• The Journal of the Acoustical Society of Korea
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• v.39 no.2
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• pp.105-112
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• 2020

An analysis model of noise map is proposed to evaluate and reduce the acoustical noise of power plant and its surroundings. The sound powers of many noise sources are estimated by measuring the sound levels of major equipments in the power plant. The analysis of noise has been made by using ENPro that is a commercial program for environmental noise prediction. The proposed model is verified by comparing the results from noise analysis and measurement at several points of the power plant units 1 through 4, and residential areas. It is shown that noise map simulation using the proposed model has a reliability, since the overall noise level approximates within the error of ±2 dB. Furthermore, through noise analysis, the increasing effect of noise due to newly established units 5 and 6 on residential areas is also analyzed. Consequently, the noise barrier is designed to meet an environmental noise standard and satisfy low cost and safety conditions.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.10
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• pp.164-170
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• 2014

A single near-field sounde source bearing and ranging method based on a independent doublet array is proposed. In the common case of bearing estimation method, unform linear array or uniform circular array are used. It is constrained retaining aperture because of array structure to estimate the distance of the sound source. Recent using independent doublet array sound source's bearing and distance esmtimation method is proposed by wide aperture. It is limited to the case doublets are located on a straight line. In this paper, we generalize the case and estimate the localization of a sound source in the various array structure. The proposed algorithm was verified performance through simulation.

• Journal of Broadcast Engineering
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• v.21 no.4
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• pp.621-629
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• 2016

This paper suggests a new structure of hybrid DRC to enhance the psychoacoustic sound quality of a conventional multiband DRC. The proposed hybrid DRC consists of two serially cascaded stages. The front stage DRC is multiband, and it compresses input based on RMS level detection, whereas, the back stage DRC is single band, and it regulates input according to peak level detection. The proposed hybrid DRC shows better loudness while suppressing distortion by clipping. The proposed algorithm was verified through MATLAB simulation, and it was implemented using an FPGA board for listening test. The test result showed that the proposed hybrid structure enhances overall psychoacoustic sound quality compared to conventional structures, which is based on only RMS or peak level detection.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.63-73
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• 1995

The conventional acoustic sounds can be synthesized by Frequency Modulation which includes the variation of frequency, amplitude, and modulation index. In this paper the number of variable synthesis parameters are limited to easily implement the existing two carrier FM algorithm by hardware. The DSP(Digital Signal Processor), which is able to carry out the modified algorithm and synthesize 16 sounds at a time, is designed with $0.8{\mu}m$ standard sells. The DSP which can synthesize 2 sounds at a time is implemented by ASIC emulator to examine the sound quality of the designed DSP. Through the objective and subjective estimation, it is confirmed that the sounds of many instruments from the implemented DSP are very closed to their real sound. Finally the designed DSP is layouted and simulated by VLSI desgn tool. According to the simulation, the designed DSP has the sufficiently fast speed for synthesizing 16 sounds at a time.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1995.04a
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• pp.196-201
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• 1995

The broadband and discrete sources of sound in small cooling fans of propeller type and centrifugal type were investigated to understand the turbulent vortex structures from many bladed fans using ANSI test plenum for small air-moving devices (AMDs). The noise measurement method uses the plenum as a test apparatus to determine the acoustic source spectral density function at each operating conditions similar to real engineering applications based on acoustic similarity laws. The characteristics of fans including the head rise vs. volumetric flow rate performance were measured using a performance test facility. The sound power spectrum is decomposed into two non-dimensional functions: an acoustic source spectral distribution function F(St,.phi.) and an acoustic system response function G(He,.phi.) where St, He, and .phi. are the Strouhal number, the Helmholtz number, and the volumetric flow rate coefficient, respectively. The autospectra of radiated noise measurements for the fan operating at several volumetric flow rates,.phi., are analyzed using acoustical similarity. The rotating stall in the small propeller fan with a bell-mouth guided is mainly due to a leading edge separation. It creates a blockage in the passage and the reduction in the flow rate. The sound power levels with respect to the rotational speeds were measured to reveal the mechanisms of stall and/or surge for different loading conditions and geometries, for example, fans installed with a impinging plate. Lee and Meecham (1993) studied the effect of the large-scale motions like impinging normally on a flat plate using Large-Eddy Simulation(LES) and Lighthill's analogy.[ASME Winter Annual Meeting 1993, 93-WA/NCA-22]. The dipole and quadrupole sources in the fans tested are shown closely related to the vortex structures involved using cross-correlations of the hot-wire and microphone signals.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.26 no.1
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• pp.89-96
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• 2016

TDOA(time delay of arrival) position estimate from acoustic measurement of artillery shell impact is studied in order to develop a targeting evaluation system. Impact position is calculated from the intersections of hyperbolic estimates based on the least square Taylor series method. The mathematical process of Taylor series estimation is known to be robust. However, the concern lays with the accuracy because it is sensitive to the bias caused by the randomness of measurement situation. The measurement error typically occurs from the distortion of waveform, change of travelling path, and sensor position error. For outdoor measurement, a consideration should be made on the atmospheric condition such as temperature and wind which can possibly change the trajectories of rays of sound. It produces wrong propagation time events accordingly. Ray tracing and optimization techniques are introduced in this study to minimize the bias induced by the ray of sound. The numerical simulation shows that the atmospheric correction improves the estimation accuracy.

• Journal of Broadcast Engineering
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• v.19 no.1
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• pp.64-74
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• 2014

The primary and ambient signal decomposition of a stereo sound is a key step to the stereo upmix. The principal component analysis (PCA) is one of the most widely used methods of primary-ambient signal decomposition. However, previous PCA-based decomposition algorithms assume that stereo sound sources are only amplitude-panned without any consideration of phase difference. So it occurs some performance degradation in case of live recorded stereo sound. In this paper, we propose a new PCA-based stereo decomposition algorithm that can consider the phase difference between the channel signals. The proposed algorithm overcomes limitation of conventional signal model using PCA with phase alignment. The phase alignment is realized by using inter-channel phase difference (IPD) which is widely used in parametric stereo coding. Moreover, Enhanced Modified PCA(EMPCA) is combined to solve the problem of conventional PCA caused by Primary to Ambient energy Ratio(PAR) and panning angle dependency. The simulation results are presented to show the improvements of the proposed algorithm.

• Journal of the Korean Society for Nondestructive Testing
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• v.36 no.1
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• pp.27-34
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• 2016

The type of ultrasound transducer used influences the quality of a reconstructed ultrasound image. This study analyzed the effect of transducer type on ultrasound computed tomography (UCT) image quality. The UCT was modeled in an ultrasound simulator by using a 5 cm anatomy model and a ring-shape 5 MHz 128 transducer array, which considered attenuation, refraction, and reflection. Speed-of-sound images were reconstructed by the Radon transform as the UCT image modality. Acoustic impedance images were also reconstructed by the delay-and-sum (DAS) method, which considered the speed of sound information. To determine the optimal combination of transducers in observation, point-source, flat, and focused transducers were tested in combination as trasmitters and receivers; UCT images were constructed from each combination. The combination of point-source/flat transducer as transmitting and receiving devices presented the best reconstructed image quality. In UCT implementation, the combination of a flat transducer for transmitting and a point transducer for receiving permitted acceptable image quality.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.393-398
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• 2002

The conventional bit allocation scheme for MPEG-1 audio encoding searches the subband with minimum MNR(mask-to-noise ratio) repetitively until its operation is completed, which occupies most of its total computational complexity. In this paper, as a computationally efficient approximation of it, we propose a new bit allocation scheme with a simple subband search and compare it with the existing schemes[1][2] in terms of the computational complexity and sound quality. For the performance comparison, we used the pop music signal contained in SQAM(sound quality assess material) CD from EBU. Simulation results show that the computational complexity of the proposed method is about 42% of that of the existing one in [1] and the sound quality difference in terms of MNR between the two schemes is within the 0.2 ㏈, for the case of using the layer II at the bit rate of 128 kbps.

• The Journal of the Acoustical Society of Korea
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• v.40 no.4
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• pp.278-289
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• 2021

In this study, the acoustic performance of a dissipative silencer used in the ship with excellent performance compared to its size was predicted and analyzed using a numerical analysis method to reduce the pipe noise. To this end, the performance of the single expansion chamber-shaped silencer was verified using experimental and numerical analysis methods. The acoustic performance of the silencer was expressed using the Transmission Loss (TL), an indicator of its own performance, and the result was derived using the two-load method, which measured by changing the impedance at the end of the pipe. For the numerical analysis method, a general-purpose finite element analysis program was used, and the Delany-Bazley-Miki model with the flow resistivity of the sound absorbing material as an input parameter was applied. Finally, we compared the experimental and simulated results for each of the acoustic performances of the single expansion type and the dissipative silencer to confirm the consistency of the results, and predicted and analyzed the simulation results for four cases according to the properties of the sound absorbing material.