• 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.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.11-20
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• 1997

The passive noise control techniques used until now cancel the noise in terms of the characteristics of materials, which increase the mass and the dimension and have a limit that is effective only to attenuate the high frequency components of the noise. But the active noise control techniques developed in recent years have merits that they decrease the mass and the dimension and are effective to attenuating the low frequency noises. In this paper, the real-time active noise control system attenuating the engine booming noise in a car using the digital signal processing(DSP) techniques in terms of the principle of active noise control. The multiple-error filtered-x LMS(Least-Mean Square) algorithm is used as the adaptive algorithm for active noise control and is implemented using the DSP processor Motorola DSP56001 as a controller. According to the result that the experiments are performed for the engine as the RPM changes in a car, the noise attenuating performances are achieved in an overall car interior and is verified to be 20 dB higher for pure-tone and globally, 15 dB.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.25 no.9
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• pp.628-636
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• 2015

In this study, theoretical and experimental analyses were performed for creating and moving the zone of quiet(ZoQ) to the ear location of a sitter by using active sound control technique. As the ZoQ is actively created at the location of the error microphone basically with an active sound control system using an algorithm such as the filtered-x least mean square(FxLMS), the virtual microphone control(VMC) method was considered to move the location of the ZoQ to around the sitter`s ear. A chair system with microphones and loudspeakers on both sides was manufactured for the experiment and thus an active headrest against the swept narrowband noise as the primary noise was implemented with a real-time controller in which the VMC algorithm was embedded. After the control experiment with and without the VMC method, the location variation of the ZoQ by analyzing the error signals measured by the error and the virtual microphones. Therefore, it is observed that the FxLMS with the VMC technique can provide the re-location of the ZoQ from the error microphone location to the virtual microphone location. Also it is found that the amount of the attenuation difference between the two locations was small.

• Smart Structures and Systems
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• v.25 no.5
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• pp.559-567
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• 2020

Vibration at the tip of various flexible manipulators may affect their operation accuracy and work efficiency. To suppress such vibrations, the feasibility of using MFC actuators and sensors is investigated in this paper. Considering the convergence of the famous filtered-x least mean square (FXLMS) algorithm could not be guaranteed while it is employed for vibration suppression of plants with varying secondary path, this paper proposes a new multiple model switching adaptive control algorithm to implement the real time active vibration suppression tests with a new multiple switching strategy. The new switching strategy is based on a cost function with reconstructed error signal and disturbance signal instead of the error signal from the error sensor. And from a robustness perspective, a new variable step-size sign algorithm (VSSA) based FXLMS algorithm is proposed to improve the convergence rate. A cantilever beam with varying tip mass is employed as flexible manipulator model. MFC layers are attached on both sides of it as sensors and actuators. A co-simulation platform was built using ADAMS and MATLAB to test the feasibility of the proposed algorithms. And an experimental platform was constructed to verify the effectiveness of MFC actuators and sensors and the real-time vibration control performance. Simulation and experiment results show that the proposed FXLMS algorithm based multiple model adaptive control approach has good convergence performance under varying load conditions for the flexible cantilever beam, and the proposed FX-VSSA-LMS algorithm based multiple model adaptive control algorithm has the best vibration suppression performance.

• 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.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.630-634
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• 2005

본 논문은 신경망을 이용한 간섭 신호 제어 복합 다층 퍼셉트론에서 DS/SS 이동 통신에서 수신된 신호를 검출하는 것에 대하여 연구한다. 수신 신호가 일정한 비트율을 갖는 채널에 전송하기 위하여 신경망을 이용한 새로운 탭 가중치 갱신 제어 방법을 제안한다. 적응 횡단선 필터는 상호 심볼간 간섭을 억압하기 위해 LMS 알고리즘 사용하고, 응답과 실제 출력간의 차인 에러를 이용하여 탭 가중치 조절 메카니즘을 통해 탭 가중치를 갱신함으로서 효과적으로 간섭을 제거한다. 본 논문은 상호 심볼간 간섭을 효율적으로 억압해온 다계층 퍼셉트론 조합을 이용하여 제안된 알고리즘을 통해 탭 가중치 갱신이 보다 효율적으로 이루어질 수 있도록 한다. 시뮬레이션을 통해 평균자승 에러의 수렴 특성이 우월하다는 것을 연구한다.

• Journal of Broadcast Engineering
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• v.8 no.2
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• pp.153-162
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• 2003

This paper describes a structure of adaptive equalizer to improve reception performance of ATSC digital television (DTV) for 8-VSB receivers. There are many strong and dynamic echoes affecting reliable reception of DTV signal. Conventional DFE based least mean square (LMS) algorithm is readily implemented and has good Performance. There are still problems to be solved, however, in handling strong echoes and indoor reception. In this paper, structure of adaptive equalizer to mitigate these Problems in strong multipath interference conditions and indoor reception environment is first presented. Methods to reduce error propagation effects on DFE and initialization scheme of filter coefficients for fast convergence are then introduced. Computer simulation results prove that an adaptive equalizer with proposed design methods can combat with Brazil Ensemble and the Threshold of Visibility(TOV) is improved.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.10-17
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• 2010

In a passive sonar, the improvement of detection performance by using noise cancellation is usually a important problem. In this paper, we have analyzed the own-ship noise cancellation in the two operation modes which are used in the HMS system. In the operator mode, an adaptive line enhancer(ALE) is applied to improve the tonal detection by using broadband noise cancellation and the normalized least mean square(NLMS) algorithm is applied to the design of an adaptive filter. The reference input that is correlated with a primary input can be used to remove the noise incident on the observation directionin the automatic mode. Computer simulations with real sea that data show that the proposed adaptive noise canceller has good performance in passive detection under HMS operation.