• Journal of the Korea Academia-Industrial cooperation Society
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• v.16 no.12
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• pp.8140-8146
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• 2015

The objective of this study is to experimentally investigate the cooling performance characteristics with the consideration of the temperature variations of the enclosure of the 400W ferrofluid active speaker having both woofer and amplifier heat sinks. In order to do this, the heat sinks for both woofer and amplifier was designed ant applied to 400W ferrofluid active speaker. As a result, the cooling performance of the developed 400W ferrofluid active speaker was improved and the temperature of the enclosure after 120 min at steady state increased by $2.8^{\circ}C$ with the increase of the outdoor temperatures from $25^{\circ}C$ to $29^{\circ}C$. In addition, the overall sound pressure level of the developed 400W ferrofluid active speaker showed 111.8 dB and improved 1.9 dB higher than 109.9 dB of the existed speaker.

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.23-30
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• 2000

In this paper, the web-based speaker verification system is designed. To decide the speaker recognition algorithm applied to the web-based speaker verification system, the recognition performance and special features of the text-dependent speaker recognition algorithms(DTW, DHMM, SCHMM) are compared through the computer simulation. Using the results of computer simulation, select DHMM as speaker recognition algorithm at web-based speaker verification system because DHMM has the proper recognition performance and initial training utterance number. And by the three-tier method using the ActiveX, DCOM techniques web-based speaker verification system is designed to be operated in the distributed processing environment.

• Journal of KSNVE
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• v.4 no.2
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• pp.187-198
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• 1994

This study is to describe the effects of the duct termination conditions conditions upon the active noise attenuation system. The adaptive filtering algorithm using FIR filter is implemented for duct noise attenuation. To avoid the instability caused by the acoustic feedback, two methods are considered. One is to use a compensating FIR filter. The other is to utilize uni-directional detecting microphone and uni-directional control speaker. Experimental results show that the reflections of sound from duct terminations greatly reduce the performance of ANC system. The directionality of detecting microphone and control speaker is a major factor to decide ANC performance. When there are some reflections from both duct terminations, the noise attenuation using finite FIR filter is not enough to model the duct plant. Especially, the reflection from the upstream termination reduces the noise attenuation in the frequencies related to the distance between control speaker and upstream termination. The performance of the noise attenuation is found to be largely enhanced by using uni-directional coupler, both on detecting microphone and control speaker, even if the duct system has an arbitrary termination conditions.

• Journal of the Korean Society for Precision Engineering
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• v.11 no.1
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• pp.108-113
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• 1994

Active noise control technique has superior performance in low frequency ranges(50 .approx. 400Hz) to the conventional passive noise control technique. For the feasibility of active noise control, it is required to develop a controller which can implement control algorithm on real-time. In this study, therefore, real-time controller is developed using TMS320c25, high speed digital processor. Unlike conventional DSP board of complete ADD ON type, it is possible for the developed controller to interface with the other computer system easily by series communication for the convenience of program development. Furthermore it is designes to be separated readily as a control device. Active noise control of duct system is implemented ti evaluate a performance of developed device. Active noise control of duct system is implemented to evaluate a performance of developed controller using filtered-x LMS algorithm.

• Journal of Korea Multimedia Society
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• v.6 no.7
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• pp.1131-1138
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• 2003

Active noise control(ANC) is an approach to noise reduction in which a secondary noise source destructively interferes with the unwanted noise is introduced. Generally, the performance of ANC is determined how well a secondary noise tracks noises. A secondary noise is generated from the cancelling speaker and a error sensor pick up error signal. The transfer function between the cancelling speaker and the error sensor is not flat and distorts secondary noises. Consequently, the performance of ANC is degraded by the transfer function. In this paper, a single sensor ANC which considers the characteristics of the speaker and the error sensor is proposed. To reduce distortion of secondary noises, the transfer function is estimated by adaptive inverse modelling and the primary noises are estimated by Kalman filter. Experimental results show that the proposed single sensor ANC effectively attenuates noises.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.04a
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• pp.247-249
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• 2009

This paper investigates active noise control(ANC) of a rectangular cavity using single channel filtered-x least mean square(FXLMS) algorithms to reduce interior noise globally. To obtain global reduction of the interior noise, multichannel active control should be incorporated in general. We, however, examined firstly the optimal location of the secondary speaker that produces a global reduction of the interior noise field. We then investigated the frequency characteristics of the reduction to yield the effective frequency band of the active control system. It follows that the secondary speaker should be located as close to the primary source as possible in order to obtain global reduction.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.955-956
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• 2014

Active noise control is noise reduction method by generate anti-phase control signal for destructive interference of through control speaker. purpose of this paper is create a virtual control source at a using the DBAP(Distance Based Amplitude Panning) algorithm which is one of the three-dimensional sound reproduction method, and verified through the experimentally for noise control method through the virtual control source. We compared active noise method by using one control speaker with active noise control method by using DBAP algorithm.

• Phonetics and Speech Sciences
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• v.9 no.3
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• pp.25-32
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• 2017

Besides their effect on the f0 contour of the following vowel, Korean stops are undergoing a sound change in which a partial or complete consonantal merger on voice onset time (VOT) is taking place between aspirated and lax stops. Many previous studies on sound change have mainly focused on group-normative effects, that is, effects that are representative of the population as a whole. Few systematic quantitative studies of change in adult individuals have been carried out. The current study examines whether the sound change holds for individual speakers. It focuses on inter-speaker and intra-speaker variability on sound change in contemporary Korean. Speech data were collected for thirteen Seoul Korean speakers studying abroad in America. In order to minimize the possible effects of speech production, socio-phonetic factors such as age, gender, dialect, speech rate, and L2 exposure period were controlled when recruiting participants. The results showed that, for nine out of thirteen speakers, the consonantal merger is taking place between the aspirated and lax stop in terms of VOT. There were also intra-speaker variations on the merger in three aspects: First, is the consonantal (VOT) merger between the two stops is in progress or not? Second, are VOTs for aspirated stops getting shorter or not (i.e., the aspirated-shortening process)? Third, are VOTs for lax stops getting longer or not (i.e., the lax-lengthening process)? The results of remarkable inter-speaker and intra-speaker variability indicate a synchronous speech sound change of the stop system in contemporary Korean. Some speakers are early adopters or active propagators of sound change whereas others are not. Further study is necessary to see whether the inter-speaker differences exceed intra-speaker differences in sound change.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.1
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• pp.27-31
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• 2012

In this paper, we constructed an array of speaker system with directivity by using FXLMS(Filtered-X LMS) algorithm and confirmed its directivity. The front $0^{\circ}$ characteristics of the controlled speaker was suppressed by interfering it with the control signal produced with filter coefficients optimized with respect to the $180^{\circ}$ characteristics of the rear speakers. The directivity of the array of rear speakers was measured and the damping effect of the signal from the front speaker array was confirmed. The frequency characteristics and directivity was investigated by using the adaptive filter coefficients on damping, the damping on the control point was verified in all the frequency range. In 100Hz, 200Hz, 1000Hz regime, the damping effect was observed in the range of front $60^{\circ}{\sim}100^{\circ}$.

• ETRI Journal
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• v.31 no.6
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• pp.680-687
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• 2009

This paper reports on an improved piezoelectric microspeaker with a high sound pressure level of 90 dB, a total harmonic distortion of less than 15%, and coherence higher than 0.9. The fabricated Pb(Zr,Ti)$O_3$ (PZT) microspeakers have a thickness of only 1 mm including the speaker frame and an active area of 18 mm${\times}$20 mm. To achieve higher sound pressure and lower distortion, the PZT piezoelectric microspeaker has a well-designed speaker frame and a piezoelectric diaphragm consisting of a tilted PZT membrane and silicone buffer layer. From the simulation and measurement results, we confirmed that the silicon buffer layer can lower the first resonant frequency, which enhances the microspeaker's sound pressure at a low frequency range and can also reduce useless distortion generated by the harmonics. The fabricated PZT piezoelectric microspeakers are implemented on a multichannel speaker array system for personal acoustical space generation. The output sound pressure at a 30 cm distance away from the center of the speaker line array is 15 dB higher than the sound pressure at the neighboring region 30 degrees from the vertical axis.