• The Journal of the Acoustical Society of Korea
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• v.25 no.2E
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• pp.36-42
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• 2006

In this paper, a novel post processing using Wiener filtering technique is proposed to p rm further interference reduction in FDICA. Using the proposed method, the target signal components are remained with little attenuation while the interference components are drastically suppressed. The results of experiments show that the proposed method achieves a reduction of the residual crosstalk. Compared to the NLMS method, the proposed method has slightly better separation performance in SIR, and even requires much less computational complexity.

• Journal of Positioning, Navigation, and Timing
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• v.11 no.2
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• pp.127-134
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• 2022

In this paper, a low-cost, flexible GPS simulator based on USRP is designed as a general-purpose software wireless front-end. The simulator consists of a software GPS signal generator and a USRP-based RF transmitter. The simulator supports various scenarios including specified reception time, quantization bit level, I/Q data types, IF frequency, sampling frequency, SNR, ionospheric delay and user dynamics. The generated GPS RF signal is verified using the spectrum analyzer and off-the-shelf GNSS receivers such as U-blox M8T. The experimental results shows that the difference between generated and real live signal is ignorable. It is expected that designed GPS simulator can be used to GNSS signal design, receiver design and signal processing algorithms such as anti-jamming.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.1
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• pp.119-127
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• 2012

This paper describes development and performance test of signal processor for the millimeter wave seeker. A ground to air guidance missile is required various beam patterns in order to counteract different kind of target. Therefore, we designed the hardware and software architecture considering flexibility. This signal processor consists of ADC, FPGA, DSP and etc. FPGA provides peripheral interface to DSP and convert digital IF signal to baseband signal. DSP performs signal processing, calculates target's information and controls devices. Each parts' hardware are connected in series and signal processing algorithms for various beam patterns are built in parallel.

• Journal of electromagnetic engineering and science
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• v.9 no.3
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• pp.130-140
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• 2009

The design theory and experimental results of a true random noise radar system are presented in this paper. Target range information can be extracted precisely by correlation processing between the delayed reference and the signal received from a target, and the velocity information by the Doppler processing with successive correlation data. A K-band noise radar system was designed using random FM noise signal, and the characteristics of the fabricated system were examined with laboratory and outdoor experiments. A C-band random FM noise signal was generated by applying a low-frequency white Gaussian noise source to VCO(Voltage Controlled Oscillator), and a K-band Tx noise signal with 100 MHz bandwidth was obtained by using a following frequency multiplier. Two modified wave-guide horn arrays were designed and fabricated, and used for the Tx and Rx antennas. The required amount of Tx/Rx isolation was attained by using a coupling cancellation circuit as well as keeping them apart with predetermined spacing. A double down-conversion scheme was used in the Rx and reference channels, respectively, for easy post processing such as correlation and Doppler processing. The implemented noise radar performance was examined with a moving bicycle and a very high-speed target with a velocity of 150 m/s. The results extracted by the Matlab simulation using the logging data were found to be in a reasonable agreement with the expected results.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.11
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• pp.1481-1488
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• 1995

Due to the low loss, broadband and accurate short time delay properties of optical fiber, it has attracted as a delay medium for high speed and broad-band signal processing. In this paper, we consider the coherent optical fiber filter of IIR lattice structure, which uses coherent light sources and consists of directional couplers whose coupling coefficients are restricted between 0 and 1. Considering restrictions of directional coupler, the design formulae and condition for realibility of optical fiber filter of IIR lattice structure which makes the optimal use of optical signal energy are derived.

• ETRI Journal
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• v.34 no.5
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• pp.655-665
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• 2012

Based on wideband precoding (WBP) in the multiple-input multiple-output orthogonal frequency division multiplexing system, an adaptive nonuniform codebook is presented in this paper. The relationship between the precoder distribution and spatial correlation is analyzed at first. A closed-form expression based on overlapped isosceles triangles is proposed as an approximation of the precoder distribution. Then, the adaptive codebook design is derived with the approximate distribution to minimize quantization errors. The capacity and bit error rate performance demonstrate that the adaptive codebook with WBP outperforms the conventional fixed uniform codebook.

• ETRI Journal
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• v.35 no.4
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• pp.697-705
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• 2013

Emerging trends in the area of digital very large scale integration (VLSI) signal processing can lead to a reduction in the cost of the cochlear implant. Digital signal processing algorithms are repetitively used in speech processors for filtering and encoding operations. The critical paths in these algorithms limit the performance of the speech processors. These algorithms must be transformed to accommodate processors designed to be high speed and have less area and low power. This can be realized by basing the design of the auditory filter banks for the processors on digital VLSI signal processing concepts. By applying a folding algorithm to the second-order digital gammatone filter (GTF), the number of multipliers is reduced from five to one and the number of adders is reduced from three to one, without changing the characteristics of the filter. Folded second-order filter sections are cascaded with three similar structures to realize the eighth-order digital GTF whose response is a close match to the human cochlea response. The silicon area is reduced from twenty to four multipliers and from twelve to four adders by using the folding architecture.

• 제어로봇시스템학회:학술대회논문집
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• 1990.10a
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• pp.559-563
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• 1990

A new speed improvement method for quantitative superimposed EMG signal analysis to diagnose the neuromuscular dysfunction is described. The improvement is achieved through the use of efficient software and hardware signal processing techniques. The software approch is composed of the MANDF filter and HRWA algorithm which provides the optimal set and time delays of-selected templates. The hardware employs a TMS32OC25 DSP chip to execute the intensive calculation part. The purposed method is verified through a simulation with real templates which are obtained from needle EMG. As a results, the proposed method provides an overall speed improvement of 32-40 times.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.41 no.5
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• pp.49-56
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• 2004

A low power consuming voice signal processing and control module was designed using a small $\mu$-controller for use in a totally implantable middle ear system. The module was designed that it can control the implanted system as well as process the fitting algorithm of input sound signal. In ordinary operation mode, the $\mu$-controller processes the applied sound signal for compensating the hearing loss of the patients. When the control signal is applied from the IR receiving module, the $\mu$-controller interrupts the signal processing and executes the order of the control signals such as power on/off, volume up/down. The designed module was implemented and verified the performance of the system through several experiments.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.627-630
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• 1998

In this study, we derived stop-and-go normalized DD, dual-mode normalized sato, dual-mode NCMA blind equalization algorithm for complex data. and then, the convergence characteristics of the proposed SG-NDD, dual-mode NSato blind equalization algorithms are compared with those of SG-DD, dual-mode sato algorithm. In genral, the normalized blind equalization algorithms have better convergence characteristics than the conventional algorithms.