• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.559-567
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• 2020

This paper proposes a noise cancelling algorithm to remove own-ship noise for a towed array sonar. Extra beamforming is performed using partial channels of the acoustic array to get a reference beam signal robust to the noise bearing. Frequency domain Adaptive Noise Cancelling (ANC) is applied based on Normalized Least Mean Square (NLMS) algorithm using the reference beam. The bearing of own-ship noise is estimated from the coherence between the reference beam and input beam signals. Own-ship noise level is calculated using a beampattern of the noise with estimated steering angle, which prevents loss of a target signal by determining whether to update a filter so that removed signal level does not exceed the estimated noise level. Simulation results show the proposed algorithm maintains its performance when the own-ship gets out off its bearing 40 % more than the conventional algorithm's limit and detects the target even when the frequency of the target signal is same with the frequency of the own-ship signal.

• Journal of Positioning, Navigation, and Timing
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• v.11 no.3
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• pp.163-172
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• 2022

Typical anti-jamming technologies based on array antennas, Space Time Adaptive Process (STAP) & Space Frequency Adaptive Process (SFAP), are very effective algorithms to perform nulling and beamforming. However, it does not perform equally well for all types of jamming signals. If the anti-jamming algorithm is not optimized for each signal type, anti-jamming performance deteriorates and the operation stability of the system become worse by unnecessary computation. Therefore, jamming classification technique is required to obtain optimal anti-jamming performance. Machine learning, which has recently been in the spotlight, can be considered to classify jamming signal. In general, performing supervised learning for classification requires a huge amount of data and new learning for unfamiliar signal. In the case of jamming signal classification, it is difficult to obtain large amount of data because outdoor jamming signal reception environment is difficult to configure and the signal type of attacker is unknown. Therefore, this paper proposes few-shot jamming signal classification technique using meta-learning and transfer-learning to train the model using a small amount of data. A training dataset is constructed by anti-jamming algorithm input data within the GNSS receiver when jamming signals are applied. For meta-learning, Model-Agnostic Meta-Learning (MAML) algorithm with a general Convolution Neural Networks (CNN) model is used, and the same CNN model is used for transfer-learning. They are trained through episodic training using training datasets on developed our Python-based simulator. The results show both algorithms can be trained with less data and immediately respond to new signal types. Also, the performances of two algorithms are compared to determine which algorithm is more suitable for classifying jamming signals.

• ETRI Journal
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• v.20 no.2
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• pp.133-148
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• 1998

The objective of this paper is to present an adaptive algorithm for computing the weight vector which provides a beam pattern having its maximum gain along the direction of the mobile target signal source in the presence of interfering signals within a cell. The conjugate gradient method (CGM) is modified in such a way that the suboptimal weight vector is produced with the computational load of O(16N), which has been found to be small enough for the real-time processing of signals in most land mobile communications with the digital signal processor (DSP) off the shelf, where N denotes the number of antenna elements of the array. The adaptive procedure proposed in this paper is applied to code division multiple access (CDMA) mobile communication system to show its excellent performance in terms of signal to interference plus noise ratio (SINR), bit error rate (BER), and capacity, which are enhanced by about 7 dB, ${\frac{1}{100}}$ times, and 7 times, respectively, when the number of antenna elements is 6 and the processing gain is 20 dB.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.413-414
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• 2007

Successful speech recognition in noisy environments for intelligent robots depends on the performance of preprocessing elements employed. We propose an architecture that effectively combines adaptive beamforming (ABF) and blind source separation (BSS) algorithms in the spatial domain to avoid permutation ambiguity and heavy computational complexity. We evaluated the structure and assessed its performance with a DSP module. The experimental results of speech recognition test shows that the proposed combined system guarantees high speech recognition rate in the noisy environment and better performance than the ABF and BSS system.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.10a
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• pp.468-472
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• 2000

This paper discusses the development of a acoustic vector channel simulator for the performance analysis of a acoustic digital communication system. The channel simulator consists of transmission module, acoustic channel model, receiver, beamformer, and adaptive equalizer. QPSK source signal is generated by the parameters specified by a user and transmitted. The transmitted signal generate multipath signals which have a different delay, amplitude, and dopper Sequency. The multipath signals with the acoustic noises is the received signal. We can analysis the communication system performance according to the antenna structure, beamforming algorithm, and equalization algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9A
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• pp.694-701
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• 2003

In this paper, we present a hardware structure and new features of a smart antenna BTS (Base Transceiver Station) for CDMA2000 1X system. The proposed smart antenna BTS is a composite system consisting of many subsystems, i.e., array antenna element, frequency up/down converters, AD (Analog-to-Digital) and DA (Digital-to-Analog) converters, spreading/despreading units, convolutional encoder/Viterbi decoder, searcher, tracker, beamformer, calibration unit etc. Through the experimental tests, we found that the desired beam-pattern in both uplink and downlink communications is provided through the calibration procedure. Also it has been confirmed that the adaptive beamforming algorithm adopted to our smart antenna BTS is fast and accurate enough to support 4 fingers to each user. In our experiments, commercial mobile terminals operating PCS (Personal Communication System) band have been used. It has been confirmed that the smart antenna BTS tremendously improves the FER (Frame Error Rate) performance compared to the conventional 2-antenna diversity system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.9C
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• pp.871-879
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• 2009

The performance variations of a spatio temporal smart antenna which is equipped at the basestation of CDMA cellular communication network due to the parametric change of multipath fading environment are studied in this paper. The smart antenna of interest employs space diversity based adaptive array structure in conjunction with rake receiver that has fingers the number of which is the same as that of multipath links. The beamforming is achieved via LMS(Least Mean Square) algorithm in which a reference signal is generated using decision directed formula. It has been shown by computer simulation that the performance of our smart antenna of interest depends significantly upon not only the degree of desired signal's DOA(Direction of Arrival)spread but the number of fingers of the rake receiver. The relative insensitivity of the smart antenna's performance on desired signal's delay spread has also been observed. Computer simulation has shown that the increase of the number of fingers brings in a nonlinear enhancement of the performance of our smart antenna. The renewal of weight vector in the beamforming procedure is taken place at post PN despread stage.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.6
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• pp.162-170
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• 2010

Adaptive arrays reject interferences while preserving the desired signal, exploiting a priori information on its arrival angle. Subspace-based adaptive arrays, which adjust their weight vectors in the signal subspace, have the advantages of fast convergence and robustness to steering vector errors, as compared with the ones in the full dimensional space. However, the complexity of theses subspace-based methods is high because the eigendecomposition of the covariance matrix is required. In this paper, we present a simple subspace-based method based on the PASTd (projection approximation subspace tracking with deflation). The orignal PASTd algorithm is modified such that eigenvectora are orthogonal to each other. The proposed method allows us to significantly reduce the computational complexity, substantially having the same performance as the beamformer with the direct eigendecomposition. In addition to the simple beamforming method, we present theoretical analyses on the SINR (signal-to-interference plus noise ratio) of subspace beamformers to see their behaviors.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.13 no.10
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• pp.989-997
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• 2002

This paper presents a new technique that enhances the performance of the smart antenna system especially in signal environments of wide angular spread by adopting a weight vector obtained from two eigenvectors of theautocovariance matrix of the received data. While the conventional beamformingtechnique employs only one eigenvector corresponding to the largest eigenvalue, the proposed algorithm uses two eigenvectors corresponding to the largest and second largest eigenvalue in such a way that it can be robust enough to the signal environments of wide angular spread. An efficient adaptive procedure is shown to verify that the optimal weight vector consisting of the two eigenvectors is obtained with a reasonable complexity(3.5$N_2$+ 12N) and accuracy. it is also shown in this paper that the numerical results obtained from the proposed adaptive procedure well agree with those obtained from a commercial tool computing the eigen-function of MATLABTM.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7C
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• pp.413-420
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• 2011

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.