• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.1
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• pp.251-258
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• 2013

We studied it about ways to detect damage of a tire about randomness ultrasonic signal which occurs repeatedly while rub a tire of driving car and a road surface. The signal randomness is decreased through the preprocess of short-time energy calculation and the average value of coherence function is used by the normalization expression of the signal randomness. The process limit that can be decide on the dominant period of a signal using the coherence threshold is analyzed and the algorithm to decide the dominant period is proposed by setting up the -3dB threshold of the maximum value on the power spectrum.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.471-473
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• 2009

In this paper, we propose a double talk and noise robust acoustic echo canceler for hands-free telephony applications. The proposed system includes a double-talk detection method that detects the double-talk durations, which uses covariance between microphone input signa and estimated microphone input signal. And proposed adaptive algorithm for estimating acoustic echo path, uses normalized auto-covariance matrix of input signal with multiplication of residual error power and projection order of AP(affine projeciton) algorithm. It is confirmed that the proposed algorithm shows better performance from acoustic interference cancellation (AIC) viewpoint in double talk and noisy environments.

• Journal of the Institute of Convergence Signal Processing
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• v.3 no.1
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• pp.8-14
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• 2002

In this paper, we realized a real-time text-independent speaker recognition system using gaussian mixture model, and applied frame level likelihood normalization method which shows its effects in verification system. The system has three parts as front-end, training, recognition. In front-end part, cepstral mean normalization and silence removal method were applied to consider speaker's speaking variations. In training, gaussian mixture model was used for speaker's acoustic feature modeling, and maximum likelihood estimation was used for GMM parameter optimization. In recognition, likelihood score was calculated with speaker models and test data at frame level. As test sentences, we used text-independent sentences. ETRI 445 and KLE 452 database were used for training and test, and cepstrum coefficient and regressive coefficient were used as feature parameters. The experiment results show that the frame-level likelihood method's recognition result is higher than conventional method's, independently the number of registered speakers.

• Geophysics and Geophysical Exploration
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• v.20 no.2
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• pp.78-87
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• 2017

For efficient data processing, trace interpolation and regularization techniques should be antecedently applied to the seismic data which were irregularly sampled with missing traces. Among many interpolation techniques, MWNI (Minimum Weighted Norm Interpolation) technique is one of the most versatile techniques and widely used to regularize seismic data because of easy extension to the high-order module and low computational cost. However, since it is difficult to interpolate spatially aliased data using this technique, model-constrained MWNI was suggested to compensate for this problem. In this paper, conventional MWNI and model-constrained MWNI modules have been developed in order to analyze their performance using synthetic data and validate the applicability to the field data. The result by using model-constrained MWNI was better in spatially aliased data. In order to verify the applicability to the field data, interpolation and regularization were performed for two field data sets, respectively. Firstly, the seismic data acquired in Ulleung Basin gas hydrate field was interpolated. Even though the data has very chaotic feature and complex structure due to the chimney, the developed module showed fairly good interpolation result. Secondly, very irregularly sampled and widely missing seismic data was regularized and the connectivity of events was quite improved. According to these experiments, we can confirm that the developed module can successfully interpolate and regularize the irregularly sampled field data.

• Journal of Biomedical Engineering Research
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• v.19 no.4
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• pp.379-384
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• 1998

To measure precisely the blood velocity in the skin microcirculation, we have used time domain correlation (called Cross-Correlation) based on the processing of the backscattered RF signal obtained with a wideband echographic imaging transducer, although it is difficulties of adaptation of the pulsed wave system, because of the data processing in real time and the hardware problem. This dedicated technology based on a 20MHz echographic imaging system has been developed. We present how the experimental data, i.e. the backscattered RF signal, have to be analyzed. After RF lines realignment, stationary echo canceling procedure and correlation level control, a velocity profile has been obtained. In-vitro result show that velocity measurements as low as 0.1mm/sec attainable with a 80${\mu}m$ in axial resolution. We have also validated with in-vivo experimentation on the external ear of a rabbit using B-mode sector scanning image and M-mode image of a custom made 20MHz skin image system. The flow of the "auriculares caudales" vein, a microvessel of 600 m diameter, has been detected and studied. This technique will allow a more precise exploration of circulatory troubles in cutaneous pathologies.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.20 no.8
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• pp.701-713
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• 2009

This paper studies a method for reducing performance degradation due to losing sub-carrier orthogonality caused by power transient between physical channels in LTE uplink transmission. The pattern of inter-carrier interference(ICI) caused by power transient is different from what has been studied for doppler shift, in that its pattern occurs at front and rear sides of channels in each period of power transient. The reason of ICI's occurrence results from power difference between channels, power transient duration, multi-path channel delay spread, and numbers of sub-carrier. New criterion is proposed to find out number of taps of multi-tap equalizer enough to improve the ICI. The scheme is to determine the number of taps of multi-tap equalizer when a normalized interference or a normalized ICI is greater than a normalized noise. Simulation results show that the number of taps is flexibly adjusted according to SNR(Signal to Noise Ratio) of a received signal to improve Bit Error Rate(BER), while the complexity of the proposed scheme is reduced down to 88 percentage of the classical method.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.26 no.3
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• pp.317-325
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• 2015

In an interference environment, adaptive sidelobe blanking(adaptive SLB: ASB) algorithm effectively cancels the high-duty cycle jammer and blocks the sidelobe signals without the auxiliary antenna. The adaptive SLB for the linearly constrained minimum variance (LCMV) is proposed in this paper. In the proposed scheme, the interference covariance matrix is modified to satisfy the direction constraints of LCMV and the normalized output can be obtained to block sidelobe signals. As the LCMV can be represented as a generalized sidelobe canceller(GSC) form, which is the general framework of various adaptive beamforming(ABF) algorithms, the proposed adaptive SLB can be applied to various ABF methods. The performance of the proposed method is verified through simulation and analysis.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.2
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• pp.123-129
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• 2017

This paper proposes a new nonnegative matrix factorization (NMF) based direction-of-arrival (DOA) estimation method for multiple sound sources using a dual microphone array. First of all, sound signals coming from the dual microphone array are segmented into consecutive analysis frames, and a steered-response power phase transform (SRP-PHAT) beamformer is applied to each frame so that stereo signals of each frame are represented in a time-direction domain. The time-direction outputs of SRP-PHAT are stored for a pre-defined number of frames, which is referred to as a time-direction block. Next, In order to estimate DOAs robust to noise, each time-direction block is normalized along the time by using a block subtraction technique. After that, an unsupervised NMF method is applied to the normalized time-direction block in order to cluster the directions of each sound source in a multiple sound source environments. In particular, the activation and basis matrices are used to estimate the number of sound sources and their DOAs, respectively. The DOA estimation performance of the proposed method is evaluated by measuring a mean absolute error (MAE) and the standard deviation of errors between the oracle and estimated DOAs under a three source condition, where the sources are located in [$-35{\circ}$, 5m], [$12{\circ}$, 4m], and [$38{\circ}$, 4.m] from the dual microphone array. It is shown from the experiment that the proposed method could relatively reduce MAE by 56.83%, compared to a conventional SRP-PHAT based DOA estimation method.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.17 no.3
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• pp.450-456
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• 2016

STDR (sequence time domain reflectometry) to detect a cable fault using a pseudo noise sequence as a reference signal, and time correlation analysis between the reference signal and reflection signal is robust to noisy environments and can detect intermittent faults including open faults and short circuits. On the other hand, if the distance of the fault location is far away or the fault type is a soft fault, attenuation of the reflected signal becomes larger; hence the correlation coefficient in the STDR becomes smaller, which makes fault detection difficult and the measurement error larger. In addition, automation of the fault location by detection of phase and peak value becomes difficult. Therefore, to improve the cable fault detection of a conventional STDR, this paper proposes the algorithm in that the peak value of the correlation coefficient of the reference signal is detected, and a peak value of the correlation coefficient of the reflected signal is then detected after removing the reference signal. The performance of the proposed method was validated experimentally in low-voltage power cables. The performance evaluation showed that the proposed method can identify whether a fault occurred more accurately and can track the fault locations better than conventional STDR despite the signal attenuation. In addition, there was no error of an automatic fault type and its location by the detection of the phase and peak value through the elimination of the reference signal and normalization of the correlation coefficient.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2000.11b
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• pp.63-67
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• 2000

디지털 영상 저장 과정에서 일어나는 문제점은 영상 저장부 센서계의 한계로 나타낼 수 있다. 센서계의 충분하지 못한 집적도는 물리적으로 피할 수 없는 현상이다. 이러한 현상을 디지털 신호처리 기술을 적용하여 극복할 수 있다. 센서계의 한계로 인한 문제는 디지털 영상의 가장 큰 문제중의 하나이며, 이러한 한계를 극복하는 고해상도 영상 복원 방법들은 많은 학자들에 의해 제안되어 왔다. 본 논문에서는, 기존의 고해상도 영상 복원 방법들과는 달리 원영상의 공간적 고주파 성분의 특성을 분석과, 주어진 저해상도 영상들의 부화소 단위 움직임 추정 오류에 대한 분석을 통해 영상 복원과정에 이러한 분석들의 결과를 반영한다. 위에서 언급한 추정 오류는 우리에게 하나의 잡음 형태로 나타날 수 있다. 이 잡음은 추정이 이루어지는 축에 따라 그 양이 다르게 나타나게 되고, 이러한 현상은 목적이 되는 영상의 공간적 고주파 성분의 분포와 밀접한 관련이 있다. 우리는 복원 과정에 기존의 영상복원 방법중의 하나인 정규화 방법을 도입한다. 위에서 분석된 현상을 이 복원 과정에 반영하여 기존의 고해상도 영상 복원 방법보다 향상된 결과를 얻을 수 있었다. 결론적으로, 제안하는 알고리즘은 부화소 단위 움직임 추정 오류의 분석 결과를 반영하므로 이러한 추정 오류에 강한 알고리즘이다.