• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.455-458
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• 2004

소나 시스템의 표적 탐지 성능을 향상시키기 위해 물리적으로 제한된 길이의 어레이를 확장하는 합성 어퍼쳐 소나에 대한 연구와 실험이 이루어지고 있다. 본 논문에서는 왜곡된 어레이의 형상을 추정하여 왜곡을 보상한 후에 빔 영역에서 어레이를 합성하는 FFTSA(Fast Fourier Transform Synthetic Aperture) 기법의 성능을 분석하였다. 실험 데이터로는 한국 근해에서 견인 어레이로부터 획득한 데이터를 이용하였으며 실측된 데이터로 부어레이 간의 시-공간적인 위상 차이를 보상함으로써 어레이 길이를 확장하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.77-80
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• 2001

본 논문에서는 HMM알고리즘을 이용한 중규모급, 화자독립, 연결음성시스템에서의 인식성능 향상을 위해, 단어 인식기가 가지고 있는 고려사항들 중에 잡음(Noise)에 강한 모델을 위해 동작환경에 따른 적절한 필터를 구성하고 이차적으로 특징 파라미터를 개선하여 Noise를 보상하는 방법을 적용하였다. 인식기의 성능에 큰 영향을 미치는 요인중 하나인 전처리 기능의 평가로 성능향상의 요인을 찾아 음질개선을 위한 보다나은 잡음보상 방법을 제시하고자 하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.171-174
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• 2002

음성신호는 성문특성으로 인해 고주파 특성이 약화되는 경향이 있다. 이를 보상하기 위해 Pre-emphasis filter를 사용한다. 수식으로 표현하면 y(n)=s(n)-As(n-1) 와 같이 차분방정식으로 나타낼 수 있다. 여기서 A값은 보통 0.9에서 1사이의 값을 주로 사용한다. 그러나 Pre-emphasis filter는 고주파 특성을 보상하는 과정에서 극점과 같이 영점도 왜곡된다. 본 논문에서는 음성특성에 따른 LSP(Line Spectrum Pairs) 분포특성을 이용하여 영점을 보존하고 vocoder 및 coding에 필연적인 고주파 특성 혹은 저주파 특성을 강조한다.

• Journal of the Korean Society for Precision Engineering
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• v.25 no.5
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• pp.89-95
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• 2008

The Acousto-Optic Tunable Filter (AOTF) is a high-speed full-field monochromator which generates two spectrally filtered light beams with ordinary and extraordinary polarization state. Thus, AOTF is widely used to build full-field spectral imaging system or spectral domain interferometer. However, AOTF has a big problem that the angle of diffracted light changes according to the scanning of wavelength, which makes image shift on CCD plane In this paper, we propose an analytic design of prism system to compensate the image shift. The detailed analysis of light paths in a prism and basic experimental results are presented to verify our proposed compensation method. The experimental results agree with simulation results based on suggested prism model and image shift is minimized at optimal condition. Also, it can be extended to compensate the image shift for ordinary and extraordinary polarized light simultaneously.

• The Journal of the Acoustical Society of Korea
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• v.38 no.2
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• pp.240-245
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• 2019

This paper is a study result on long distance transmission of underwater acoustic sensor data over cable. The data transceiver is designed using the LVDS (Low Voltage Differential Signaling) transmission scheme, and the jitter characteristics are analyzed by measuring the long distance transmission signal through the cable. In order to reduce the jitter, a pre-emphasis technique is applied to compensate the transmitting signal to be attenuated by long distance transmission, and the transmission characteristics were verified according to the distance.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.603-611
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• 2003

This paper proposes an effective feature compensation scheme based on speech model for achieving robust speech recognition. Conventional model-based method requires off-line training with noisy speech database and is not suitable for online adaptation. In the proposed scheme, we can relax the off-line training with noisy speech database by employing the parallel model combination technique for estimation of correction factors. Applying the model combination process over to the mixture model alone as opposed to entire HMM makes the online model combination possible. Exploiting the availability of noise model from off-line sources, we accomplish the online adaptation via MAP (Maximum A Posteriori) estimation. In addition, the online channel estimation procedure is induced within the proposed framework. For more efficient implementation, we propose a selective model combination which leads to reduction or the computational complexities. The representative experimental results indicate that the suggested algorithm is effective in realizing robust speech recognition under the combined adverse conditions of additive background noise and channel distortion.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.58-68
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• 2005

In this paper, we propose a scheme to improve discriminative property in the feature compensation method for robust speech recognition under noisy environments. The estimation of noisy speech model used in existing feature compensation methods do not guarantee the computation of posterior probabilities which discriminate reliably among the Gaussian components. Estimation of Posterior probabilities is a crucial step in determining the discriminative factor of the Gaussian models, which in turn determines the intelligibility of the restored speech signals. The proposed scheme employs minimum classification error (MCE) training for estimating the parameters of the noisy speech model. For applying the MCE training, we propose to identify and determine the 'competing components' that are expected to affect the discriminative ability. The proposed method is applied to feature compensation based on parallel combined mixture model (PCMM). The performance is examined over Aurora 2.0 database and over the speech recorded inside a car during real driving conditions. The experimental results show improved recognition performance in both simulated environments and real-life conditions. The result verifies the effectiveness of the proposed scheme for increasing the performance of robust speech recognition systems.

• The Journal of the Acoustical Society of Korea
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• v.18 no.4
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• pp.38-46
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• 1999

In this paper, two methods were proposed to improve the performance of adaptive feedback cancellation in hearing aids. One is “Orthogonality principle acoustic feedback cancellation method(Orthogonality principle method)” to track optimal solution with monitoring the instantaneous gradient, the other is a method using the CLMS algorithm(CLMS method). In many simulation conditions, adaptive feedback cancellation method proposed in this paper was much better than Greenberg method by Sum-method LMS algorithm which is known the most excellent method by now in case of system mismatch, SNR and segmental SMR. Also. Orthogonality principle method is as good as CLMS method in terms of adaptive feedback cancellation in many simulation conditions.

• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.159-165
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• 2022

In this paper, a frame synchronization structure for the Binary Phase Shift Keying (BPSK) modulation method in underwater acoustic communication was proposed. The proposed frame synchronization structure is largely divided into two. First, the approximate position and frequency offset of the frame are obtained by non-coherent correlation and sliding Fast Fourier Transform (FFT) method. Second, after compensating for the frequency error to the received signal, the exact position of the frame is obtained by coherent correlation method. Maritime experiments were conducted to confirm the performance of the 2-STEP frame synchronization structure. It was showed that the limitations of the non-coherent correlation and sliding FFT method can be verified when the power of the received signal was greatly reduced due to the channel characteristics. As a result, stable frame synchronization could be obtained by compensating for the frequency error and then using the coherent correlation method.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.2
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• pp.99-108
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• 2004

This thesis deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. I used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound quality assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the test chamber which satisfies the ITU-R specifications. I uses the IACC(Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38㏈ separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4 point increased then before.