• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.11-14
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• 2000

본 논문에서는 재귀적 지연추정기를 갖는 적응여파기를 이용하여 음성신호에 내재한 배경잡음을 제거하는 단일입력 적응잡음제거기를 TMS320C30 EVM 보드에서 실시간 구현하였다. 이를 위하여 샘플시간마다 지연정보를 구할 수 있는 재귀적 평균 절대차 함수를 사용하고, 정규화 된 최소평균자승(NLMS: Normalized Least Mean Square) 알고리듬을 사용하는 단일입력 잡음제거 시스템을 시뮬레이션한 (1)의 내용을 EVM 보드에 구현하였다. 그리고, (1)과 동일한 방법으로 백색 가우시안 잡음에 의해 왜곡된 음성에 대하여 SNR(Signal-to-Noise Ratio)에 따른 잡음제거 실험을 하였으며, EVM 보드에서의 실험결과를 (1)의 시뮬레이션 결과와 비교/검토하였다.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.107-113
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• 1997

To achive a robust Korean word recognition system for both channel distortion and additive noise, maximum a posteriori estimation(MAP) adaptation is proposed and the effectiveness of environmental adaptation for improving recognition performance is investigated in this paper. To do this, recognition experiments using MAP adaptation are carried out for the three different speech ; 1) channel distortion is introduced, 2) environmental noise is added, 3) both channel distortion and additive noise are presented. Theeffectiveness of additive feature parameters, such as regressive coefficients and durations, for environmental adaptation are also investigated. From the speaker independent 100 words recognition tests, we had 9.0% of recognition improvement for the case 1), more than 75% for the case 2), and 11%~61.4% for the case 3) respectively, resulting that a MAP environmental adaptation is effective for both channel distorted and noise added speech recognition. But it turned out that duration information used as additive feature parameter did not played an important role in the tests.

• The Journal of the Acoustical Society of Korea
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• v.40 no.4
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• pp.359-369
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• 2021

In underwater communications, to compensate performance degradation induced from rapidly changing channel transfer characteristic, multi-band communication method which allocate the same data to different frequency bands is used. However, the multi-band configuration may have worse performance than the single-band one because performance degradation in a particular band affects the output from the entire bands. This problem can be solved through a receiving end that analyzes error rates of each band, sets threshold values and allocates lower weights to inferior bands. Therefore, this paper proposed a weighting algorithm based on estimated Bit Error Rate (BER) which analyzes reliability of received data based on the performance difference between demodulated and decoded data. Employing turbo codes with coding rate of 1/3, we evaluate the performance of the proposed weighted multi-band transmission model in real underwater environments based on optimal simulation parameters. Through the sea trial experiment, we confirmed error performance was improved by applying the proposed weighting algorithm.

• Journal of Korea Multimedia Society
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• v.11 no.3
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• pp.281-291
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• 2008

In this paper, we propose a new double talk detection algorithm which detects near end signals with less degradation, tracking echo path variation of echo canceler simultaneously. Our method makes use of a cross-correlation between channel input signals and estimated error signals and a normalized cross-correlation between microphone input signals and estimated error signals. By combing thresholds for these cross-correlations pertinently, this algorithm discriminates between variation of echo path and occurrence of double talk. These two cross-correlation are used to detect double talk periods, tracking echo path variation. During the detection period, adjustive adaptive filter is ceased to prevent the echo canceler from being disturbed by near end signals. Also, the echo canceler will still be kept on for tracking any variation in echo path. Through computer simulation results, it was confirmed that the proposed algorithm shows better performance, tracking echo path variation and detecting the double talk periods, than the Ye et. al's and the NLMS algorithms from ERLE viewpoint.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2
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• pp.30-37
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• 1994

Since the morden targets are of high speed and getting quiet in both active and passive mode, the necessities of developing advanced SONAR system capable of performing target motion analysis (TMA) and target classification are evident. In order to develop such a system, the scattering mechanism of complex bodies needs to be, some extent, fully understood and modeled. In this paper, MOving Spread Target(MOST) signal simulation model is presented and discussed. The model is based on the highlight distribution method, and simulates pulse elongation of spread target, doppler effect due to kinematics of the target as well as SONAR platform, and distribution target strength of each highlight point (HL) with directivity. The model can be used in developing and evaluating advanced SONAR system through system simulation, and can also be used in the development of target state estimation algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2397-2404
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• 2012

Underwater acoustic communication has multipath error because of reflection by sea-level and sea-bottom. The multipath of underwater channel causes signal distortion and error floor. In order to improve the performance, it is necessary to employ an iterative coding scheme. Among the iterative coding scheme, turbo codes and LDPC codes are dominant channel coding schemes in recent. This paper concluded that turbo coding scheme is optimal for underwater communications system in aspect to performance, coded word length, and equalizer combining. Also, decision directed phase recovery was used for correcting phase offset induced by multipath. Based on these algorithms, we confirmed the performance in the environment of oceanic experimentation.

• The Journal of the Acoustical Society of Korea
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• v.29 no.7
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• pp.468-474
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• 2010

The major factor that disturbs practical use of speech recognition is distortion by the ambient and channel noises. Generally, the ambient noise drops the performance and restricts places to use. DSR (Distributed Speech Recognition) based speech recognition also has this problem. Various noise cancelling algorithms are applied to solve this problem, but loss of spectrum and remaining noise by incorrect noise estimation at low SNR environments cause drop of recognition rate. This paper proposes methods for speech enhancement. This method uses MMSE-STSA for noise cancelling and ideal binary mask to compensate damaged spectrum. According to experiments at noisy environment (SNR 15 dB ~ 0 dB), the proposed methods showed better spectral results and recognition performance.