• The Journal of the Acoustical Society of Korea
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• v.30 no.6
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• pp.353-359
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• 2011

This paper proposes a vocal enhancement technique for improving the performance of vocal pitch detection in polyphonic music signal. The proposed vocal enhancement technique predicts an accompaniment signal from the input signal and generates an accompaniment replica signal according to the vocal power. Then, it removes the accompaniment replica signal from the input signal, resulting in a vocal-enhanced signal. The performance of the proposed method was measured by applying the same vocal pitch extraction method to the original and the vocal-enhanced signal, and the vocal pitch detection accuracy was increased by 7.1 % point in average.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.8
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• pp.95-104
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• 2016

Subspace-based direction-of-arrival (DOA) estimation algorithms have a difficulty in dealing with coherent signals caused by multi-path environment. As one of attempts to solve this problem, a spatial differencing method is known to be useful for not only estimating DOAs of the coherent signals but also improving the number of resolvable wavefronts even more than the number of antenna elements. However, since the conventional spatial differencing method uses only the partial statistics of the observed data, this method suffers from the performance degradation in estimation accuracy caused by the residual correlation between the uncorrelated signals. To cope with this problem, in this paper, a generalized spatial differencing method is proposed. Unlike the conventional method, the proposed method utilizes the entire statistics of the received signals. Therefore, the additional performance enhancement in both estimation accuracy and the number of resolvable wavefronts can be achieved. The performance analyses with computer simulations show that the proposed method outperforms the conventional method in terms of the estimation accuracy and the number of resolvable wavefronts.

• The Journal of the Acoustical Society of Korea
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• v.42 no.4
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• pp.357-363
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• 2023

In underwater signal processing, separating individual signals from mixed signals has long been a challenge due to low signal quality. The common method using Short-time Fourier transform for spectrogram analysis has faced criticism for its complex parameter optimization and loss of phase data. We propose a Triple-path Recurrent Neural Network, based on the Dual-path Recurrent Neural Network's success in long time series signal processing, to handle three-dimensional tensors from multi-channel sensor input signals. By dividing input signals into short chunks and creating a 3D tensor, the method accounts for relationships within and between chunks and channels, enabling local and global feature learning. The proposed technique demonstrates improved Root Mean Square Error and Scale Invariant Signal to Noise Ratio compared to the existing method.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.4
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• pp.330-336
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• 2016

In this paper, we studied for modified bartlett method to estimate desired information signal. Constrained length of bartlett method is assigned as one, and estimate desired information signal to compensate for delay time. Modified bartlett method is an optimum direction-of-arrival (DoA) estimation algorithm to apply delay time compensation to update optimum weight. The optimum weight is used linear constrained minimum variance method(LCMV). Through simulation, we are comparative analysis proposed algorithm and general Bartlett and MUSIC method. In desired signal estimation, condition simulation is an array antenna element numbers 6 or 9 and desired information signals number 3. We show the superior performance of the proposed algorithm relative to the existing method in estimation of desired information signal.

• The Journal of the Acoustical Society of Korea
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• v.35 no.2
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• pp.110-117
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• 2016

Passive sonar systems detect and classify the target by analyzing the radiated noises from vessels. If multiple noise sources exist within the sonar detection range, it gets difficult to classify each noise source because mixture of noise sources are observed. To overcome this problem, a beamforming technique is used to separate noise sources spatially though it has various limitations. In this paper, we propose a new method that uses a FDICA (Frequency Domain Independent Component Analysis) to separate noise sources from the mixture. For experiments, each noise source signal was synthesized by considering the features such as machinery tonal components and propeller tonal components. And the results of before and after separation were compared by using LOFAR (Low Frequency Analysis and Recording), DEMON (Detection Envelope Modulation On Noise) analysis.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.20 no.3
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• pp.23-28
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• 2019

A track type underwater construction robot(URI-R) which can trench on seabed is being developed by Korea Institute of Ocean Science & Technology. During the underwater trenching work, the robot is exposed high intensive noise and vibration so the underwater localization signal may not be obtained properly by the acoustic tracking system. Therefore it is necessary to research about continuous localization even though the measured position signal comes in intermittently. In this paper, the experiments were carried out on land to simulated the underwater operating environment characteristics. To estimate its position, inertial navigation system and global navigation satellite system are used. The effects of the period variation while localizing is investigated by the experiments, and the application for URI-R is proposed.

• Journal of IKEEE
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• v.16 no.1
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• pp.20-27
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• 2012

In order to overcome jamming, a hysteresis tunable monolithic comparator circuit based on a 0.35 ${\mu}m$ CMOS process is suggested, designed, fabricated, measured and analyzed in this paper. To tune the threshold voltage of the hysteresis in the comparator circuit, two external digital bits are used with supply voltage of 3.3V. An improved variable hysteresis comparator circuit controlled by serial digital bits is suggested, designed and simulated to overcome jamming in modern warfare.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.2
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• pp.65-72
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• 2006

In digital signal processing, the polyphase decomposition of signal has been often used in the implementation of multirate system. Especially, in the design of digital filter and so forth the method in very useful to improve the performance of various algorithms because it provides the multi-channel for paralled processing. Generally, the polyphase-decomposed signals tend to expand the frequency band by including more high frequencies than original signal from decimation for down sampling. This property brings about the significant limitation in the structure or the performance of digital polyphase signal processing system. In this paper we theoretically propose a perfect band compression and reconstruction method for polyphase component signals, then experimentally show its effectiveness through Matlab simulation.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2004.07b
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• pp.1195-1198
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• 2004

본 논문에서는 레이더(Radar)나 소나(Sonar) 시스템 등에서 중요하게 여겨왔던 신호의 방위각 추정기법들은 신호원의 수를 정확히 알아야 한다는 문제점이 있었다. 이러한 문제점들을 최근에는 빠른 수렴속도를 가지는 칼만 필터를 사용하여 신호원의 정보 없이 잡음 부공간을 성공적으로 추정하는 방법을 제안하게 되었는데 이 방법은 중간대 이상의 신호대 잡음비(SNR)가 유지되는 경우 신호의 정보 얼이 DOA를 추정할 수 있는 것으로 평가되어왔다. 그러나 이러한 알고리즘은 임펄스성 잡음이 섞인다면 성능저하를 가져을 수 있다. 따라서 본 논문에서는 칼만 필터를 바탕으로 한 MUSIC 알고리즘에 시변메모리인자와 후버 영향 함수를 도입하여 임펄스 잡음에 강인한 DOA 추정알고리즘을 제안한다.

• Proceedings of the KSLP Conference
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• 1994.06a
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• pp.128-137
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• 1994

신호처리의 목적은 신호를 변형하여 우리가 원하는 형태로 만드는 것으로 신호를 변환시키는 장치 즉 시스템이 신호에 응답하여 다른 형태의 신호를 만들어 내는 것을 신호처리라 한다. 현재는 음성신호 처리시에 대부분 입력시호인 아날로그 신호(Analog Signal)를 표본화(Sampling)하고 양자화(Quantizing)하여 디지털 신호(Digital Signal)로 변환한 후 필요한 신호처리를 수행한다. 디지털 신호를 처리하므로써 정확성, 신뢰성, 처리속도를 증가시키게 되고 전자시스템(Electronic System)의 크기를 줄일 수가 있다. (중략)