• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.7
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• pp.1603-1609
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• 2010

This Paper deals with a design, making a prtotype and test methods of Real-time towed Line Array Sensor Data processing board for fast data communication and long range data transmission with SFM(Serial FPDP Module) through Optic-fiber channel. Towed line array sensors are installed in Frigate and the each LAS A, B, C group data from LAS is packed a previously agreed protocol and transmitted to the signal processing unit. Considering the limited space of VME 6U size, LAS Data processing board is designed with MPC8265 PowerPC Controller of Freescale for main system control and Altera's CycloneIII FPGA for sensor data packing, self-test simulation data generation, S/W FIFO et cetera. LAS Data processing board have VxWorks, the RTOS(Real Time Operating System) that present many device drivers, peripheral control libraries on board for real-time data processing.

• Journal of the Institute of Convergence Signal Processing
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• v.8 no.4
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• pp.266-271
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• 2007

In mobile channel environments, the effect of the multi-path fading reduces the performance of the mobile communication severely. Previous studies on the effect of the multi-path fading show that, only the channel between fixed base stations and mobile terminals are considered. However, the future mobile communication service such as telematics may use the communication between moving mobile terminals, so called mobile ad-hoc communication, the studies on the channel is required for such situations. In this paper we have measured the mobile channel and analyzed the fading characteristics when using the linear array antennas. In the data measurements, we sent the QPSK modulated signal with carrier frequency of 2GHz band and received the signal using the array antennas equipped on moving vehicles. We have analyzed the fading characteristics and the effects of the fading reduction using the CDF of fading depth using data collected from two moving vehicles. The simulation results show that the beamforming technique adds 6 or 7dB to the fading margin than that of the selection diversity depending on channel environments.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.21 no.5
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• pp.11-18
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• 2021

The small radio frequency tracking radar is a tracking system with a radio frequency sensor that identifies a target through all-weather radio frequency signal processing for a target and searches, detects and tracks the target for the major target. In this paper, we describe the development of a board equipped with TMS320C6678 and XILINX FPGA (Field Programmable Gate Array), a high-speed multi-core DSP that acquires target information through all-weather radio frequency and identifies a target through real-time signal processing. We propose DSP-FPGA combination architecture for DSP and FPGA selection and signal processing, and also explain the design of SRIO for high-speed data transmission.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.19 no.1
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• pp.9-14
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• 2019

A small millimeter-wave tracking radar is a pulse-based radar that searches, detects, and tracks a target in real time through a TWS (Track While Scan) method for a traps target on the sea with a large RCS running at low speed. It is necessary to develop a board equipped with a high-speed CPU to acquire and track target information through LPRF, DBS, and HRR signal processing techniques for a trap target operating various kinds of dexterous objects such as chaff and decoy, We designed a signal processor structure including DFT (Discrete Fourier Transform) module design that can perform real - time FFT operation using FPGA (Field Programmable Gate Array) and verified the signal processor implemented through performance test.

• The Transactions of the Korea Information Processing Society
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• v.5 no.8
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• pp.1985-1996
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• 1998

A parallelising algorithm for partitioning and mapping methods of matrix/vector multiplication into linear processor array/VLW simulator is presented in this paper. First we discuss the mapping methods for input matrix or vector into the arbitrarily size of processor arrays. Then, we show partitioning the algorithmss of the large size of computational problem into the size of the processor array. We execute the algorithm on VLIW simuhator and show to effectiviness of algorithm. The result which we achived better parallelising performance on our VLIW simulator dsign than on linear processor array.

• Journal of Positioning, Navigation, and Timing
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• v.5 no.3
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• pp.131-136
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• 2016

The present paper shows that beamforming algorithm such as Minimum Variance Distortionless Response (MVDR) based on array antenna signal processing can have not only anti-jamming but also anti-spoofing characteristics. A beam pattern due to the beamforming algorithm strengthens received signal power as it is formed in the incident direction of desired signal. During the process, the effect of unnecessary signals such as spoofing signals can be reduced because the beam pattern reduces received signal power in the incident directions excluding the beam pattern-directed direction. In order to analyze the anti-spoofing effect due to the beamforming algorithm, a software-based simulation environment was configured. An arbitrary error was applied between incident direction of Global Positioning System (GPS) satellite signal and steering vector direction of the beamforming algorithm to analyze the received signal power and required conditions were provided to see the anti-spoofing effect due to the beamforming algorithm. The used antenna was 7-element planar circular array and beam patterns were formed through the MVDR algorithm.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.305-309
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• 2004

The problem for parameters estimation of the received signals impinging on array sensors has long been of great research interest in a great variety of applications, such as radar, sonar, and land mobile communications systems. Conventional subspace-based algorithms, such as MUSIC and ESPRIT, require an extensive computation of inverse matrix and eigen-decomposition. In this paper, we propose a new parameters estimation algorithm via nonlinear minimization, which is simplified computationally and estimates signal parameters simultaneously.

• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.10
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• pp.1263-1273
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• 1988

An adaptive nulling technique is presented to synthetically overcome the integrated problems associated with the conventional LMS array in the performances of jammer rejection, convergence rate, misadjustment, and reference signal generation. The proposed method is to remove the target signal from the array input and to eliminate the reference signal prior to minimization processing. The algorithm is constrained to the residue noise level in adaptive processor. Analysis shows effectiveness of the algorithm for coherent and/or incoherent interference rejection, wide dynamic range of convergence factor, rapid adaptation rate, and small mean square error. Simulation results confirm the theoretical prediction.

• 제어로봇시스템학회:학술대회논문집
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• 1999.10a
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• pp.266-269
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• 1999

An electronically Scannable microphone system is in the Planning stage. For this Purpose, a multiple microphone array with controllable delay is available. To achieve effective point-listening characteristics, we proposed an enclosed microphone array system with a complex weighting method. In this system, both the microphone arrangement and the value of the complex weighting are important. In this report, the construction of microphone array system and the signal-processing method are explained, and the calculation method for optimal complex weighting is also presented. A prototype experimental setup is designed and fabricated to verify the expected characteristics.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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• pp.38-46
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• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.