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

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.23 no.1
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• pp.73-83
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• 2013

In previous works, psychoacoustic parameters have been used for objective quantification. However, these parameters do not agree well with subjective assessment. Therefore, the correlation between psychoacoustic parameters and the subjective rating of door closing sounds of sampled cars is low, and it is not sufficient to use psychoacoustic parameters as an objective metric to quantify the sound quality of door closing sounds. In this paper, a new method is proposed to objectively quantify the sound quality based on physiological acoustics and statistical signal processing. The gammatone filter, as a pre-processing, is used in models of the auditory system and kurtosis, which is the fourth-order moment of temporal signal, and is used to extract information about sound quality quantification for door closing sounds. The new metric obtained through the proposed method is highly correlated with subjective rating, and it is successfully applied to the quantification of the sound quality of door closing sounds.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.2
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• pp.126-132
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• 2012

The technique for estimation of sound source direction is one of the important methods necessary for various engineering fields such as monitoring system, military services and so on. As a new approach for estimation of sound source direction, this paper propose the bio-mimetic localization sensor based on mechanically coupling structure motivated by hearing structure of fly, Ormia Ochracea. This creature is known for its outstanding recognition ability to the sound which has large wavelength compared to its own size. ITTF (Inter-Tympanal Transfer Function) which is the transfer function between displacements of the tympanal membranes on each side has the all inter-tympanal information dependent on sound direction. The peak and notch features of desired ITTF can be generated by using the appropriate mechanical properties. A example of estimation of sound source direction using generated ITTF with monotonically changing notch and peak patterns is shown.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.447-450
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• 2005

Multi-channel artificial reverberation algorithm to control perceived direction and distance is described in this paper. In conventional algorithms using IIR filters, reverberation time is the only parameter to be controlled. Moreover, since the convolution-based conventional algorithms apply only same impulse responses, but not considering sound localization, it was not realistic enough. The new algorithm proposed in this paper utilizes early reflections segmented according to the azimuth from which direct sound comes and controls perceived direction by panning the direct sound, and controls perceived distance by adjusting Energy Decay Curve (EDC) of reverberation and gain of the direct sound. In addition, the algorithm enhances Listener Envelopment(LEV) to make late reverberation incoherent among channels.

• Proceedings of the Acoustical Society of Korea Conference
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• 1985.10a
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• pp.68-71
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• 1985

In this paper, Sentence-Sip E Il Ka Gi Seo U1 E Gan Da was spoken by 4 men and 3 see sound is used for the experiment. A/D conversion time is 30 sec. Data are obtained using the microcomputer and compressed by ADPCM Rate of compression is 1/8. Data compressed by ADPCM are synthesized and compared to the original sound. Rate of speech identification is analysed using the sound pressure, white noise. Coding of ADPCM is done for 5bit. As the result of fixing starting voltage by 2.6V. It is acertained that variable value increases in initial speech signal and then process is made by minimum value "3". From the result of processing, synthesized sound is almost eaual to original sound. Minimum values cause distorition, Dummy Head System is used in this experiment.xperiment.

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1363-1366
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• 2003

This paper introduces computerized version of Hangout(Korean Language) Standard Pronunciation Rule that can be used in Korean processing systems such as Korean voice synthesis system and Korean voice recognition system. For this purpose, we build Petri net models for each items of the Standard Pronunciation Rule, and then integrate them into the vocal sound conversion table. The reversion of Hangul Standard Pronunciation Rule regulates the way of matching vocal sounds into grammatically correct written characters. This paper presents not only the vocal sound conversion table but also character conversion table obtained by reversely converting the vocal sound conversion table. Making use of these tables, we have implemented a Hangeul character into a vocal sound system and a Korean vocal sound into character conversion system, and tested them with various data sets reflecting all the items of the Standard Pronunciation Rule to verify the soundness and completeness of our tables. The test results shows that the tables improves the process speed in addition to the soundness and completeness.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.3
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• pp.24-38
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• 1994

In this paper we develop the software simulator written in a C language for a frequency modulation synthesis and the approximate range of parameters, for a musically satisfactory timbre, obtained by using the software simulator will be applied to develop an algorithm for parameter extraction. For a frequency modulation synthesis, we also develop an algorithm for parameter extraction through waveform analysis in the time domain as well as spectrum analysis using a FFT in the frequency domain. To verify the validity of the developed algorithm as well as software simulator experimentally, we extract parameters for the several music instruments using the suggested algorithm and analyze the synthesized sound by applying the parameters to the software simulator. The evaluation of the synthesized sound is first done by listening the sound directly as a subjective testing. 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.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.5
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• pp.109-114
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• 2011

In this paper, we apply a context awareness based on Gaussian mixture model (GMM) to a sound reinforcement for hearing impaired. In our approach, the harmful sound amplified through the sound reinforcement algorithm according to context awareness based on GMM which is constructed as Mel-frequency cepstral coefficients (MFCC) feature vector from sound data. According to the experimental results, the proposed approach is found to be effective in the various acoustic environments.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.41 no.3
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• pp.33-38
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• 2004

In this paper we presents a underwater sound recognition algorithm by which we can identify the sound signal without the influence of disturbances due to underwater environmental changes. The proposed method provides a means suitable for acoustic releaser which require low power dissipation and long-time underwater operation. We demonstrate its ability of securing stability and fast sound recognition through both numerical and experimental methods.

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