• The Journal of the Acoustical Society of Korea
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• v.34 no.2
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• pp.130-137
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• 2015

In this paper, we propose a new Modified Filtered-x Affine Projection Sign Algorithm(MFxAPSA) to improve the convergence speed of the conventional MFxAPSA which has been proposed for active control of impulsive noise. Under the impulsive noise environment, the adaptive algorithms based on the second order moment such as the Filtered-x Least Mean Square(FxLMS) show slow convergence speed or diverge because the noise source tends to have infinite variance. The MFxAPSA is the algorithm derived by applying the Affine Projection Sign Algorithm(APSA) to active noise control. The APSA has an advantage that it does not need the calculation for the inverse matrix, so it may be suitable for the active noise control that requires low computational burden. The proposed MFxAPSA also has APSA's advantage and furthermore, better performance than the conventional MFxAPSA. We carried out a performance comparison of the proposed MFxAPSA with the conventional MFxAPSA. It is shown that the proposed MFxAPSA has the faster convergence speed than the conventional MFxAPSA.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.9A
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• pp.1332-1339
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• 2000

This paper presents the architecture and design of a high speed asymmetric data transmission baseband MODEM ASIC chip for CATV networks. The implemented MODEM chip supports the physical layer of the DOCSIS(Data Over Cable Service Interface Specification) standard in MCNS(Multimedia Cable Network System) The chip consists of a QPSK/16-QAM transmitter and a 64/256-QAM receiver which contain a symbol timing recovery circuit, a carrier recovery circuit, a blind equalizer using MMA and LMS algorithms. The chip can support data rates of 64Mbps at 256 QAM and 48Mbps at 64-QAM and can provide symbol rates up to 8MBaud. This symbol rate is faster than existing QAM receivers. We have performed logic synthesis using the $0.35\mu\textrm{m}$ standard cell library. The total number of gates is about 290,000 and the implemented chip is being fabricated and will be delivered soon.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.2
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• pp.227-236
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• 2001

This paper proposes a frequency-domain partially adaptive algorithm, called a censoring algorithm, to reduce the computational complexity of the frequency domain adaptive array. The proposed censoring algorithm determines the existence of interferences in the frequency-domain at each frequency bin using a constant false alarm rate (CFAR) processor. The censoring algorithm adapts only those parts of the weights that correspond to the frequency bins expected to contain interferences. The censoring algorithm is also expanded to overcome the signal cancellation phenomenon caused by smart jammers. Accordingly, a censoring spatial smoothing, which combines the censoring algorithm with spatial smoothing, is proposed. Simulation results show that the proposed algorithms are effective in removing interferences with only part of the computational complexity of conventional algorithms yet with the same level of performance.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1132-1136
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• 2005

In this paper, adaptive equalizer receiver for a turbo code direct sequence code division multiple access (DSCDMA) by using least mean square (LMS) adaptive algorithm is presented. The proposed adaptive equalizer is using soft output of decision feedback adaptive equalizer (DFE) to examines the output of the equalizer and the Log- maximum a posteriori (Log-MAP) algorithm for the turbo decoding process of the system. The objective of the proposed equalizer is to minimize the bit error rate (BER) of the data due to the disturbances of noise and intersymbol interference (ISI)phenomenon on the channel of the DS-CDMA digital communication system. The computer program simulation results shown that the proposed soft output decision feedback adaptive equalizer provides a good BER than the others one such as conventional adaptive equalizer, infinite impulse response adaptive equalizer.

• The Journal of the Acoustical Society of Korea
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• v.7 no.4
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• pp.77-85
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• 1988

Many kinds of conditioning monitoring technique have been studied, so this study has inverstigated the possibility of checking the trend in the fault diagnosis of ball bearing, one of the elements of rotating machine, by applying the cepstral analyisis method using the adaptive noise cancelling (ANC) method. And computer simulation is conducted in order to verify the usefulness of ANC. The optimal adaptation gain in adaptive filter is estimated, the performance of ANC according to the change of the signal to noise ratio and convergence of least mean square algorithm is considered by simulation. It is verified that cepstral analysis using ANC method is more effective than the conventional cepstral analysis method in bearing fault diagnosis.

• The Journal of the Acoustical Society of Korea
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• v.37 no.4
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• pp.206-214
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• 2018

This paper proposes an impact point estimation algorithm of the rifle using microphone sensors. The proposed algorithm resolves the time synchronization problem by expanding the existing ToA (Time of Arrival) method to TDoA (Time Difference of Arrival) method and verifies the performance of the algorithm through the actual shooting experiments. By comparing analysis of the actual and the estimated impact points by the algorithm, it is confirmed that the proposed algorithm has excellent performance by estimating the impact point accurately within the tolerance range.

• Journal of Korea Multimedia Society
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• v.8 no.9
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• pp.1216-1226
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• 2005

In this paper, a preventive transmission rate decision algorithm, called PTRD, is proposed for the transmission of live video over networks with dynamic bandwidth allocation capability. Frame analyzer predicts the bit-rates of future frames before encoding by analyzing the source information such as spatial variances and the degree of scene changes. By using the predicted bit-rates, transmission rate bounds are derived from the constraints of encoder and decoder buffers. To resolve the problem of renegotiation cost increment due to frequent renegotiations, the PTRD algorithm is presented to decide transmission rates considering the elapsed time after the recent renegotiation and the perceived video quality. From the simulation results, compared to the normalized LMS based method, PTRD is shown to achieve high channel utilization with low renegotiation cost and no delay violation.

• Journal of KSNVE
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• v.10 no.6
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• pp.1048-1058
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• 2000

Thipaper presents a methodology to suppress the vibration of thin rectangular plate clamped all edges using piezo-ceramic material as actuators and sensors. Dynamic characteristics of the structure bonded with distributed actuators/sensors are identified by the Multi-Input Multi-Output (MIMO) frequency domain modeling technique based on the experimental data. Hybrid control scheme is adopted and feedback controller is designed by LQG(Linear Quadratic Gaussian). Feedforward controller is adapted by multiple filtered -$x$ LMS(least mean square) algorithm. Experiment result demonstrates the effective reduction of the vibration label for both the transient and persistent external disturbances.

• Journal of the Korean Institute of Intelligent Systems
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• v.22 no.5
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• pp.663-668
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• 2012

A TPMS(Tire Pressure Monitoring System) is a wireless communication system designed to monitor the pressure and temperature of pneumatic tires of a vehicle. In order to provide the aid in protecting a driver, this system reports tire pressure information to the driver of the vehicle. Since the wireless communication technique should be employed to transmit the TPMS data from each tire to signal processing unit in the vehicle, it suffers from interference signals from external electrical or electronics equipments. In this paper, we propose the TPMS interference cancellation technique based on GSC(Generalized Sidelobe Canceler), which does not have only the excellent performance like MVDR(Minimum-Variance-Distortionless-Response) but also has the low computational complexity comparing with MVDR. The performance of interference suppression is conformed by computer simulation examples.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.12C
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• pp.1262-1267
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• 2005

Elliptical basis function (EBF) neural network works stably under high-level background noise environment and makes the nonlinear processing possible. It can be adapted real time VAD with simple design. This paper introduces VAD implementation using EBF and the experimental results show that EBF VAD outperforms G729 Annex B and RBF neural networks. The best error rates achieved by the EBF networks were improved more than $70\%$ in speech and $50\%$ in silence while that achieved by G.729 Annex B and RBF networks respectively.