• Journal of the Korean Society for Railway
• /
• v.2 no.1
• /
• pp.28-37
• /
• 1999

Many residential areas are situated near to railroad tracks so that a railroad noise has been one of the major environmental issues. In this paper two important aspects have been investigated in order to properly evaluate the railroad vehicle noise : sound power levels for different types and sound propagation characteristics of the railroad vehicles. For noise source characteristics of railroad vehicles, sound power values for various types of trains that are in active service have been measured. In this paper, domestic railroad vehicles are measured and compared with high speed train(TGV). Based on sound power information of railway vehicles, prediction on the sound pressure level and equivalent noise level near to railway areas have been evaluated.

• The Journal of the Acoustical Society of Korea
• /
• v.15 no.4E
• /
• pp.3-8
• /
• 1996

The bubble size distribution is critical information to understand sound propagation and ambient noise in the ocean. To estimate the bubble size distribution in a bubbly water, the sound attenuation has been only in the conventional acoustic bubble sizing method without considering the sound speed variation. However, the effect of the sound speed variation in bubbly water cannot be neglected because of its compressibility variation. The sound attenuation is also affected by the sound speed variation. In this paper, a coherent acoustic bubble sizing inverse technique is introduced as a new bubble sizing technique with considering sound speed variation as well as the sound attenuation. This coherent sizing method is theoretically verified with the bubble distribution functions of single-size, Gaussian, and power-law functions. Its numerical test results with the coherent acoustic bubble sizing method show good agreement with the given bubble distributions.

• Proceedings of the IEEK Conference
• /
• 1999.06a
• /
• pp.1107-1110
• /
• 1999

In speech signal processing, the accurate decision of the voiced/unvoiced sound is important for robust word recognition and analysis and a high coding efficiency. In this paper, we propose the mehod of the voiced/unvoiced decision using the LSP parameter which represents the spectrum characteristics of the speech signal. The voiced sound has many more LSP parameters in low frequency region. To the contrary, the unvoiced sound has many more LSP parameters in high frequency region. That is, the LSP parameter distribution of the voiced sound is different to that of the unvoiced sound. Also, the voiced sound has the minimun value of sequantial intervals of the LSP parameters in low frequency region. The unvoiced sound has it in high frequency region. we decide the voiced/unvoiced sound by using this charateristics. We used the proposed method to some continuous speech and then achieved good performance.

• 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.

• KIPS Transactions on Software and Data Engineering
• /
• v.8 no.5
• /
• pp.193-204
• /
• 2019

The explosion of data due to the improvement of sensor technology and computing performance has become the basis for analyzing the situation in the industrial fields, and various attempts to detect events based on such data are increasing recently. In particular, sound signals collected from sensors are used as important information to classify events in various application fields as an advantage of efficiently collecting field information at a relatively low cost. However, the performance of sound-event classification in the field cannot be guaranteed if noise can not be removed. That is, in order to implement a system that can be practically applied, robust performance should be guaranteed even in various noise conditions. In this study, we propose a system that can classify the sound event after generating the enhanced sound signal based on the deep learning algorithm. Especially, to remove noise from the sound signal itself, the enhanced sound data against the noise is generated using SEGAN applied to the GAN with a VAE technique. Then, an end-to-end based sound-event classification system is designed to classify the sound events using the enhanced sound signal as input data of CNN structure without a data conversion process. The performance of the proposed method was verified experimentally using sound data obtained from the industrial field, and the f1 score of 99.29% (railway industry) and 97.80% (livestock industry) was confirmed.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2007.11a
• /
• pp.33-36
• /
• 2007

Acoustic holography exhibits the spatial distribution of sound pressure in time or frequency domain. The obtained picture often contains far more than what we need in practice. For example, when we need to know only the locations and overall propagation pattern of sound sources, a method to show only what we need has to be introduced. One way of obtaining the necessary information is to use envelope in space. The spatial envelope is a spatially slowly-varying amplitude of acoustic waves which contains the information of sources' location. A spatial modulation method has been theoretically developed to get a spatial envelope. By applying the spatial envelope, not only the necessary information is obtained but also computation time is reduced during the process of holography. The spatial envelope is verified as an effective visualization scheme in time domain by being applied to complicated sound fields.

• The Journal of the Acoustical Society of Korea
• /
• v.25 no.1E
• /
• pp.25-31
• /
• 2006

Two twin microphones may produce particular patterns of binaural directivity by time delays between the twin microphones. The boundary element method (BEM) was used for the simulation of the sound pressure field around the KEMAR head model in order to quantify the acoustic head effect. The sound pressure onto the microphone was calculated by the BEM to an incident sound pressure. Then a planar directivity pattern was formed by four sound pressure signals from four microphones. The optimal binaural directivity pattern may be achieved by adjusting time delays at each frequency while maintaining the forward beam pattern is relatively bigger than the backward beam pattern. The simulation results were verified by the experimental measurement.

• The Journal of the Acoustical Society of Korea
• /
• v.24 no.3E
• /
• pp.115-122
• /
• 2005

Two twin microphones may produce particular patterns of binaural directivity by time delays between twin microphones. The boundary element method (BEM) was used for the simulation of the sound pressure field around the head model in order to quantify the acoustic head effect. The sound pressure onto the microphone was calculated by the BEM to an incident sound pressure. Then a planar directivity pattern was formed by four sound pressure signals from four microphones. The optimal binaural directivity pattern may be achieved by adjusting time delays at each frequency while maintaining the forward beam pattern is relatively bigger than the backward beam pattern.

• 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.