• The Journal of the Acoustical Society of Korea
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• v.23 no.5
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• pp.370-375
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• 2004

This paper proposes an algorithm that estimates three dimensional array shape calibration about the bottom-mounted sensor array. under the assumption that the active sources are in the far-field with unknown positions. Under some assumptions. we calculate the sensor positions via an algebraic solutions of a least squares problem that the linear equations are related to the sensor positions and directions or arrival. We give examples of algorithm performance from both computer simulations and sea test. We also illustrate the performance of sensor positions estimation as a function of time delay estimation variance and the distribution of the localizing sources.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.3
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• pp.251-260
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• 2009

In this paper, we studied on an attenuation effect of automobile exhaust noise according to the direction of canceling speaker in ANC system. Automobile exhaust noise was recorded at 800 rpm, 3500 rpm and 5000 rpm of a diesel engine. Directions of canceling speaker can be set to $30^{\circ}$, $90^{\circ}$ and $150^{\circ}$ against the primary noise flow by acrylic ducts to be made for the experimentation. DSP board with TMS320C6416 chip of Texas Instrument Co. used to control the ANC system. The algorithm of this ANC system applied the Filtered-x-LMS algorithm that is modified to compensate for a property of DSP input signal and the secondary-path effect. As an experiment result, the direction of canceling speaker was proved to influence the reduction effect of noise. The $150^{\circ}$ duct in the attenuation effect of noise showed a better result than the $90^{\circ}$ or $30^{\circ}$ duct.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.3
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• pp.251-255
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• 2010

In this study, the effects leading and following tone with single frequency on sound lateralization were investigated. The tone with level difference and ISI(Inter Stimuli Interval) were used. The width of test tone was 2ms, leading tone and following tone were 10ms and 1kHz was used. The arrived time difference of subject's ears 0.5ms. We set four levels on each ISI and let them decide whether they hear the provided sound from left or right. As a result, it knew the fact that leading tone had more effect on sound lateralization than following tone.

• The Journal of the Acoustical Society of Korea
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• v.40 no.6
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• pp.615-625
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• 2021

Prior to proposing appropriate standard for subway station platform, this study conducted field measurements to examine characteristics of room acoustics on platform of two subway stations. As a result of analyzing the longitudinal length of the platform, Sound Pressure Level (SPL) decreased (maximum difference : 14 dB), Reverberation Time (RT) tended to increase (maximum difference of 0.8 s ~ 1.5 s), and C50 and D50 were decreased (maximum difference: 5.9 dB ~ 9.1 dB and 31.8 % ~ 37.6 %, respectively) as measurement positions moved away from the sound source. The Interaural Cross-correlation Coefficient (IACC) did not show clear tendency, but it was lower than 0.3 in entire points. It is judged that the subway platform has non-uniform sound field characteristics due to various combinations of direct and reflective sound even though it is finished with a strong reflective material.This indicates that the room acoustic characteristics of the near and far sound field are clearly expressed depending on the source-receiver distances in the subway platform having a long flat shape with a low height compared to the length.Therefore, detailed architectural and electric acoustic design based on the characteristics of each location of speaker and sound receiver in the platform is required for an acoustic design with clear sound information at all positions of the platform.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.4
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• pp.352-356
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• 2010

In this study, it is discussed how the directional property of the sound lateralization is influenced by proceeding or succeeding tone. The acoustic source applied here is a reference sound which has 0.5 msec interaural time difference(ITD). Based on this reference sound, interfering sounds with five levels of magnitude are applied to the subjects with four kinds of inter-stimuli time intervals(ISI). The interfering sounds are also added as two different types, proceeding tone and succeeding tone. Additionally, in order to investigate a frequency influence, the reference sound and the interfering sounds are generated by using 2kHz, 4 kHz and a white noise. As a result, the influence on lateralization by proceeding tone is lager than that by succeeding tone. It can consider this result as the effect of temporal masking on lateralization. Moreover, there are small differences of masking effect on lateralization by combinations of pure tone. This result shows that the dependency of frequency domain between reference sound and interfering sound is small on the sound lateralization.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.3-10
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• 2000

This paper proposes acoustic holographic measuring system to estimate an absolute position of sound source. Using the measured signals, the estimation of the position is calculated by the Cross-spectrum algorithm of the beamformed signal and a linear arrayed microphone's signals. As the results of comparing the reference microphone method with beamforming method through the measurement of sound field, the beamforming acoustic holographic method is progressed above 20 percent than that of a reference microphone method in the resolution, and the utility of the proposed system could be confirmed.

• The Journal of the Acoustical Society of Korea
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• v.31 no.3
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• pp.197-204
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• 2012

In this thesis, studied the method to minimize the changes in frequency response and level due to the variation of the distance from the source to the microphone. selecting three microphones (omni directional, cardioid, super cardioid) which are being used generally, frequency responses were measured in accordance with the distance changes. Gotten the difference from the reference as the result of measurement, changed responses for each frequency range were compensated in comparison of the original human vocal source. In low frequency range, the low frequency boost caused by the proximity effect and decrease in accordance with the distance were compensated. The variation in mid-frequency range is comparatively small, however since the mid-range is the most important part of the human vocal signal, were compensated the mid-frequency range in comparison of the reference. The human vocal signal variation in high frequency range is extremely small and the high frequency is compensated close to the original source without difficulty. Understanding the microphone characteristics and compensations, this study showed that the response can be maintain among the change of the distance from the source to the microphone.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.245-255
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• 2002

This thesis deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. I used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound quality assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the test chamber which satisfies the ITU-R specifications. I uses the IACC (Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38 dB separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4 point increased then before.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.109-116
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• 2010

In this paper, we propose a method for extracting predominant melody of polyphonic music based on harmonic structure. Since polyphonic music contains multiple sound sources, the process of melody detection consists of extraction of multiple fundamental frequencies and determination of predominant melody using those fundamental frequencies. Harmonic structure is an important feature parameter of monophonic signal that has spectral peaks at the integer multiples of its fundamental frequency. We extract all fundamental frequency candidates contained in the polyphonic signal by verifying the required condition of harmonic structure. Then, we combine those harmonic peaks corresponding to each extracted fundamental frequency and assign a rank to each after calculating its harmonic average energy. We finally run pitch tracking based on the rank of extracted fundamental frequency and continuity of fundamental frequency, and determine the predominant melody. We measure the performance of proposed method using ADC 2004 DB and 100 Korean pop songs in terms of MIREX 2005 evaluation metrics, and pitch accuracy of 90.42% is obtained.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.2
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• pp.99-108
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• 2004

This thesis deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. I used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound quality assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the test chamber which satisfies the ITU-R specifications. I uses the IACC(Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38㏈ separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4 point increased then before.