• The Journal of the Acoustical Society of Korea
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• 제25권4E호
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• pp.148-153
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• 2006

A new normalized MBD algorithm is presented for nonstationary convolutive mixtures and its properties/modifications are discussed in details. The proposed algorithm normalizes the signal spectrum in the frequency domain to provide faster stable convergence and improved separation without whitening effect. Modifications such as nonholonomic constraints and off-diagonal learning to the proposed algorithm are also discussed. Simulation results using a real-world recording confirm superior performanceof the proposed algorithm and its usefulness in real world applications.

• 대한전기학회논문지:시스템및제어부문D
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• 제54권3호
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• pp.198-205
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• 2005

In this paper, we proposed a new algorithm for the separation of fetal ECG from single channel abdominal ECG. The algorithm consists of a stage of demixing vector calculation for initial signal and a stage of fetal beat detection for the rest of signal. The demixing vector was obtained by applying independent component analysis technique to projected signals into time-frequency domain. For the test of this algorithm, simulation signals, De Lathauwer's data and some measured data, which was acquired from 8 healthy volunteers whose pregnant periods ranged from 22 weeks to 35 weeks and whose ages from 27 to 37, were used. For each data, the accuracy of fetal beat detection was $100\%$ and with the location of fetal beats, fetal heart rate variability and morphology could be offered. In conclusion, this proposed algorithm showed the possibility of fetal beat separation with a single channel abdominal ECG and it might be adopted to a fetal health monitoring system, by which a single channel abdominal ECG is acquired.

• 대한음성학회지:말소리
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• 제48호
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• pp.117-126
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• 2003

source of speech signal is actually composed of combination of periodic and aperiodic components, although it is often modeled to either one of those. In the paper an experiment which can separate periodic and aperiodic components from speech source. Linear predictive residual signal was used as a approximated vocal source the original speech to obtain the estimated aperiodic part. Iterative extrapolation method was used to compute the aperiodic part.

• 대한조선학회 특별논문집
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• 대한조선학회 2013년도 특별논문집
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• pp.47-49
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• 2013

According to the new Singapore Authority Rule(Sn.1/Cir293) which has taken effect from $1^{st}$ July, 2011 at 000hrs, all vessel crossing the Traffic Separation Scheme(TSS) and precautionary areas in Singapore Strait are recommended to display the night signals consisting of 3 all-around green lights in a vertical line. So, this paper presents methods for design of SINGAPORE STRAIT TRANSITION NIGHT SIGNAL.

• 융합신호처리학회논문지
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• 제15권3호
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• pp.91-96
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• 2014

두 개의 입력소자에 의한 음원방향 및 분리방법으로서는 연산량이 적고, 음원분리 성능이 높은 주파수 양귀 모델(Frequency Domain Binaural Model : FDBM)이 있다. FDBM은 주파수 영역에서 양귀간 위상차(Interaural Phase Difference : IPD) 및 양귀간 레벨차(Interaural Level Difference : ILD)를 구하여 음향신호가 오는 방향과 음원의 분리처리를 한다. 그러나 실제 환경에서는 반사음의 문제가 되고 있다. 이러한 반사음에 의한 영향을 줄이기 위하여 선행음 효과에 의한 직접음의 음상정위를 모의하여 초기 도착음을 검출하고 직접음이 오는 방향과 음원분리 방법을 제시하였다. 제시한 방법을 이용하여 음원방향 추정 및 분리에 대한 성능을 시뮬레이션으로 검토하였다. 그 결과, 방향추정은 음원이 오는 방향에서 ${\pm}10%$의 범위로 집중되어 음원의 방향과 가까운 값으로 추정되었다, 반사음이 존재하는 경우의 음원분리는 기존의 FDBM에 비하여 코히런스(Coherence), 음성품질 지각평가 PESQ(Perceptual Evaluation of Speech Quality : PESQ)가 높고, 정면에서의 지향특성 감쇠량이 작아 분리의 정도가 개선됨을 나타내었다. 그러나 반사음이 존재하지 않는 경우는 분리 정도가 낮았다.

• 대한음성학회지:말소리
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• 제65호
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• pp.153-168
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• 2008

We propose a new method to apply shape analysis techniques to a computational auditory scene analysis (CASA)-based speech separation system. The conventional CASA-based speech separation system extracts speech signals from a mixture of speech and noise signals. In the proposed method, we complement the missing speech signals by applying the shape analysis techniques such as labelling and distance function. In the speech separation experiment, the proposed method improves signal-to-noise ratio by 6.6 dB. When the proposed method is used as a front-end of speech recognizers, it improves recognition accuracy by 22% for the speech-shaped stationary noise condition and 7.2% for the two-talker noise condition at the target-to-masker ratio than or equal to -3 dB.

• 한국멀티미디어학회논문지
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• 제8권6호
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• pp.753-760
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• 2005

This paper introduces a variable step-size adaptive algorithm for blind source separation. From the frequency characteristics of mixed input signals, we need to adjust the convergence speed regularly in each frequency bin. This algorithm varies a step-size according to the magnitude of input at each frequency bin. This guarantee of the regular convergence in each frequency bin would become more efficient in separation performances than conventional fixed step-size FDICA. Computer simulation results show the improvement of about 5 dB in signal to interference ratio (SIR) and the better separation quality.

• 대한음성학회지:말소리
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• 제67호
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• pp.195-216
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• 2008

This paper addresses the problem of single-channel speech separation to extract the speech signal uttered by the speaker of interest from a mixture of speech signals. We propose to apply time-frequency smoothing to the existing statistical single-channel speech separation algorithms: The soft mask and the minimum-mean-square-error (MMSE) algorithms. In the proposed method, we use the two smoothing later. One is the uniform mask filter whose filter length is uniform at the time-Sequency domain, and the other is the met-scale filter whose filter length is met-scaled at the time domain. In our speech separation experiments, the uniform mask filter improves speaker-to-interference ratio (SIR) by 2.1dB and 1dB for the soft mask algorithm and the MMSE algorithm, respectively, whereas the mel-scale filter achieves 1.1dB and 0.8dB for the same algorithms.

• 한국음향학회지
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• 제38권6호
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• pp.697-702
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• 2019

방송 콘텐츠의 혼합 신호에서 배경음악 신호를 분리하는 확장된 합성곱을 이용한 멀티 밴드 멀티 스케일 DenseNet을 제안한다. 확장된 합성곱은 스펙트로그램의 다양한 스케일 문맥 정보를 학습하기 용이하도록 한다. 컴퓨터 모의실험 결과, 제안한 구조는 신호대잡음비(Signal to Noise Ratio, SNR) 0 dB, -10 dB의 환경에서 각각 0.15 dB, 0.27 dB의 신호대왜곡비(Signal to Distortion Ratio, SDR)를 개선하였다.

• 로봇학회논문지
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• 제11권2호
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• pp.100-107
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• 2016

To realize a robot hand interacting like a human hand, there are many tactile sensors sensing normal force, shear force, torque, shape, roughness and temperature. This sensing signal is essential to manipulate object accurately with robot hand. In particular, slip sensors make manipulation more accurate and breakless to object. Up to now several slip sensors were developed and applied to robot hand. Many of them used complicate algorithm and signal processing with vibration data. In this paper, we developed novel principle slip sensor using separation layer. These two layers are moved from each other when slip occur. Developed sensor can sense slip signal by measuring this relative displacement between two layers. Also our principle makes slip signal decoupled from normal force and shear force without other sensors. The sensor was fabricated using the NBR(acrylo-nitrile butadiene rubber) and the Ecoflex as substrate and a paper as dielectric. To verify our sensor, slip experiment and normal force decoupling test were conducted.