control systems. In this paper, the development of signal monitoring platform is presented for sound source localization. This platform is designed in virtual instrument program and implemented in two stages. In first stage, data acquisition system is proposed and designed to analyze time delay estimation using cross correlation. In second stage, data obtaining system which is applied and designed to monitor analog signal processing is proposed.

• Proceedings of the KSPS conference
• /
• 2006.05a
• /
• pp.127-130
• /
• 2006

This paper suggests an algorithm that can estimate the direction of the sound source with three microphones arranged on a circle. The algorithm is robust to microphones' gains because it uses only the time differences between microphones. To make this possible, a cost function which normalizes the microphone's gains is utilized and a procedure to detect the rough position of the sound source is also proposed. Through our experiments, we obtained significant performance improvement compared with the energy-based localizer.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2014.10a
• /
• pp.952-954
• /
• 2014

Urban sound environments consists of various sound sources such as traffic noise, sounds of people and natural sounds affecting the perception of soundscape in a place. However, noise maps based on sound pressure levels could not distinguish different sound sources and limited to represent perception of acoustic environments. Accordingly, soundscape map based on perception of sounds is necessary to provide useful information for the description of the acoustic environment. Therefore, the aims of this study are to examine soundscape perception in different urban contexts including commercial, office, park and residential spaces and to suggest a method for soundscape maps using GIS techniques. Soundscape perceptions and physical characteristics of acoustic environments at various urban contexts were obtained from surveys and acoustic measurements, respectively. The results show that dominant factors affecting soundscape perceptions were different in urban contexts and spatial variations of urban soundscape are closely related to various urban contexts.

• Korean Journal of Fisheries and Aquatic Sciences
• /
• v.36 no.2
• /
• pp.170-177
• /
• 2003

This study is to analyze the influence of the Tsushima Warm Current (TWC) on the variation of sound speed in the southern part of the East Sea. Sound speed is calculated by method of Chen and Millero (1977:, based on the CTD data measured in June of 1996. Sound speed in the central part of the TWC is about $45ms^{-1}$ more fast than that in the other regions without the TWC. Sound speed minimum layer (SML) in the TWC region exists between loom and 341 m, while it exists between 260m and 290m in the non-TWC region. SML distributes along the path of TWC over continental shelf in the coastal waters of Japan.

• International Journal of Advanced Culture Technology
• /
• v.7 no.2
• /
• pp.218-226
• /
• 2019

In modern society, people live in an environment with artificial or natural noise. Especially, the sound that corresponds to the artificial noise makes the noise itself and affects each other because many people live and work in the city. Sounds are generated by the activities and causes of various people, such as construction sites, aircraft, production machinery, or road traffic. These sounds are essential elements in human life and are recognized and judged by human auditory organs. Noise is a sound that you do not want to hear by subjective evaluation, and it is a loud sound that gives hearing damage or a sound that causes physical and mental harm. In this study, we introduce the method of stimulating the human hearing by finger vibration and explain the advantages of the proposed method in various kinds of a noise environment. And how to improve the sound quality to improve efficiency. In this paper, we propose a method to prevent the loss of hearing loss and the transmission of sound information based on proper signal to noise ratio when using portable IT equipment in various noise environments.

• Journal of KIISE:Software and Applications
• /
• v.31 no.10
• /
• pp.1412-1417
• /
• 2004

Sparse coding or independent component analysis (ICA) which is a holistic representation, was successfully applied to elucidate early auditor${\gamma}$ processing and to the task of sound classification. In contrast, parts-based representation is an alternative way o) understanding object recognition in brain. In this thesis we employ the non-negative matrix factorization (NMF) which learns parts-based representation in the task of sound classification. Methods of feature extraction from the spectro-temporal sounds using the NMF in the absence or presence of noise, are explained. Experimental results show that NMF-based features improve the performance of sound classification over ICA-based features.

• JSTS:Journal of Semiconductor Technology and Science
• /
• v.14 no.2
• /
• pp.175-183
• /
• 2014

In this paper, we present the design and implementation of real-time sound localization based on $0.13{\mu}m$ CMOS process. Time delay of arrival (TDOA) estimation was used to obtain the direction of the sound signal. The sound localization chip consists of four modules: data buffering, short-term energy calculation, cross correlation, and azimuth calculation. Our chip achieved real-time processing speed with full range ($360^{\circ}$) using three microphones. Additionally, we developed a dedicated sound localization circuit (DSLC) system for measuring the accuracy of the sound localization chip. The DSLC system revealed that our chip gave reasonably accurate results in an experiment that was carried out in a noisy and reverberant environment. In addition, the performance of our chip was compared with those of other chip designs.