• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.9A
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• pp.931-939
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• 2007

This paper describes the system which automatically generates instruction-set simulators cores using the SMDL. SMDL describes structure and instruction-set information of a target DSP machine. Analyzing behavioral information of each pipeline stage of all instructions on a target ASIPS, the proposed system automatically generates a cycle-accurate instruction set simulator in C++ for a target processor. The proposed system has been tested by generating instruction-set simulators for ARM9E-S, ADSP-TS20x, and TMS320C2x architectures. Experiments were performed by checking the functions of the $4{\times}4$ matrix multiplication, 16-bit IIR filter, 32-bit multiplication, and the FFT using the generated simulators. Experimental results show the functional accuracy of the generated simulators.

• International Journal of Control, Automation, and Systems
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• v.6 no.6
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• pp.794-799
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• 2008

This paper presents a user identification method at H.264 streaming using watermarking with fingerprints. The watermark can efficiently reduce the potential danger of forgery or alteration. Especially a biometric watermark has convenient, economical advantages. The fingerprint watermark can also improve reliability of verification using automated fingerprint identification systems. These algorithms, however, are not robust against common video compression. To overcome this problem, we analyze H.264 compression pattern and extract watermark after restoring damaged watermark using various filters. The proposed algorithm consists of enhancement of a fingerprint image, watermark insertion using discrete wavelet transform and extraction after restoring. The proposed algorithm can achieve robust watermark extraction against H.264 compressed videos.

• 제어로봇시스템학회:학술대회논문집
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• 1996.10a
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• pp.157-160
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• 1996

Physical systems axe generally continuous-time in nature. However as the data measured from these systems is generally in the form of discrete samples, and most modern signal processing is performed in the discrete-time domain, discrete-time models are employed. This paper describes methods for estimating the coefficients of continuous-time system within a closed loop control system. The method employs a recursive estimation algorithm to identify the coefficients of a discrete-time bilinear-operator model. The coefficients of the discrete-time bilinear-operator model closely approximate those of the corresponding continuous-time Laplace transform transfer function.

• Proceedings of the KIEE Conference
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• 1991.07a
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• pp.833-836
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• 1991

This paper proposes several types of 2-D interpolation lifters for HD-MAC decoder. Filters considered here are FIR, IIR, median, and FMH. Their structure and coefficients have been determined to be appropriate for real-time computation as well as to have good reconstructability. Results or computer simulation are also presented to show the performance of those filters.

• Proceedings of the KIEE Conference
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• 2011.07a
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• pp.2007-2008
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• 2011

In this paper, we implemented active noise control systems for active barriers with attenuation of road noise. And then we constructed by multi-channel audio systems and DAQ part, high performance DSP H/W. Active noise control firmware program was implemented for multi-channel off-line/on-line estimation methods for secondary path transfer functions and FIR/IIR filter structure main noise control algorithms. To evaluate performance of implemented systems, active barriers test bed was implemented and several experiments was performed by various noise cases.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.2
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• pp.306-317
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• 1997

This ppaer presents a bit-rate conversion system for efficient communications between 3 channel CVSD systems with different bit-rates. The proposed conversion system is implemented in the digital domain and specially, the conversion problem between 32 Kbps and 16 Kbps CVSD systems is studied. The conventional conversion system implemented in the analog domain allows signals to be easily degraded by external noises. To overcome this problem, a digital CVSD bit-rate conversion system robust to external noises is developed. the new systemdecodes CVSD bit sequences and converts sampling rates of decoded signals, then encodes signals at target bit-rates. Since linear phase property does not matter in this application, instead of FIR filters a IIR filter is employed to reduce the system complexity. Therefore, a 3 channel digital CVSD bit-rate conversion system was successfully real-time implemented using a general purpose DSP. In addition, conversion problems with unkown time constants were experimented and good experimental results were obtained.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.32B no.1
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• pp.96-102
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• 1995

This paper presents the VHSIC Hardware Description Language(VHDL) implementation of the Fixed Point Covariance Lattice(FLAT) algorithm for an Linear Predictive Coding(LPC) analysis and its related algorithms, such as the forth order high pass Infinite Impulse Response(IIR) filter, covariance matrix calculation, and Spectral Smoothing Technique(SST) in the Vector Sum Exited Linear Predictive(VSELP) speech coder that has been Selected as the standard speech coder for the North America and Japanese digital cellular. Existing Digital Signal Processor(DSP) chips used in digital cellular phones are derived from general purpose DSP chips, and thus, these DSP chips may not be optimal and effective architectures are to be designed for the above mentioned algorithms. Then we implemented the VHDL code based on the C code, Finally, we verified that VHDL results are the same as C code results for real speech data. The implemented VHDL code can be used for performing logic synthesis and for designing an LPC Application Specific Integrated Circuit(ASOC) chip and DsP chips. We first developed the C language code to investigate the correctness of algorithms and to compare C code results with VHDL code results block by block.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.334-336
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• 2005

According as quality of life improves, pursuit of agreeable iife became realistic problem. Specially, noise had been appraised to element that infiuence in human life directly and indirectly Therefore, necessity of study about noise control is increased for better labor conditions and agreeable habitat. In this paper, implementation of active mufflers using recursive LMS algorithms is presented. Analyze exhaust pipe noise of a gasoline and Diesel car and use adaptation IIR filter algorithm that stability is solidified and controled exhaust pipe noise of a car. computer simulation is performed to show the effectiveness of a proposed algorithm.

• Journal of Institute of Control, Robotics and Systems
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• v.15 no.7
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• pp.683-687
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• 2009

Track jump control is a random access strategy for short distance movement. The most common track jump scheme is a bang-bang control of a kick and brake manner. In a conventional track jump scheme, a track-following compensator is turned off during kick and brake periods, and restarted at a target track for track pull-in. The inevitable controller switching with non-zero initial condition results in undesirable transient response, and excessive overshoot in the transient response causes track pull-in failure. In this paper, a new track jump scheme is proposed for enhancing track jump stability. Instead of control switching, internal states of a track-following controller are artificially manipulated for kick and brake actions in a digital control environment. Experimental results are provided in comparison with conventional track jumps.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.9
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• pp.23-30
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• 1997

In this paper, a decision feedback recurrent neural equalization (DFRNE) scheme is proposed for adaptive equalization problems. The proposed equalizer models a nonlinear infinite impulse response (IIR) filter. The modified Real-Time recurrent Learning Algorithm (RTRL) is used to train the DFRNE. The DFRNE is applied to both linear channels with only intersymbol interference and nonlinear channels for digital video cassette recording (DVCR) system. And the performance of the DFRNE is compared to those of the conventional equalizaion schemes, such as a linear equalizer, a decision feedback equalizer, and neural equalizers based on multi-layer perceptron (MLP), in view of both bit error rate performance and mean squared error (MSE) convergence. It is shown that the DFRNE with a reasonable size not only gives improvement of compensating for the channel introduced distortions, but also makes the MSE converge fast and stable.