• Journal of Communications and Networks
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• v.14 no.1
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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.

• Journal of Semiconductor Engineering
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• v.1 no.2
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• pp.64-73
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• 2020

As CMOS technology scales down, the design of analog signal processing circuit becomes far more difficult because of steadily decreasing supply voltage and smaller intrinsic gain of transistors. With sub-1V supply voltage, the conventional analog signal processing relying on high-gain amplifiers is not an effective solution and different approach has to be sought. One of the promising approaches is "time-domain analog signal processing" which exploits the improving switching speed of transistors in a scaled CMOS technology. In this paper, various time-domain analog signal processing techniques are explained with some experimental results.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.302-305
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• 2005

Recently, many methods to analyze signal have been proposed and representative methods are the Fourier transform and wavelet transform. In these methods, the Fourier transform represents signal with combination cosine and sine at all locations in the frequency domain. However, it doesn't provide time information that particular frequency occurs in signal and denpends on only the global feature of the signal. So, to improve these points the wavelet transform which is capable of multiresolution analysis has been applied to many fields such as speech processing, image processing and computer vision. And the wavelet transform, which uses changing window according to scale parameter, presents time-frequency localization. In this paper, we proposed a new approach using a wavelet of cosine and sine type and analyzed features of signal in a limited point of frequency-time plane.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.12
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• pp.31-39
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• 1993

In the playback signal processing of a digital magnetic recording system, the major signal processing processes consist of pulse equalization. pulse detection, clock recovery, and data recovery. Equalizer which compensates interference occurrde between pulses recorded in high density on a magnetic media is realized by pulse slimming method, and pulse detection by a integrating detector. Clock recovery from the detector output was accomplished by using PLL. and data recovery to reduce noise effects was carried out by utilizing the three sampling clocks recovered in clock recovery process. In this paper these processes are implemented in hardware and its performance is evaluated by experimenting with a commercial DAT. It was found that the playback signal processor proposed is suitable to the practical high density magnetic recording system.

• Journal of the Korean Institute of Telematics and Electronics C
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• v.34C no.9
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• pp.33-42
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• 1997

Fixed-point optimization utility software that can aid scaling and wordlength determination of digital signal processign algorithms written in C or C$\^$++/ language is developed. This utility consists of two programs: the range estimator and the fixed-point simulator. The former estimates the ranges of floating-point variables for automatic scaling purpose, and the latter translates floating-point programs into fixed-point equivalents for evaluating te fixed-point performance by simulation. By exploiting the operator overloading characteristics of C$\^$++/ language, the range estimation and the fixed-point simulation can be conducted just by modifying the variable declaration of the original program. This utility is easily applicable to nearly all types of digital signal processing programs including non-linear, time-varying, multi-rate, and multi-dimensional signal processing algorithms. In addition, this software can be used for comparing the fixed-point characteristics of different implementation architectures.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.6
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• pp.67-76
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• 1997

In radar signal processing, an A/D converter with sufficient dynamic range and high sampling speed is required to detect the weakest target signals in heavy clutter and ECM environments. As the sampling frequency increases, the amount of digital data transfered to the signal processing module is also increased. To overcome these massive data transfer burden, we need an A/D conversion module with an enough data transfer rate. In this paper, we proposed an implementation scheme of a new A/D conversio module that can be used in multi-mode 3-D phased array radar signal processing system, and evaluated the performance. The proposed A/D conversion module is implemented with a standard A/D converter and a 6U-standard VME bus.

• ETRI Journal
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• v.40 no.1
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• pp.133-145
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• 2018

Robustness, payload, and imperceptibility of audio watermarking algorithms are contradictory design issues with high-level security of the watermark. In this study, the major issue in achieving high payload along with adequate robustness against challenging signal-processing attacks is addressed. Moreover, a security code has been strategically used for secure transmission of data, providing tamper detection at the receiver end. The high watermark payload in this work has been achieved by using the complementary features of third-level detailed coefficients of discrete wavelet transform where the human auditory system is not sensitive to alterations in the audio signal. To counter the watermark loss under challenging attacks at high payload, Daubechies wavelets that have an orthogonal property and provide smoother frequencies have been used, which can protect the data from loss under signal-processing attacks. Experimental results indicate that the proposed algorithm has demonstrated adequate robustness against signal processing attacks at 4,884.1 bps. Among the evaluators, 87% have rated the proposed algorithm to be remarkable in terms of transparency.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1216-1221
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• 2005

The GPS software receiver based on the SDR(Software Defined Radio) technology provides the ability to easily adopt other signal processing algorithms without changing or modifying the hardware of the GPS receiver. However, it is difficult to implement the GPS software receiver using a commercial processor because of heavy computation load for processing the GPS signals in real time. This paper proposes an efficient GPS signal processing scheme and correlator structure to reduce the computation load for processing the GPS signal in the GPS software receiver, which uses a patterned look-up table method to generate the correlation value between the GPS signals and the replica signals. In this paper, it is explained that the computation load of the proposed scheme is much smaller than that of the previous GPS signal processing scheme. Finally, the processing time of the proposed scheme is compared with that of the previous scheme, and the improvement is shown from the viewpoint of the computation load.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.5
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• pp.50-57
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• 1994

In the non-destructive evaluation techniques using ultrasonic signal, backscattering noise from grain interface decreases the SNR of received signal. In this paper, SSP(split-spectrum processing) based on the constant FBR decomposition method has been applied to enhance the SNR. This algorithm helps to find optimal parameters of filter bank through a simple theory and has an advantage that reduce the signal processing time compared with the conventional constant bandwidth decomposition method. In this experiment, the 304 stainless steel sample is heat-treated and received ultrasonic signal is processed by SSP using the constand bandwidth decomposition method and the constand FBR decomposition method enhanced the SNR by 1.4 dB and reduced the required number of filters by 4 compared with the constant bandwidth decomposition method.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.655-656
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• 2006

A parallel analog Viterbi decoder which decodes PR (1,2,2,1) signal of optical disc has been fabricated into chip. The proposed parallel analog Viterbi decoder implements the functions of the conventional digital Viterbi decoder utilizing the analog parallel processing circuits. In this paper, the analog parallel Viterbi decoding technology is applied for the PR signal. The benefit of analog processing is the low power consumption and the less silicon consumption. The test results of the fabricated chip are reported in this paper.