• 제어로봇시스템학회:학술대회논문집
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• 1986.10a
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• pp.128-133
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• 1986

설비진단에 응용되는 신호처리의 기법으로는 파워스펙트럼, 바이스펙트럼, 켑스트럼 등이 사용되었다. 파워스펙트럼은 이론적인 면과 계산과정 그리고 신호처리에서의 적용방법등이 잘 알려져서 성공적으로 사용되어져 왔다. 특히 음향분야에서는 여러가지 응용기술이 개발되어 실제계에 적용되고 있으며 계측장비도 파워스펙트럼 해석법에 알맞게 개발되어져 왔다. 파워스펙트럼해석법을 사용하여 진동계를 구성하는 각 요소들의 고유진동수와 진동계 전체를 나타내는 진동파들의 주파수성분 간의 관계에 의하여 진동의 원인 및 소음원 등을 추정하는 것이 가능하다. 그러나 파워스펙트럼은 일반적으로 정상적인 신호를 갖는 진동계에 대한 해석 일 때는 그 이론과 실제가 잘 일치하지만, 진동계 자체가 항시 임의의 주파수를 갖고서 움직일 때 그 해석에는 다음과 같은 문제점이 생긴다. 첫째, 불규칙한 진동계에서는 규칙적인 진동계보다 잡음의 영향을 많이 받기 때문에 실제로 잡음이 진동계의 고유주파수 부근에 있을 경우에는 파워스펙트럼해석으로는 불가능한 경우가 있다. 둘째, 진동파 중에 포함되어 있는 위상이라는 중요한 정보가 없다. 셋째, 시간지연에 따른 진동계의 정확한 정보를 얻을 수 없다. 이상에서 볼 때 파워스펙트럼해석법은 한계가 있음을 알 수 있다. 따라서 본 논문은 바이스펙트럼이라는 해석법을 사용하여 정상과정에서 비정상과정으로 시간지연에따라 변하는 진동계 또는 정상적인 진동계의 저주파에서의 상호간섭 정도 및 위상관계를 관찰함으로써 파워스펙트럼과 비교하여 바이스펙트럼해석법의 타당성을 검토한다. 바이스펙트럼의 실제적인 계산방법은 P. J. Huber가 세가지 접근 방법을 제안했는데 시간영역에서의 평균화를 행하여 계산하는 법, 연속된 기록들을 평균화하는 것, 주파수 영역에서의 평균화를 행하는 것 등이 있다. 본 논문에서는 FFT를 먼저 행하고 파워스펙트럼과 바이스펙트럼 및 바이코히어런스를 구하였다. 그러나 바이스펙트럼해석법은 수치해석적인 면에서 볼 때 파워스펙트럼해석법에 비하여 미약한 점이 많고 통계학적인 그 의미가 확실하게 알려져 있지 않기 때문에 본 논문에서는 시뮬레이션을 통하여 그 물리적 의미를 규명하고져 한다.

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.234-242
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• 2016

In this paper, an on-line noise coherence estimation algorithm for binaural speech enhancement system is proposed. A number of noise Power Spectral Density (PSD) estimation algorithms based on the noise coherence between two microphones have been proposed to improve the speech enhancement performance. In the conventional algorithms, the noise coherence was characterized using a real-valued analytic model. However, unlike the analytic model, the noise coherence between the two microphones is time-varying in real environments. Thus, in this paper, the noise coherence is updated in accordance with the variation of the acoustic environment to track the realistic noise coherence. The noise coherence can be updated only during the absence of speech, and the simulation results demonstrate the superiority of the proposed algorithm over the conventional algorithms based on the analytic model.

• Journal of the Korean Society for Nondestructive Testing
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• v.19 no.2
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• pp.85-92
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• 1999

The fact that material degradation can be evaluated by measuring nonlinear acoustic effect has been proposed by previous studies. The most conventional method to measure nonlinear acoustic effect is to measure the absolute magnitude of fundamental and $2^{nd}$ order harmonic frequency component in the propagated ultrasonic wave. For this aim, power spectral analysis technique has been used widely. However, the power spectral analysis has fatal disadvantage that the gaussian additive noise superimposed in the wave signal remains in the power spectrum domain. Moreover, the magnitude of $2^{nd}$ order harmonic frequency component generated by nonlinear effect is so small that it may be suppressed by the noise remained in the power spectrum. In order to overcome this problem, this paper proposes an alternative method using bispectrum analysis, which can reduce the effect of addictive gaussian noise and. the nonlinear parameter can be obtained more stably. Simulations showed that the proposed method can obtain the value of nonlinear parameter near to the true value in the case of low SNR signal. Also, in order to confirm the usefulness of our method in actual case, we compared the nonlinear parameter obtained by using both of power spectral and bispectral analysis for several specimen intentionally degraded by fatigue load.

• The Journal of the Acoustical Society of Korea
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• v.35 no.2
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• pp.102-109
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• 2016

This paper proposes a noise suppression method through SNR (Singal-to Noise Ratio) estimation in the two microphone array environment of close spacing. The conventional method uses a noise suppression method for a gain function obtained through the SNR estimation based on coherence function from full band. However, this method cause performance decreased by the noise damage that affects all the feature vector component. So, we propose a noise suppression method that allocates a frequency domain signal into N constant multi frequency band and each frequency band gets a gain function through SNR estimation based on coherence function. Performance evaluation of the proposed method is shown by comparison with PESQ (Perceptual Evaluation of Speech Quality) value which is an objective quality evaluation method provided by the ITU-T (International Telecommunications Union Telecommunication).

• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.383-391
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• 2009

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. The waveform which is stationary region of voiced speech is very correlated by pitch period. So we can estimate the SNR by correlation of near waveform after dividing a frame for each pitch. For unvoiced speech signal, vocal track characteristic is reflected by noise, so we can estimate SNR by using spectral distance between spectrum of received signal and estimated vocal track. Lastly, energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced.

• Journal of Advanced Navigation Technology
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• v.20 no.4
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• pp.367-373
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• 2016

We propose an improved direction of arrival (DoA) estimation algorithm based on co-prime array and propagator method. The propagator method with co-prime array does not require singular value decomposition (SVD) requiring much less computational complexity but exhibiting somewhat worse performance in comparison with MUSIC based on co-prime array. We notice that one cause of the performance degradation was in the avoidance of the usage of the diagonal elements of the signal autocorrelation matrix that contains the noise power spectral density. So we propose an algorithm with the diagonal elements of the signal autocorrelation matrix based on the fact that the noise power spectral density can be estimated using noise observation over a long period of time. We observe, through simulations, that the proposed scheme in this paper improves the performance, with 4 times more computational requirement, by signal-to-noise ratio of 1.5dB and by DoA resolution of $0.7^{\circ}$ at the detection probability of 95% compared with the previously introduced co-prime array propagator scheme, resulting in performance much closer to that of co-prime array-based MUSIC scheme.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.38 no.6
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• pp.56-64
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• 2001

In this paper, we proposed the simple and selective method for the order of model that reflected the feature of the heart rate variability without the complicated calculation in the power spectral analysis of heart rate variability using autoregressive model. The power spectral analysis of short-term of heart rate variability using autoregressive have been problem to resolution of spectral estimates by the selective model order. As a result that the proposed method for the order comparative tested with the AIC and the fixed order method, the calculation process could become very simple and select the order which correspond with the feature of the time series. We verified it could removed the noisy power components by the fixed order.

• The Journal of the Acoustical Society of Korea
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• v.30 no.3
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• pp.123-128
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• 2011

In this paper, we propose the improved noise power estimation in speech enhancement under various noise environments. The previous MS algorithm tracking the minimum value of finite search window uses the optimal power spectrum of signal for smoothing and adopts minimum probability. From the investigation of the previous MS-based methods it can be seen that a fixed size of the minimum search window is assumed regardless of the various environment. To achieve the different search window size, we use the noise classification algorithm based on the Gaussian mixture model (GMM). Performance of the proposed enhancement algorithm is evaluated by ITU-T P.862 perceptual evaluation of speech quality (PESQ) under various noise environments. Based on this, we show that the proposed algorithm yields better result compared to the conventional MS method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.9C
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• pp.922-927
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• 2009

Utilization problem of the limited spectrum is the one of the most important issues in wireless communication systems. Cognitive radio technique which is finding and utilizing frequency holes is also one of those techniques. Specially, the spectrum sensing technique to detect the primary user signal is a core technology in cognitive radio area. In this paper, we propose the spectrum sensing algorithm using Bussgang theorem. The proposed algorithm calculates the statistical difference between the Gaussian noise and the primary user signal by applying Bussgang theorem to the received signal. The algorithm is not affected by noise uncertainty and can detect the primary user signal in the very low SNR environment. We evaluate the algorithm through computer simulations with 12 ATSC A/74 DTV signal captures based on IEEE 802.22 WRAN and formulate the sensing threshold for the proposed scheme.