• 한국군사과학기술학회지
• /
• 제16권3호
• /
• pp.382-389
• /
• 2013

In anti-jamming GPS receiver, adaptive signal processing techniques in which the radiation pattern of adaptive array antenna of elements may be adaptively changed used to reject interference, clutter, and jamming signals. In this paper, I describes adaptive signal processing technique using the sample matrix inversion(SMI) algorithm. This adaptive signal processing technique can be applied effectively to wideband/narrowband anti-jamming GPS receiver because it does not consider the satellite signal directions and GPS signal power level exists below the thermal noise. I also analyzed the effects of covariance matrix sample size and diagonal loading technique on the system performance of five-element circular array antenna. To attain near optimum performance, more samples required for calculation covariance matrix. Diagonal loading technique reduces the system nulling capability against low-power jamming signals, but this technique improves robustness of adaptive array antenna.

• 한국철도학회논문집
• /
• 제2권2호
• /
• pp.13-20
• /
• 1999

The vibration signals of driving parts of electric train are distorted its signal patterns due to the impact components, which occurs when wheel passes rail joints. An elimination method of the impact components is investigated using adaptive signal processing technique in this study The result shows that adaptive interference canceling method seems to be more effective than line enhancement technique. The application of adaptive interference canceling method to the signal measured at bogie shows that the extractions of the signals of driving parts of traction motor, reduction gear, and axle bearing are successful. Therefore, only the signals of bogie, which is the place to attach an accelerometer easily, is sufficient for the fault diagnosis and the safety evaluation of electric train. Also, adaptive interference canceling method can be applicable to evaluate the performance of vibration isolation between bogie and car body and to investigate the characteristics of indoor sound.

• 전기의세계
• /
• 제29권9호
• /
• pp.584-593
• /
• 1980

Conventional radar antenna systems are susceptible to performance degradation caused by unwanted signals received via the antenna sidelobes and/or mainlobes. Adaptive array systems offer possible solution to this interference problem by automatically steering nulls to unwanted signals providing significant system performance improvement. Another important andvantage of the adaptive array is its self-optimization capability which uses the collective incoming noise data for the nulling purposes. This paper provides a tutorial introduction to adaptive arrays as well as some new development of recent research in this area. Optimum link between the antenna theory and signal processing has been sought by illustrating the gain patterns and output signal-to-noise ratio. Signal acqusition methods are shown including a new attempt of the use of spread-spectrum techniques in conjuction with array systems.

• Journal of Positioning, Navigation, and Timing
• /
• 제9권4호
• /
• pp.379-385
• /
• 2020

The satellite employs an adaptive beamformer to efficiently detect various signals and to suppress multiple interference signals, simultaneously. Although the adaptive beamforming satellite system needs Angle-of-Arrival (AOA) information of the desired signal, it is difficult to estimate the signal AOAs on the satellite environment. However, the AOA estimation on the ground control tower is more efficient and accurate comparing to the satellite environment. In this paper, we propose an adaptive beamforming satellite system based on the signal location information on the ground, consisting on an angle estimator, an adaptive beamformer, and signal processing & D/B unit. The ground control tower estimates the accurate location of the signal source, and it sends the estimated coordinates of the desired signal to the satellite. The angle estimator mounted on the satellite calculates the desired signal AOA, based on the signal location information transmitted from the ground control center. The satellite beamformer detects the desired signal and suppresses unwanted signals based on the signal AOA calculated by the angle estimator. We provide computer simulation results to present the performance of the proposed satellite adaptive beamforming system based on the signal location information.

• 한국철도학회:학술대회논문집
• /
• 한국철도학회 1999년도 춘계학술대회 논문집
• /
• pp.143-150
• /
• 1999

The vibration signals of driving parts of electric train are distorted its signal patterns due to the impact components, which occurs when wheel passes rail joints. An elimination method of the impact components is investigated using adaptive signal processing technique in this study. The result shows that adaptive interference canceling method seems to be more effective than line enhancement technique. The application of adaptive interference canceling method to the signal measured at bogie shows that the extractions of the signals of driving parts of traction motor, reduction gear, and axle bearing are successful. Therefore, only the signals of bogie, which is the place to attach an accelerometer easily, is sufficient for the fault diagnosis and the safety evaluation of electric train. Also, adaptive interference canceling method can be applicable to evaluate the performance of vibration isolation between bogie and car body and to investigate the characteristics of indoor sound.

• 대한전자공학회논문지
• /
• 제25권2호
• /
• pp.119-124
• /
• 1988

Real-time adaptive lattice predictor was implemented on the TMS32020 DSP chip for digital signal processing. The implemented system was composed of Input-Output units and centrla processing-control unit and its supporting assembly soft ware. The performance of hardware realization was verified by comparing input signal and one-step prediction signal which are calcualted by the real-time adaptive lattice predictor. As a result, for 4 stage lattice structure, the maximum running frequency was obtained as 6.41 KHz in this experiment.

• 융합신호처리학회 학술대회논문집
• /
• 한국신호처리시스템학회 2003년도 하계학술대회 논문집
• /
• pp.242-246
• /
• 2003

본 논문에서는 친사용자 환경의 시뮬레이션 환경을 제공하는 SPW 시뮬레이션 플랫포음을 이용하여 스마트 안테나 CDMA 이동통신 시스템을 구현하여 빔형성에 있어서 고정(non-adaptive)의 경우와 적응(adaptive)의 경우로 나누어 성능 분석을 수행하였다. 특히 적응인 경우는 LS(Least Square) 와 LMS (Least Mean Square) 의 적응 알고리즘을 비교하였으며 간섭 기지국의 각도 (Interferer signal angle) 변화에 따른 성능도 비교하였다. SPW 시뮬레이션 결과, 고정 경우보다는 적응 경우가 3［㏈］이상의 이득을 얻을 수 있었으며 LMS 보다는 LS 의 성능이 우수함을 알 수 있었다.

• 대한의용생체공학회:의공학회지
• /
• 제22권3호
• /
• pp.285-291
• /
• 2001

In this paper, we designed two preprocessing adaptive filter - high pass filter and notch filter - using FPGA. For minimizing the calculation load of multi-channel and high-resolution ECG system, we utilize FPGA rather than digital signal processing chip. To implement the designed filters in FPGA, we utilize FPGA design tool(Altera corporation, MAX-PLUS II) and CSE database as test data. In order to evaluate the performance in terms of processing time, we compared the designed filters with the digital filters implemented by ADSP21061(Analog Devices). As a result, the filters implemented by FPGA showed better performance than the filters based on ADSP21061. As a consequence of examination, we conclude that FPGA is a useful solution in multi-channel and high-resolution signal processing.

• Journal of Information Processing Systems
• /
• 제15권4호
• /
• pp.748-764
• /
• 2019

In order to deal with the filtering delay problem of least mean square adaptive filter noise reduction algorithm and music noise problem of spectral subtraction algorithm during the speech signal processing, we combine these two algorithms and propose one novel noise reduction method, showing a strong performance on par or even better than state of the art methods. We first use the least mean square algorithm to reduce the average intensity of noise, and then add spectral subtraction algorithm to reduce remaining noise again. Experiments prove that using the spectral subtraction again after the least mean square adaptive filter algorithm overcomes shortcomings which come from the former two algorithms. Also the novel method increases the signal-to-noise ratio of original speech data and improves the final noise reduction performance.

• 융합신호처리학회논문지
• /
• 제19권2호
• /
• pp.42-46
• /
• 2018

다양한 신호처리 및 통신환경에서 적응신호처리는 매우 중요하다. 적응신호처리 방식 중에서 least mean square(LMS) 알고리즘은 단순하면서도 강인하기 때문에 널리 사용되고 있다. 가변스텝 LMS 알고리즘은 스텝을 가변하므로 빠른 수렴속도와 작은 초과자승오차를 얻을 수 있는 방식이다. 성능향상을 위하여 다양한 가변스텝 LMS 알고리즘이 연구되어 왔다. 하지만 성능향상을 위하여 가변스텝 LMS 알고리즘의 계산 복잡도는 일부 방식에서는 크게 높아지게 되었다. 계산 복잡도가 낮은 고정스텝 LMS 알고리즘과 빠른 수렴속도의 가변스텝 LMS 알고리즘의 장점을 같이 가질 수 있는 간헐적 스텝 갱신 알고리즘을 제안한다. 간헐적으로 스텝 갱신을 할 때 피보나치 수열을 사용하여 스텝 갱신 횟수를 상당히 낮추면서도 가변스텝 LMS 알고리즘의 성능을 유지할 수 있었다. 적응 등화기에 제안한 가변스텝 LMS 알고리즘을 적용하여 그 성능을 확인하였다.