• The Journal of the Acoustical Society of Korea
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• v.43 no.3
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• pp.344-350
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• 2024

A Reference Sound Source (RSS) is an important standard device employed in measuring sound power. The specifications of RSS is specified in international standards, and it is classified as a major calibration item in the field of acoustic metrology. Since the output power of RSS is affected by the supply voltage, each country needs to secure its own calibration service system. In this study, a procedure for calibrating a RSS is established based on the reverberant room conditions and uncertainty evaluation is conducted. Basically, the calibration procedure can apply a precision measurement process of acoustic power, and here, the measurement method using the reverberation chamber of ISO 3741 is applied. For this purpose, a measurement system is constructed, measurements are conducted with two types of RSS, and measurement uncertainty is evaluated. Through measurement examples, it is confirmed that the non-uniformity of the sound pressure distribution in the reverberation room and the volume measurement uncertainty contributed significantly to the overall uncertainty. Additionally, the influence of input voltage is experimentally examined to examine the uncertainty contribution that can be reflected in acoustic power measurements.

• Proceedings of the Korean Institute of IIIuminating and Electrical Installation Engineers Conference
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• 2007.05a
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• pp.57-59
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• 2007

전자식 안정기 회로구성은 EMI 필터부분과 고주파 공진인버터 그리고 안정기 출력을 제어하는 출력전압 및 주파수를 제어하는 부분으로 구성되며 half-bridge LCC 직렬공진회로가 주로 사용되고 있다. 고주파로 점등하기 때문에 음향공진 현상이 발생하여 아크의 떨림으로 인한 제반 요소들을 해결하기 위한 제어회로를 설계하여 해결한다. 본 논문에서는 메탈할라이드 램프와 전자식 안정기 회로의 시뮬레이션 데이터를 활용하여 전자식 안정기의 입력전압 변동시 출력을 일정하게 유지하는 정전력 회로 방식을 설계하여 보다 안정화된 400W급 전자식 안정기를 설계하고 시작품을 제작하여 얻은 결과를 검증하였다.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.71-75
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• 1997

In this paper, some problems that occur from faulty elements in a direction finding system composed of the linear array are studied and the method which improves the performance is proposed. The fault element means the sensor that has no output or highly reduced gain than other normal sensors. In the case of the presence of faulty elements, the performance of the conventional the spatial spectrum subject to a constraint. The corrected spatial spectrum is obtained by this vector. The computer simulations have been performed to study the performance of the proposed method. We have compared the proposed method with the subaperture processing method of one of the previous works.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.37-48
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• 1995

This paper presents an sstimation of ARMA coefficients of noisy ARMA system using generalized total least square (GTLS) method. GTLS problem for ARMA system is defined as minimizing the errors between the noisy output vectors and estimated noisy-free output. The GTLS problem is solved in closed form by eigen-problem and the perturbation analysis of GTLS is presented. Also its recursive solution (recursive GTLS) is proposed using the power method and the covariance formula of the projected output error vector into the input vector space. The simulation results show that GTLS ARMA coefficients estimator is an unbiased estimator and that recursive GTLS achieves fast convergence.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.105-113
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• 1995

For the case of using analog FM system and CDMA system together in the mobile communications, we experimentally calculate the appropriate frequency offset to improve the frequency efficiency by analizing the interference between both systems. As the results of the analysis, we obtained that frequency offset is 840KHz(Path Loss : 80dB) and 720KHz(Path Loss : 120dB) for the Path Loss when a output of CDMA Base Station is 0.91 W ERP. Therefore, to minimize the frequency offset, a output of Base Station will be controlled corresponding to Cover Area.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.94-100
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• 2001

When using a sample covariance matrix data in paucity of snapshots, adaptive matched field processing will have problem in inverting covariance matrix due to the rank deficiency. The general solutions are diagonal loading and eigenanalysis methods, but there is a significant bias in the power output. This paper presents a quantitative study of bias of power output and the performance of source localization through the simulation and the measured data analysis in fixed source case using the diagonal loading method for the minimum variance processor. Results show that the bias in power output is reduced and the performance of source localization is improved when the number of snapshots is greater than the number of array sensors.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.51-57
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• 1995

In this paper, we study a noise-robust feature extraction method of speech signal based on auditory modeling. The auditory model consists of a basilar membrane, a hair cell model and spectrum output stage. Basilar membrane model describes a response characteristic of membrane according to vibration in speech wave, and is represented as a band-pass filter bank. Hair cell model describes a neural transduction according to displacements of the basilar membrane. It responds adaptively to relative values of input and plays an important role for noise-robustness. Spectrum output stage constructs a mean rate spectrum using the average firing rate of each channel. And we extract feature vectors using a mean rate spectrum. Simulation results show that when auditory-based feature extraction is used, the speech recognition performance in noisy environments is improved compared to other feature extraction methods.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.21-24
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• 2001

본 논문은 신경 회로망을 이용하여, 전화망 대역의 음성, 즉, narrowband 음성에서 wideband 음성을 복원하고자 했다. BP 알고리즘을 사용하는 기존의 신경 회로망의 경우에는 음성과 같이 복잡하고 크기가 큰 훈련데이터에 대해서는 훈련이 제대로 되지 않는 단점이 있다. 그러므로 븐 논문에서는 이를 해결하기 위해 입력으로 들어온 LPC 켑스트럼 벡터를 k-means 알고리즘을 이용하여 미리 정한 개수의 cluster로 나눈 다음, 각각의 cluster에 대해 독립적인 신경 회로망을 적용했다 이로 인해 각각의 신경 회로망은 제한되고 서로 상관관계가 많은 음성들만 훈련하면 되므로, 기존의 신경 회로망에서 생기는 훈련의 정체를 개선할 수 있었다. 또 clustering 과정에서 생기는 오류를 보완하기 위해 후보신경 로망들의 출력에 fuzzy 개념을 적용해서 최종 출력을 내도록 했다 실험 결과에서, 제안한 알고리즘은 기존의 codebook mapping 알고리즘보다 스펙트럼 거리척도에 의한 비교 및 주관적인 음질 평가 양쪽에서 개선된 성능을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.84-88
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• 1999

The accurate estimation of segmental duration is crucial for natural-sounding text-to-speech synthesis. For predicting Korean segmental durations, conventional methods utilized phonemic context, part-of-speech context and locational information in prosodic phrase. In this paper, the tonal information of phonemes is employed for more accurate prediction. After defining two non-boundary tones and six boundary tones, we annotated the tonal label on each syllable of 400 sentences. To predict segmental duration using tonal information, we constructed neural networks with a real-valued output node predicting phonemic duration and trained them by backpropagation algorithm. Experimental results showed that the proposed features are effective for predicting Korean segmental durations, and we got 0.863 correlation coefficient of the observed durations and predicted ones.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.5-9
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• 1997

In this paper, when we use the chirp signal as input signal of ultrasonic signal system the technique for determining the bandwidth of the chirp signal that maximizes the amplitude of the compressed ultrasonic echo signal has been studied. In ultrasonic signal processing systems, the signal-to-noise ratio of the echo signal can be too low due to damping and scattering of the ultrasonic wave during transmission. Method of pulse compression using chirp signal is a means to increase the signal-to-noise ratio in ultrasonic pulse-echo systems. Simulation and experimental results showed that the output signal of ultrasonic system was increased by pulse width of chirp signal and the optimum-bandwidth of the chirp signal was 1.15 times larger than the bandwidth of the ultrasonic system.