• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.1398-1403
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• 2003

In this paper, a simple method to design an MIMO sound field control system was proposed. The control system was designed to achieve 1) noise attenuation and 2) sound equalization by utilizing feedback and feedforward controllers. The method was based on partial model matching on frequency domain which only required measured frequency response data, or impulse response data in order to tune parameters of the controller. The proposed method was applied to a normal office room and results of experiment showed effectiveness of the proposed method.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.8
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• pp.823-831
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• 2008

Since it needs the light computational load and small database, sound source localization method using time delay of arrival(TDOA method) is applied at many research fields such as a robot auditory system, teleconferencing and so on. Researches for time delay estimation, which is the most important thing of TDOA method, had been studied broadly. However studies about factors for time delay estimation are insufficient, especially in case of real environment application. In 1997, Brandstein and Silverman announced that performance of time delay estimation deteriorates as reverberant time of room increases. Even though reverberant time of room is same, performance of estimation is different as the specific part of signals. In order to know that reason, we studied and analyzed the factors for time delay estimation using speech signal and room impulse response. In result, we can know that performance of time delay estimation is changed by different R/D ratio and signal characteristics in spite of same reverberant time. Also, we define the performance index(PI) to show a similar tendency to R/D ratio, and propose the method to improve the performance of time delay estimation with PI.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.58-65
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• 2001

Recently, a 5-degrees-of-freedom (DOF) reverberation model was proposed as a method of representing subjective perception of reverberation as objective measures[1]. This model approximates sound energy decay curve by five objective measures, widely used in which have been concert hall acoustics. However, it is note worthy that there can be infinite number of impulse responses which correspond to a selected 5-DOF reverberation model. There may exist some filters making very unnatural and unrealistic sound. In this paper, the limitation of the 5-DOF reverberation model when it is used as a filter design criteria is investigated. When a 5-DOF reverberation model is given, additional constraints to get natural reverberation are suggested. This is based on the listening tests for several quite different source sounds.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.37-41
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• 2002

In this paper, we made shorter EDT(early decay time) of room reverberation curve using multiple-channel. The speech signal was processed inverse filtering with full-band and sub-band in the basis MINT, and then the multiple-channel adaptive filters were used LMS (Least Mean Square) and NLMS (Normalized Least Mean Square) algorithm. Experimental results, we could get 1/3 of time reduction at 20dB level in the reverberation curve using full-band NLMS when two microphones were used. Also, it is shown that the speech articulation was improved 80% from the test listeners with the speech, which was to shorten EDT by MINT in the subjective assessments using real room impulse response.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.17-30
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• 1999

This paper proposes a new stereophonic acoustic echo canceller to prevent impairments on the voice quality and to remove acoustic echo effectively appearing in stereo environment at the instant of abrupt change of the transmission room environment in teleconferencing system. In stereophonic acoustic echo canceller, the major defective problems are the large computational complexity of estimating echo path systems due to the long impulse response of the true echo paths and the performance degradation of echo canceller due to large correlation between dual stereo signals. Moreover, the change of the suboptimal solution for the echo canceller was considered as a critical deficient factor on to the performance of stereophonic echo canceller. To overcome these problems, this paper proposes pseudo stereophonic acoustic echo canceller using Hyper-plane projection algorithm, which shows the robustness to the environment change of the transmission room and the efficiency of computational complexity.

• Research in Community and Public Health Nursing
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• v.13 no.1
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• pp.137-146
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• 2002

Purpose: To analyze the exposure of middle school students to pornography. Method The Questionnaire. The data collected were from November 6 to November 12, 2000. The subjects of this study consisted of 664 middle school students in Kangwon Province. The data collected were analyzed using, percentage, mean, standard deviation, t-test and Pearson Correlation. The Result: The most frequently viewed program was computer pornography. Except no response about eighty-five percent (85%) of the subjects viewed the pornography prior to elementary school. Moreover, The subjects viewed pornography for the first time in a PC room. Lastly, this study found that there was a significant relationship between the frequency. consumed time and numbers of viewing media violence and the impulse to imitate and imitation acting after viewing the pornography. Conclusion: Realistic and future-directed strategies should be developed for middle school students to decrease their exposure to pornography.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.04a
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• pp.909-915
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• 2008

Researches for time delay estimation had been studied broadly. However studies about factors for time delay estimation are insufficient, especially in case of real environment application. In 1997, Brandstein and Siverman announced that performance of time delay estimation deteriorates as reverberant time of room increases. Even though reverberant time of room is same, performance of estimation is different as the specific part of signals. In order to know that reason, we studied and analyzed the factors for time delay estimation using speech signal and room impulse response. In result, we can know that performance of time delay estimation is changed by different R/D ratio and signal characteristics in spite of same reverberant time.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.919-926
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• 2013

The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1
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• pp.1-11
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• 2008

To overcome the mid frequency limitation of geometrical acoustic techniques, the phased geometrical method was suggested by introducing the phase information into the sound propagation from the source. By virtue of phase information, the phased tracing method has a definite benefit in taking the interference phenomenon at mid frequencies into account. Still, this analysis technique has suffered from difficulties in dealing with low frequency phenomena, so called, wave nature of sound. At low frequencies, diffraction at corners, edges, and obstacles can cause errors in simulating the transfer function and the impulse response. Due to the use of real valued absorption coefficient, simulated results have shown a discrepancy with measured data. Thus, incorrect phase of the reflection characteristic of a wall should be corrected. In this work, the uniform theory of diffraction was integrated into the phased beam tracing method (PBTM) and the result was compared to the ordinary PBTM. By changing the phase of the reflection coefficient, effects of phase information were investigated. Incorporating such error compensation methods, the acoustic prediction by PBTM can be further extended to low frequency range with improved accuracy in the room acoustic field.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.205-209
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• 2013

A conventional acoustic echo suppressor (AES) considering only room impulse response between a loudspeaker and a microphone eliminates acoustic echo from the microphone input. However, in a nonlinear acoustic echo environment, the conventional AES degraded because of a nonlinearity of the loudspeaker. In this paper, we adopt AES based on the frequency-domain second-order Volterra filter using Least Square method. For comparing performances, we conduct objective tests including Echo Return Loss Enhancement (ERLE) and Speech Attenuation (SA). The proposed algorithm shows better performance than the conventional in both linear and nonlinear acoustic echo environments.