• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.5
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• pp.421-427
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• 2010

In order to visualize the transfer of sound waves, we performed real-time processing with the fast operating system of GPU, the Graphics Processing Unit. Simulation by using the method of the discrete Huygens' model was also implemented. The sound waves were visualized by varying the real-time processing, the reflecting surfaces within the two-dimensional virtual sound field, and the states of the sound source. Experimental results have shown that reflection and diffraction patterns for the sound waves were identified at the reflecting objects.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.274-277
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• 2005

Panning is the way in which to realize a spatial sound in MIDI by moving sound images by the loudness of each channel. However, there is a limitation for the natural spatial sound. The HRTF (Head Related Transfer Function) has been widely known as one of the ways to realize spatial sound using the two channels, but it needs much processing power. It is very hard to implement a real time processing structure. In this paper, we propose an improved 3D sound model for the spatial sound location by changing the acoustic parameters. We could get a good result from the experiment with MIDI Pan and our Model.

• The Transactions of the Korea Information Processing Society
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• v.5 no.3
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• pp.851-861
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• 1998

Korean language is to be characterized by the rich vocal sound variation. In order to increase the probability of vocal sound recognition and to provide a natural vocal sound synthesis, a systematic and thorough research into the characteristics of Korean language including its vocal sound changing rules is required. This paper addresses an effective way of vocal sound recognition and synthesis by providing the design and implementation of the Korean vocal sound variation rule. The regulation we followed for the design of the vocal sound variation rule is the Phonetic Standard(Section 30. Chapter 7) of the Korean Orthographic Standards. We have first factor out rules for each regulations, then grouped them into 27 groups for eaeh final-consonant. The Phonological Change Processing System suggested in the paper provides a fast processing ability for vocal sound variation by a single application of the rule. The contents of the process for information augmented to words or the stem of innected words are included in the rules. We believe that the Phonological Change Processing System will facilitate the vocal sound recognition and synthesis by the sentence. Also, this system may be referred as an example for similar research areas.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.1009-1009
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• 2003

VIPER a new tool for the VIsual PERception of sound quality and for sound design will be presented. Requirement for the visualization of sound quality is a signal analysis modeling the information processing of the ear. The first step of the signal processing implemented in VIPER, calculates an auditory spectrogram by a filter bank adapted to the time- and frequency resolution of the human ear. The second step removes redundant information by extracting time- and frequency contours from the auditory spectrogram in analogy to contours of the visual system. In a third step contours and/or auditory spectrogram can be resynthesised confirming that only aurally relevant information were extracted. The visualization of the contours in VIPER allows intuitively to grasp the important components of a signal. Contributions of parts of a signal to the overall quality can be easily auralized by editing and resynthesising the contours or the underlying auditory spectrogram. Resynthesis of time contours alone allows e.g. to auralize impulsive components separately from the tonal components. Further processing of the contours determines tonal parts in form of tracks. Audible differences between two versions of a sound can be visually inspected in VIPER through the help of auditory distance spectrograms. Applications are shown for the sound design of several interior noises of cars.

• Proceedings of the IEEK Conference
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• summer
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• pp.237-243
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• 2004

A multichannel smart sound sensor capable to detect and identify sound events in noisy conditions is presented in this paper. Sound information extraction is a complex task and the main difficulty consists is the extraction of high­level information from an one-dimensional signal. The input of smart sound sensor is composed of data collected by 5 microphones and its output data is sent through a network. For a real time working purpose, the sound analysis is divided in three steps: sound event detection for each sound channel, fusion between simultaneously events and sound identification. The event detection module find impulsive signals in the noise and extracts them from the signal flow. Our smart sensor must be capable to identify impulsive signals but also speech presence too, in a noisy environment. The classification module is launched in a parallel task on the channel chosen by data fusion process. It looks to identify the event sound between seven predefined sound classes and uses a Gaussian Mixture Model (GMM) method. Mel Frequency Cepstral Coefficients are used in combination with new ones like zero crossing rate, centroid and roll-off point. This smart sound sensor is a part of a medical telemonitoring project with the aim of detecting serious accidents.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.4
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• pp.23-29
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• 2003

In this paper, we propose an idea for the improved 3-D sound system using conventional stereo headphones to obtain a better sound diffusion from the mono-sound recorded at an anechoic chamber. We use the HRTF(Head Related Transfer Function) for the sound localization and the wavelet filter bank with time delay for the sound diffusion. And we test the modified HRTF with the various sampling rate. We investigate the effects of the 3-D sound depending on the length of time delay at lowest frequency band. Also the correlation coefficient of the signals between the left channel and the right channel is measured to identify the sound diffusion. At last we obtain the diffusion sound using Cool Edit for reverberation.

• The KIPS Transactions:PartB
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• v.14B no.4
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• pp.273-280
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• 2007

This paper presents a new 3D sound toolkit called $EVE-Sound^{TM}$ that consists of pre-processing tool for environment simplification preserving sound effect and 3D sound API for real-time rendering. It is designed so that it can allow users to interact with complex 3D virtual environments by audio-visual modalities. $EVE-Sound^{TM}$ toolkit would serve two different types of users: high-level programmers who need an easy-to-use sound API for developing realistic 3D audio-visually rendered applications, and the researchers in 3D sound field who need to experiment with or develop new algorithms while not wanting to re-write all the required code from scratch. An interactive virtual environment application is created with the sound engine constructed using $EVE-Sound^{TM}$ toolkit, and it shows the real-time audio-visual rendering performance and the applicability of proposed $EVE-Sound^{TM}$ for building interactive applications with complex 3D environments.

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

• Journal of the Institute of Convergence Signal Processing
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• v.16 no.2
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• pp.40-43
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• 2015

In this paper, the effects of the leading and the following sounds with single frequency on sound localization are investigated. The sounds with different levels and ISIs(Inter Stimuli Intervals) were used. The width of test sound is 2ms, and those of the leading and the following sounds are 10ms. 1 kHz of the test sound is utilized. The arrival time difference in the subject's ears is set to be 0.5ms. The four kinds of level differences used for one ISI are 0, -10, -15, and -20dB interval. The leading sound is found to have more effect on sound localization than the following sound is. The effect of the leading sound is also found to be dependent on the value of ISI. When the value of the ISI is small, different effects affecting the sound localization are observed.

• The Journal of the Acoustical Society of Korea
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• v.30 no.3
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• pp.115-122
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• 2011

In this paper, we present a method to estimate an accurate real-time sound source direction based on time delay of arrival by using generalized cross correlation with four cross-type microphones. In general, existing systems have two disadvantages such as system embedding limitation due to the necessity of data acquisition for signal processing from microphone input, and real-time processing difficulty because of the increased number of channels for sound direction estimation using DSP processors. To cope with these disadvantages, the system considered in this paper proposes hardware design for enhanced real-time processing using microphone array signal processing. An accurate direction estimation and its design time reduction is achieved by means of an efficient hardware design using spatial segmentation methods and verification techniques. Finally we develop a system which can be used for embedded systems using a sound codec and an FPGA chip. According to experimental results, the system gives much faster real-time processing time compared with either PC-based systems or the case with DSP processors.