• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.4
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• pp.1-9
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• 2008

This paper presents a rapid watermarking algorithm based on DC image, which provides a resilience to geometric distortion. Our proposed scheme is based on $8{\times}8$ block DCT that is widely used in image/video compression techniques (e.g., JPEG and MPEG). In particular, a DC image is analyzed by DWT to embed a watermark. To overcome a quality degradation caused by a watermark insertion into DC components, we discern carefully the intensity and amount of watermark along the different subbands of DWT. Note that the proposed technique supports a high throughput for a real-time watermark insertion and extraction by relying on a partial decoding (i.e., DC components) on $8{\times}8$ block DCT domain. Experimental result shows that the proposed watermarking scheme significantly reduces computation time of 82% compared with existing DC component based algorithm and yet provides invariant properties against various attacks such as geometric distortion and JPEG compression, etc.

• Journal of Broadcast Engineering
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• v.1 no.1
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• pp.67-73
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• 1996

In general, the high quality audio coding(HQAC) has the structure of the convertional data compression techniques combined with moodels of human perception. The primary auditory characteristic applied to HQAC is the masking effect in the spectral domain. Therefore spectral techniques such as the subband coding or the transform coding are widely used[1][2]. However no effort has yet been made to apply the temporal masking effect and temporal redundancy removing method in HQAC. The audio data compression method proposed in this paper eliminates statistical and perceptual redundancies in both temporal and spectral domain. Transformed audio signal is divided into packets, which consist of 6 frames. A packet contains 1536 samples($256{\times}6$) :nd redundancies in packet reside in both temporal and spectral domain. Both redundancies are elminated at the same time in each packet. The psychoacoustic model has been improved to give more delicate results by taking into account temporal masking as well as fine spectral masking. For quantization, each packet is divided into subblocks designed to have an analogy with the nonlinear critical bands and to reflect the temporal auditory characteristics. Consequently, high quality of reconstructed audio is conserved at low bit-rates.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6C
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• pp.637-644
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• 2006

This paper presents performance and implementation comparisons of standard and fast DCT algorithms that are commonly used for subband filter bank in MPEG audio coders. The comparison is made according to the architectural difference of the implementation hardware. Fast DCT algorithms are known to have much less computational complexity than the standard method that involves computing a vector dot product of cosine coefficient. But, due to structural irregularity, fast DCT algorithms require extra cycles to generate the addresses for operands and to realign interim data. When algorithms are implemented using DSP processors that provide special operations such as single-cycle MAC (multiply-accumulate), zero-overhead nested loop, the standard algorithm is more advantageous than the fast algorithms. Also, in case of the finite-precision processing, the error performance of the standard method is far superior to that of the fast algorithms. In this paper, truncation errors and algorithmic suitability are analyzed and implementation results are provided to support the analysis.

• Journal of IKEEE
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• v.23 no.3
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• pp.800-804
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• 2019

This paper presents a dual-loop sub-sampling digital PLL for a 2.4 GHz IoT applications. The PLL initially performs a divider-based coarse lock and switches to a divider-less fine sub-sampling lock. It achieves a low in-band phase noise performance by enabling the use of a high resolution time-to-digital converter (TDC) and a digital-to-time converter (DTC) in a selected timing range. To remove the difference between the phase offsets of the coarse and fine loops, a phase offset calibration scheme is proposed. The phase offset of the fine loop is estimated during the coarse lock and reflected in the coarse lock process, resulting in a smooth transition to the fine lock with a stable fast settling. The proposed digital PLL is designed by SystemVerilog modeling and Verilog-HDL and fully verified with simulations.

• The Journal of the Acoustical Society of Korea
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• v.27 no.5
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• pp.256-261
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• 2008

In this paper, we propose a novel approach to improve the performance of minima controlled recursive averaging (MCRA) which is based on the conditional maximum a posteriori criterion. A crucial component of a practical speech enhancement system is the estimation of the noise power spectrum. One state-of-the-art approach is the minima controlled recursive averaging (MCRA) technique. The noise estimate in the MCRA technique is obtained by averaging past spectral power values based on a smoothing parameter that is adjusted by the signal presence probability in frequency subbands. We improve the MCRA using the speech presence probability which is the a posteriori probability conditioned on both the current observation the speech presence or absence of the previous frame. With the performance criteria of the ITU-T P.862 perceptual evaluation of speech quality (PESQ) and subjective evaluation of speech quality, we show that the proposed algorithm yields better results compared to the conventional MCRA-based scheme.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11a
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• pp.535-538
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• 2002

InfiniBand is a switched-fabric architecture for next generation I/O systems and data centers. The InfiniBand Architecture (IBA) promises to replace bus-based architectures, such as PCI, with a switched-based fabric whose benefits include higher performance, higher RAS (reliability, availability, scalability), and the ability to create modular networks of servers and shared I/O devices. The switched-fabric InfiniBand consists of InfiniBand subnets with channel adapters, switches, and routers. In order to fully grasp the operational characteristics of InfiniBand architecture (IBA) and use them in ongoing design specification, simulation of subnet management of IBA is inevitable. In this paper, thus, we implement an IBA simulator and test some practical sample networks using it. The simulator shows the flow of operation by which the correctness and effectiveness of the system can be verified.

• Journal of the Korean Institute of Intelligent Systems
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• v.20 no.4
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• pp.522-527
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• 2010

In this paper, a new MBHF(Morphology-Based Homomorphic filter) is presented to enhance contrast in mammographic images. The MBH filtering is performed based on the morphological sub-bands, in which an image is morphologically decomposed. The filter is designed to have optimal gain and structuring element in each sub-band through differential evolution. Experimental results show that the proposed method improves the contrast in mammographic images such that an evaluation criterion, WPSNR(Weighted Peak Signal to Noise Ratio) which takes into account human visual system is increased compared with a wavelet-based Homomorphic filter.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.17 no.7
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• pp.334-341
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• 2016

In this paper, the study of a waveguide aperture-coupled feed-structured antenna has been conducted for the purpose of applying it to a wireless back-haul system sufficient for high-capacity gigabits-per-second data rates. For this study, a $32{\times}32$ waveguide slot sub-array antenna with a corporate-feed structure was designed and produced. Also, this antenna is used at 57 GHz to 66 GHz in the V-band. The construction of the antenna is a laminated form with radiating parts (outer groove and slot, cavity), a coupled aperture, and feeds in each. The antenna was designed with HFSS, which is based on 3D-FEM, produced with aluminum processed by a precision-controlled milling machine, and assembled after a silver-plating process. The measurement result from analysis of the characteristics of the antenna shows that return loss is less than -12 dB, VSWR < 2.0, and a wide bandwidth ranges up to 16%. An overall first side lobe level is less than -12.3 dB, and a 3 dB beam width is narrow at about $1.85^{\circ}$. Also, antenna gain is 38.5 dBi, offering high efficiency exceeding 90%.

• Journal of Satellite, Information and Communications
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• v.2 no.2
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• pp.41-47
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• 2007

The first geostationary satellite made by Korea, COMS, has the three different payload ; Meteorological sensor, Oceanographic sensor and Ka-band communication payload. There are Meteorological & Ocean Data Communication Subsystem(MODCS) and Telemetry, Command and Ranging Subsystem(TC&R) as other RF radiation sources. MODCS transmits and receives Meteo and Ocean measurement data from/to earth using L-band and TC&R using S-band. The Ka-band communication payload will provide high-speed multimedia services and communication services for natural disaster such as prediction, prevention, and recovery services in the government communications network.Ka-band beacon is for the earth antenna pointing and the experiment of rain fading. This paper gives the analysis results about the mutual radiation effect on Ka-band communication payload, Ka-band beacon, MODCS and TC&R. Up/Down link power and coupling factor including the geometrical position and distance of antenna, filter rejection and degradation factor due to the different polarization are considered. The results show MODCS and TC&R are compatible for Ka-band communication payload and Ka-band beacon does not interfere with MODCS and TC&R normal operation.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.2
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• pp.33-41
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• 2001

Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.