• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.412-421
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• 2018

The Line array sonar consisting of several hydrophones increases array gain and improves the performance for detecting the direction of the target compared to single hydrophone. However, line array sonar produces the bearing error that makes it difficult to determine the bearing of incoming source signal due to the relation between bearing angle of target and vertical angle of multipath signals. Vertical angles of multipath are varied with the geometry of receiver and target and various underwater environments, therefore it is necessary to consider the bearing error to estimate accurately the bearing of the target. In this study, acoustic modelling was performed to understand the effect of multipath signals on the target signal. The errors of bearing angle estimated from the bottom bounced signals are calculated with several environment. In addition, the expected bearing line, as a function of source-receiver range, compensated for the bearing error is predicted from the estimated bearing angle.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.1
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• pp.11-17
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• 2009

This paper deals with the estimation performance DoA (Direction of Arrival) using the 3D adaptive array antenna in W-CDMA signal. For this, we proposed the 5 types of 3D array antenna, and appied for the MUSIC in order to the estimation of DoA algorithm for that antenna, commonly, then the DoA estimation error was used for the estimation performance. In the mobile communication and radio positioning service, performing the spatial filtering after the DoA estimation in array antenna, the quality of receiving signal can improve by the nulling or minimization of interfering signal which is from the undesired direction and the forming of beam which is from the desired direction. The result of DoA estimation and the DoA estimation error by varying the signal to noise ration and the number of interfering signal and power of each type antenna was calculated by computer simulation. As a result of simulation, the other propose antenna has good performance excluding the stack 3D antenna in DoA estimation and the Curved-B type antenna has more superior performance when increasing the number of interfering signal and power in the DoA estimation error.

• The Journal of the Acoustical Society of Korea
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• v.10 no.1
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• pp.20-29
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• 1991

A fundamental assumption in conventional linear predictive coding (LPC) analysis procedure is that the input to an all-pole vocal tract filter is white process. In the case of periodic inputs, however, a pitch bias error is introduced into the conventional LP coefficient. Multi-pulse (MP) LP analysis can reduce this bias, provided that an estimate of the excitation is available. Since the prediction error of conventional LP analysis can be modeled as the sum of an MP excitation sequence and a random noise sequence, we can view extracting MP sequences from the prediction error as a classical detection and estimation problem. In this paper, we propose an algorithm in which the locations and amplitudes of the MP sequences are first obtained by applying a likelihood ratio test (LRT) to the prediction error, and LP coefficients free of pitch bias are then obtained from the MP sequences. To verify the performance enhancement, we iterate the above procedure with adaptive threshold at each step.

• Journal of the Institute of Convergence Signal Processing
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• v.8 no.3
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• pp.171-177
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• 2007

In this paper, we have designed and implemented an optical signal processor in order to use in a fiber-optic current CT for electric equipments where its properties were discussed. The fabricated optical signal processor is used to reduce a measurement current error that induced by the effects of intensity variation in the optical output signal due to losses coming from optical components or polarization variation in a PFOCS. Also, the optical signal processor was fabricated in compact/lightweight with unification of opto-electronic transducer part, analog signal process part, and real-time measurement part consisted of a level shift and ${\mu}-processor$. The experiment of optical signal processor has been performed in the range of $0{\sim}7,500A$ using the PFOCS made all fiber-optic components. As a result of experiment, the linearity error of measurement current is less than 1.7% and its average error is less than 0.3% in the range of $1,000A{\sim}7,000A$.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.3
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• pp.89-95
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• 2016

In this paper, we propose a dual-mode blind equalization algorithm that two of the blind equalization algorithm using the size of the decision-directed error signal is automatically switched. The proposed algorithm has a faster convergence speed due to operation of the MSAGF-SMMA with large fixed step-size mainly in the initial equalization. After the equalization has been made to some extent, the proposed algorithm has a smaller residual error in the steady- state by operation of the MSAGF-SMMA with a variable step-size mainly. The variable step-size is determined by multiplying the size of the decision-directed error signal of a fixed step-size. In this paper, we analyze the performance of the proposed algorithm. The computer simulation results demonstrate that the proposed algorithm has a significantly improved performance in terms of a residual inter-symbol interference and residual error in the steady-state compared with the MMA, SMMA, and MSAGF-SMMA.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1B
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• pp.89-98
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• 2000

This paper presents an active noise cancelling model using wavelet transform and subband filter banks based on adaptive filter. The analysis filter banks decompose input and error signals into QMF filter banks of lowpass and highpass bands. Each filter bank uses wavelet filter with dyadic tree structure. The decomposed input and error signals are iterated by adaptive filter coefficients of each subband using filtered-X LMS algorithm. The synthesis filter banks make output signal of wideband with perfect reconstruction to prepare adaptive filter output signals of each subband. The analysis and synthesis niter hants use conjugate quadrature filters for Pefect reconstruction. Also, The delayed LMS algorithm model for on-line identification of error path transfer characteristics is used gain and acoustic time delay factors. The proposed adaptive active noise cancelling modelis suggested by system retaining the computational and convergence speed advantage using wavelet subband filter banks.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.21 no.1
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• pp.55-60
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• 2021

This paper relates with the performance evaluation of QE-MMA (Quantized Error-MMA) adaptive equalization algorithm based on the stepsize and quantizer bit number in order to reduce the intersymbol interference due to nonlinear distortion occurred in the time dispersive channel. The QE-MMA was proposed using the power-of-two arithmetic for the H/W implementation easiness substitutes the multiplication and addition into the shift and addition in the tap coefficient updates process that modifies the SE-MMA which use the high-order statistics of transmitted signal and sign of error signal. But it has different adaptive equalization performance by the step size and quantizer bit number for obtain the sign of error in the generation of error signal in QE-MMA, and it was confirmed by computer simulation. As a simulation, it was confirmed that the convergence speed for reaching steady state depend on stepsize and the residual quantities after steady state depend on the quantizer bit number in the QE-MMA adaptive equalization algorithm performance.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.143-146
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• 1999

특정 플랜트내에 원하지 않는 음이나 진동신호를 능동적으로 제어하기 위해 LMS의 Filtered reference와 Filtered error 방법으로 피드포워드 적응제어 시스템을 사용할 때 오차신호의 수렴특성을 플랜트의 추정 임펄스응답의 정확도로써 나타낼 수 있는 식을 제안하고 수치 시뮬레이션과 실험으로 그 유효성에 대해 기술한다. 플랜트내 음향전달계의 추정 임펄스응답의 안정성 평가식은 이미 W. Ren과 P. R. Kumar에 의해 발표되었으나[1], 그 평가방법은 플랜트내의 실제(True) 임펄스응답을 필요로하고있어 수치 시뮬레이션에서는 가능한 방법이지만 임펄스응답의 측정이 불가능한 실제의 적응제어 시스템에 적용하는 것은 곤란하다. 따라서, 플랜트의 추정 임펄스응답을 크로스스펙트럼법(Cross-Spectrum)으로 추정하여 그 임펄스응답을 피드포워드 적응제어 시스템에 적용했을 때 안정성 여부를 평가하고, 또한 오차신호의 수렴특성을 Filtered reference와 Filtered error 방법에서 각각 확인하였다. 수치 시뮬레이션 결과와 실험 결과가 일치함을 확인하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.05a
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• pp.78-81
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• 1999

The position and time errors of a conventional L1-band GPS receiver (1575.42MHz) are known to be about 100 m and 70 ns, respectively. These errors are mainly due to the propagation delay of GPS satellite signals through ionosphere. Various L1/L2 dual-band GPS receivers are normally used to compensate for those position and time errors by detecting an accurate propagation delay. These receivers detect the propagation delay difference between the L1 and L2 signals based on the fact that the propagation delay through ionosphere is dependent on frequency and, from which, calculate an accurate propagation delay of the GPS signals through ionosphere. In this paper, we analyzed the architecture of a L1/L2 dual-band CPS receiver by high-level simulations with Synopsys's COSSAP Tool.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1325-1331
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• 1996

In this paper, a new current-memory circuit is proposed that reduces the clock feedthrough(CFT) error voltage causing total harmonic distortion(THD) increment in switched-current(SI) systems. Using PMOS transistor in CMOS complementary, the proposed one reduces output distortion current due to the CFT errorvoltage. A proposed current-memory is designed using a 1.2.mu.m CMOS process anda 1MHz sinusoidal signal having a 68.mu.A amplitude current is applied as input (sampling frequency:20MHz). It hasbeen shown from the simulation that the output distortion current effected by the CFT error voltage is reduced by approximately 10 times the error voltage of conventional one, THD is -57dB in case ofappling 1kHz frequency input signalwith 0.5 peak signal-to-bias current ratio.