• Journal of Electrical Engineering and Technology
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• 제13권2호
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• pp.868-874
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• 2018

In this paper, we propose a new jammer suppression algorithm that uses orthogonal waveform space projection (OWSP) processing for a multiple input multiple output (MIMO) radar system exposed to a jamming signal. Generally, a conventional suppression algorithm based on adaptive beamforming (ABF) needs a covariance matrix composed of the jammer and noise only. By exploiting the orthogonality of the transmitting waveforms of MIMO, we can construct a transmitting waveform space (TWS). Then, using the OWSP processing, we can build a space orthogonal to the TWS that contains no SOI. By excluding the SOI from the received signal, even in the case that contains the SOI and jamming signal, the proposed algorithm makes it possible to evaluate the covariance matrix for ABF. We applied the proposed OWSP processing to suppressing the jamming signal in bistatic MIMO radar. We verified the performance of the proposed algorithm by comparing the SINR loss to that of the ideal covariance matrix composed of the jammer and noise only. We also derived the computational complexity of the proposed algorithm and compared the estimation of the DOD and DOA using the SOI with those using the generalized likelihood ratio test (GLRT) algorithm.

• Journal of Information Processing Systems
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• 제18권3호
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• pp.293-301
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• 2022

Direction of arrival (DOA) estimation of space signals is a basic problem in array signal processing. DOA estimation based on the multiple signal classification (MUSIC) algorithm can theoretically overcome the Rayleigh limit and achieve super resolution. However, owing to its inadequate real-time performance and accuracy in practical engineering applications, its applications are limited. To address this problem, in this study, a DOA estimation algorithm with high parallelism and precision based on an analysis of the characteristics of complex matrix eigenvalue decomposition and the coordinate rotation digital computer (CORDIC) algorithm is proposed. For parallel and single precision, floating-point numbers are used to construct an orthogonal identity matrix. Thus, the efficiency and accuracy of the algorithm are guaranteed. Furthermore, the accuracy and computation of the fixed-point algorithm, double-precision floating-point algorithm, and proposed algorithm are compared. Without increasing complexity, the proposed algorithm can achieve remarkably higher accuracy and efficiency than the fixed-point algorithm and double-precision floating-point calculations, respectively.

• 한국산학기술학회논문지
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• 제11권4호
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• pp.1272-1277
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• 2010

표적 탐지 및 추적을 위한 무선 센서 네트워크에서 센서 노드는 다양한 신호처리 기능을 가져야 한다. 센서 노드의 에너지 제약과 통신 대역폭 제한은 센서 노드에서의 가벼운 신호처리 기법을 필요로 한다. 일반적인 센서 노드에서의 신호처리 기법은 센서 노드에 수신된 신호를 잡음제거와 같은 전처리를 수행하고, 에너지를 계산하여 표적의 위치를 탐지하고 기지국에서의 위치추정 및 식별을 위하여 특징 추출하거나 압축하여 전송하는 등의 방법으로 구성된다. 이러한 센서 노드에서 필수적인 신호처리 기법들은 무선 센서 네트워크의 생존 시간과 표적 탐지 및 식별 성능에 큰 영향을 끼치게 된다. 본 논문에서는 무선 센서 네트워크에서 센서 노드의 필수적인 신호처리들을 리프팅 스킴 웨이블릿 변환 방법을 이용하여 센서 노드에서 에너지 효율적인 신호처리를 수행하고 표적의 정확한 위치를 추정하는 방법을 제안한다.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2000년도 추계종합학술대회논문집
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• pp.129-132
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• 2000

In this paper, we deal with content based searching algorithm for brand image using GHA(Generalized Hebbian Algorithm). GHA is a part of PCA(Principal Component Analysis), that has single-layer perceptron operates and self-organizing performances. We used this algorithm for feature extracts of brand images, and our simulations verify the high performance than present text based methods.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2003년도 하계학술대회 논문집
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• pp.134-136
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• 2003

This paper proposes an algorithm for fingerprint identification using phase only correlation. This algorithm uses the phase of fast Fourier transform and correlation function to calculate the similarity. The algorithm gives very clear result for identification because it shows only one conspicuous sharp peak for the same person's fingerprint. Further, it shows good results even for the finger print images which are printed not clearly and does not need to preprocess the images. It also shows good results for parallel displacement of fingerprint. The experiment result shows the effectiveness of the proposed algorithm.

• Journal of Communications and Networks
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• 제14권1호
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• pp.27-33
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• 2012

In this paper, a new blind signal processing scheme for equalization in fading and impulsive-noise channel environments is introduced based on probability density functionmatching method and a set of Dirac-delta functions. Gaussian kernel of the proposed blind algorithm has the effect of cutting out the outliers on the difference between the desired level values and impulse-infected outputs. And also the proposed algorithm has relatively less sensitivity to channel eigenvalue ratio and has reduced computational complexity compared to the recently introduced correntropy algorithm. According to these characteristics, simulation results show that the proposed blind algorithm produces superior performance in multi-path communication channels corrupted with impulsive noise.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.383-384
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• 2006

The P-ECLMS algorithm adapted the existing Post-Processing method occurs the distortion of the near-end signal at the double-talk situationt. To solve this problem, we propose the SP-ECLMS(Selective Post-Processing ECLMS) algorithm which makes the Post-Processing coefficient differently at the case of the single-talk and the double-talk. When the correlation level is not output less than 30%, the proposed algorithm output the original signal to prevent the signal's distortion.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2005년도 추계학술대회 논문집
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• pp.250-255
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• 2005

본 논문에서는 적응 신호처리의 수렴속도를 향상 시키고 복잡한 계산량을 줄이는 새로운 필터 구조를 제안한다. 그리고 제안한 알고리즘을 웨이블렛 기반 적응 알고리즘에 적용한다. 실제로 합성 음성을 사용하여 적응 잡음 제거기에 적용하여 컴퓨터 시뮬레이션을 통해 제안한 알고리즘과 기존 알고리즘과의 성능을 비교한다. 그 결과 변환 영역 알고리즘은 기존의 시간영역의 알고리즘보다 수렴속도의 향상을 보였고, 웨이블렛 알고리즘, short-length fast running FIR 알고리즘, fast-short-length fast FIR 알고리즘 그리고 제안한 알고리즘에 대한 비교 연구를 수행하였다.

• 한국멀티미디어학회논문지
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• 제9권12호
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• pp.1572-1579
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• 2006

An usual acoustic emission(AE) event has two widely characterized parameters in time domain, peak amplitude and event duration. But noise in AE measuring may disturb the signals with its parameters and aggrandize the signal incertitude. Experiment activity of detection of the nick inside of porcelain with AE was made and study on AE signal processing with statistic be presented in this paper in order to pick-up information expected from the signal with noise. Effort is concentrated on developing a novel arithmetic to improve extraction of the characteristic from stochastic signal and to enhance the voracity of detection. The main purpose discussed in this paper is to treat with signals on amplitudes with statistic mutuality and power density spectrum in frequency domain, and farther more to select samples for neural networks training by means of least-squares algorithm between real measuring signal and deterministic signals under laboratory condition. By seeking optimization with the algorithm, the parameters representing characteristic of the porcelain object are selected, while the stochastic interfere be weakened, then study for detection on neural networks is developed based on processing above.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2000년도 하계종합학술대회논문집
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• pp.65-68
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• 2000

In this paper, as a compatible software radio transmission system, Digital-Radio conversion system which can directly change the digital signal generated by the logic circuit into radio signal is proposed. By the vector synthesis method, the digital signals can change directly into radio signal. If such a circuit is realized, RF circuit and an antenna can be composed by the simple one device, and the radio is directly controlled and performed by the software processing which is the essence of software radio. This Digital-Radio conversion system of this paper give many number of communication channels being offered by PN code and offer a hardware design flexibility by digitization, therefore it decrease the percentage ratio of hardware of system and give a more flexible function of software basis. In this paper, this proposed Digital-Radio conversion system is called D/R converter, and the principle of this D/R converter, radio signal generation algorithm is explained and the performance characteristics of proposed algorithm is shown in time base by the computer simulation method.