• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• v.2
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• pp.143-148
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• 2006

The goal of this paper is to analyze, through simulations and experiments, GNSS interference mitigation performance under various types of antenna structures against wideband and narrowband interferences using spatial-temporal adaptive signal processing (STAP) techniques. The STAP approach, which combines spatial and temporal processing, is a viable means of GNSS array signal processing that enhancing the desired signal quality and providing protection against interference. In this paper, we consider four types of 3D antenna array structure - Uniform Linear Array (ULA), Uniform Rectangular Array (URA), Uniform Circular Array (UCA), and the Single-Ring Cylindrical Array (SRCA) under an interference environment. Analytical evaluation and simulations are performed to investigate the system performance. This is followed by simulation GPS orbits in interfered environment are used to evaluate the STAP performance. Furthermore, experiments using a 2x2 URA hardware simulator data show that with the removal of wideband and narrowband interference through the STAP techniques, the signal tracking performance can be enhanced.

• Journal of Information Processing Systems
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• v.15 no.4
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• pp.748-764
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• 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.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.17-20
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• 2001

In this paper, we realized a transceiver system for short distance communication with a commercial RF module working in ISM band. Also we measured system performance by transmitting baseband data in a building and then we compared the demodulated data bits with stored data bits in a PC connected with demodulator. The RF module In the experiments works only in the bandwidth of 424MHz-429MHz. The signal level degrades as the distance between transmitter and receiver increases. We measured the signal level and bit error and present the measured results with various locations in the building. And it is concluded that the measured data may be used in the design of short distance network in a building.

• The Journal of the Acoustical Society of Korea
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• v.8 no.5
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• pp.95-101
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• 1989

For developing new algorithms or dedicated hardware by using general purpose Digital Signal Processor chip, emulator H/W and simulator S/W are indispensible. But the most of DSP emulators have limitations on H/W flexibility according to their generalized architectures. In this paper, a DSP evaluation system for real time signal processing was developed using TMS 32020. The I/O buffers storing acquisition data of the system were designed to have variable length $(1\sim2048samp1es) &$ sampling frequency $l00\sim8KHz$.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.462-465
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• 2000

It is important to compress three dimensional (3D) data efficiently, since 3D data are too large to store or transmit in general. In this paper, we propose a lossless compression algorithm of the 3D mesh connectivity, based on the vertex degree. Most techniques for the 3D mesh compression treat the connectivity and the geometric separately, but our approach attempts to exploit the geometric information for compressing the connectivity information. We use the geometric angle constraint of the vertex fanout pattern to predict the vertex degree, so the proposed algorithm yields higher compression efficiency than the conventional algorithms.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1296-1299
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• 2002

We implemented a CD signal processor operated on a CAV 48-speed CD-ROM drive into a VLSI. The CD signal processor is a mixed mode monolithic IC including servo-processor, data recovery, data-processor, and I-bit DAC. For servo signal processing, we included a DSP core, while, for CAV mode playback, we adopted a PLL with a wide recovery range. Data processor (DP) was designed to meet the yellow book specification.［2］So, the DP block consists of EFM demodulator, C1/C2 ECC block, audio processor and a block transferring data to an ATAPI chip. A modified Euclid's algorithm was used as a key equation solver for the ECC block To achieve the high-speed decoding, the RS decoder is operated by a pipelined method. Audio playability is increased by playing a CD-DA disc at the speed of 12X or 16X. For this, subcode sync and data are processed in the same way as main data processing. The overall performance of IC is verified by measuring a transfer rate from the innermost area of disc to the outermost area. At 48-speed, the operating frequency is 210 ㎒, and this chip is fabricated by 0.35 um STD90 cell library of Samsung Electronics.

• Journal of IKEEE
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• v.22 no.3
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• pp.532-538
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• 2018

This paper proposes an algorithm for signal processing which is used in pulse induction metal mine detectors. The detection power can be obtained from magnetic variation on the search coil. The calibration data should be made when there is no target because the detection power is difference between with and without a target. And it is also updated periodically because of surrounding various noises. Lastly, we keep a watch on the signal slope to identify exact position and signal power of mine detection.

• The Journal of Society for e-Business Studies
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• v.23 no.2
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• pp.111-121
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• 2018

Many problems in rotating machinery such as aircraft engines, wind turbines and motors are caused by bearing defects. The abnormalities of the bearing can be detected by analyzing signal data such as vibration or noise, proper pre-processing through a few signal processing techniques is required to analyze their frequencies. In this paper, we introduce the condition monitoring method for diagnosing the failure of the rotating machines by analyzing the vibration signal of the bearing. From the collected signal data, the normal states are trained, and then normal or abnormal state data are classified based on the trained normal state. For preprocessing, a Hamming window is applied to eliminate leakage generated in this process, and the cepstrum analysis is performed to obtain the original signal of the signal data, called the formant. From the vibration data of the IMS bearing dataset, we have extracted 6 statistic indicators using the cepstral coefficients and showed that the application of the Mahalanobis distance classifier can monitor the bearing status and detect the failure in advance.

• 전기의세계
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• v.29 no.9
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• pp.584-593
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• 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.

• 제어로봇시스템학회:학술대회논문집
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• 1992.10a
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• pp.829-833
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• 1992

This paper presents the design of an automated EEG analyzing system. The design considerations including processing speed, A/D conversion, filtering, and waveforms detection, are overviewed with the description of the associated EEG characteristics. The architecture of the currently implemented system consists of a p-controller based front-end signal processing unit and a host computer system. The data acquisition procedures are described along with a couple of illustrations of the acquired EEG/EOG signal.