• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.770-774
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• 1994

The volume of an acoustic source is important in determining various acoustic parameters. One of the suggested techniques is the internal pressure method incorporating a loudspeaker attached to a chamber wall and a microphone inserted into the cavity. Although the method is easy to handle with a very simple measurement setup, the coupling effects between the dynamic system of the loudspeaker and acoustic field, and the effects of higher order modes introduced by the discontinuities in the acoustic field, and the effects of higher order modes introduced by the discontinuities in the acoustic field should be considered for precise result. In this study, higher order modes due to the discontinuities of loudspeaker and microphone boundaries are included and the electro-acoustic coupling effects are compensated for by using the results of two cylinders with different lengths. The volume velocity of a loudspeaker thus obtained agrees very with that measured by laser sensor.

• The Journal of the Acoustical Society of Korea
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• v.40 no.3
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• pp.241-246
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• 2021

Modern loudspeaker systems are changing an active loudspeaker system that have multiple input channels and power amplifier. So, we cannot measure these active loudspeakers with IEC 60268-5 standard. This paper introduces an IEC 60268-21 standard that can measure an active loudspeaker having many analog and digital and wireless or wired inputs and having various control parameters.

• The Journal of the Acoustical Society of Korea
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• v.33 no.3
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• pp.184-190
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• 2014

Although it is ideal that the sound with the range of 20 ~ 20,000 Hz is produced by the loudspeaker system composing of only one loudspeaker unit, it is almost impossible now. Therefore, the audible frequency range is now produced by the loudspeaker system, which is consisted of more than two loudspeaker units. The multi-way loudspeaker system certainly requires a crossover filter, which is divided the audible frequency into low and high frequency ranges. The crossover frequency of filter usually is in the range of 1 ~ 4 kHz, the frequency range can largely affect to the sound articulation and quality. It is an example that the phase and amplitude at the crossover frequency can be different from each other, due to the variously reasons. Conversely, they can not produce the smooth frequency response due to the different distance between the two loudspeaker units and electrical properties. As a result, the sound articulation and quality can be degraded. Therefore, the phase and amplitude response at the crossover frequency has to be deeply considered, in order to exactly match the woofer and tweeter. In this study, it is proposed the methods which are the flat frequency and phase response to be obtained by adjusting of the delay time between loudspeaker units, and the choice of filter to be considered the roll-off properties of each unit. It is achieved the frequency response with ${\pm}1$ dB, and the sound articulation is also improved.

• Journal of Satellite, Information and Communications
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• v.8 no.4
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• pp.30-36
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• 2013

Recently the globalization of the media content industry, various activities have been made in the field of art and the speaker system is very important in the sound industry which is one of the arts. The directional characteristics of a loudspeaker refer to the radiation of sound in certain directions and are among the most important features of a loudspeaker. Designing a loudspeaker that can keep all of its constant directivity at all frequencies is difficult due to the wavelengths of audio frequencies and the size of horns and transducers. This study proposed gradient array methods to improve low frequency pattern control of full-range speakers to maximize Direct to Reverberant Ratios at the listeners.

• ETRI Journal
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• v.33 no.6
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• pp.977-980
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• 2011

Wave field synthesis (WFS) has been gathering more and more attention recently due to its ability to perfectly reproduce an original sound field. However, to realize theoretically perfect WFS, a four-sided loudspeaker array that encloses the listener is required. However, it is difficult to build such a system except in large listening spaces, such as a theater or concert hall. In other words, if the listening space is a home, installing a side loudspeaker array is impractical. If the two side walls located to the left and right of the listener can be omitted, a setup using only front and rear loudspeaker arrays may be a solution. In this letter, we present a subjective listening experiment of sound localization/distance based on a WFS using a front and rear loudspeaker array system which is conducted on two listening points and shows average localization errors of $6.1^{\circ}$ and $9.18^{\circ}$, while the average distance errors are -27% (0.5 m) and -29% (0.6 m), respectively.

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.723-731
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• 2009

In this paper, we propose a novel sound field reconstruction algorithm using a three dimensional loudspeaker array for providing realistic sound field to multiple listeners. The proposed algorithm is based on minimization of the squared error between the original sound field and the reconstructed sound field by the loudspeaker array over a predefined three dimensional region of the space using a loudspeaker array surrounding the listening area. For evaluating the proposed algorithm, we constructed the three dimensional array composed of 40 loudspeakers and discuss the relevant experiment results.

• Journal of KSNVE
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• v.8 no.2
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• pp.274-282
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• 1998

A systematic procedure for designing a direct-radiator-type loudspeaker has been developed, based on the numerical vibro-acoustic analysis and the Taguchi method. The finite-element model of the speaker cone has been used to calculated the vibration response of the cone excited by the voice coil. The vibration displacement of the speaker cone has been converted into the vibration velocity and used as a boundary condition for the acoustic analysis. The acoustic frequency characteristics of the loudspeaker have been calculated by the boundary element method. The numerical results have been verified by the experiments carried out in an anechoic chamber. Some design parameters have been selected by using the Taguchi method, and the variations of the acoustic characteristics due to the changes of the parameter values have been examined using the numerical model.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.17 no.8
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• pp.757-765
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• 2007

Enhanced PSPICE circuit model for vented-box loudspeaker system with lumped parameters was proposed to more accurately simulate real vented-box loudspeaker system. Frequency dependent parameters were used to model voice coil inductor. Acoustic pressure response curves and electrical impedance curves were simulated and investigated by PSPICE circuit model according to circuit parameters' variations. Finally, it was used to identify faults detection in woofers manufactured by unskilled persons.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.165-170
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• 1995

In this paper, it is attempted to apply the adaptive digital signal processing to compensate for a linear distortion of a loudspeaker and implement a real time hardware for that purpose. The real time system is implemented by using the DSP56001, a general purpose signal processor, as a host processor and the DSP56200, a cascadable adaptive FIR filter peripheral chip, as an adaptive digital filter. The system has 1000 taps at a 44.1kHz. After inverse modeling of under_compensation_speaker, the system reduces loudspeaker's linear distortions by pre-processing an input audio signal to loudspeaker. The experiment shows satisfactory results; after adaption with white noise as input signal for 60sec, the flat amplitude and linear phase frequency characteristics is found to lie over a wide frequency range of 100Hz to 20kHz.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.11a
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• pp.166-169
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• 2004

In this paper, computational analysis and experiments of subwoofer for car audio speaker system were performed and discussed to analyze acoustical phenomena for subwoofer. Ported enclosure system with subwoofer were manufactured and provided for test and simulation purposes. Subwoofer with single voice coil and double voice coil were identified by linear and nonlinear parameter identification method for loudspeaker parameters. For high power inputs to subwoofer, sound pressure levels were compared according to input powers with linear and nonlinear loudspeaker models. For subwoofer system with high power nonlinear speaker model was showed to be adequate to describe the behaviour of loudspeaker.