• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4B
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• pp.308-315
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• 2002

This paper proposes new probability models for wideband code division multiple access (W-CDMA) signals. The performance of a W-CDMA system is evaluated by calculating the average bit error rate(BER) which is derived from the probability distribution of the W-CDMA receiver output. If a probability model of the receiver output is available, the performance evaluation becomes much simpler and it enables diverse analyses of the system for channel coding and other purposes. In this paper, probability distributions of W-CDMA signals, more specifically those of the receiver output, are represented as Rayleigh and noncentral chi distribution, considering various bandwidths and channel environments. The adequacy of a probability model is verified by chi-square test of 1% significance level. The BER of the system obtained from the simulation results is compared to that obtained from the probability model to demonstrate the usefulness of the proposed models.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.281-284
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• 2003

Multi-code CDMA is an appropriate scheme for transmitting high rate data. However, dynamic range of the signal is large, and power amplifier with good linearity is required. Code select CDMA (CS/CDMA) is a variation of multi-code CDMA scheme that ensures constant amplitude transmission. In CS/CDMA input data selects multiple orthogonal codes, and sum of these selected codes are MPSK modulated to convert multi-level symbol into different carrier phases. CS/CDMA system employs level clipping to limit the number of levels at the output symbol to avoid hish density of signal constellation. In our previous work we showed that by encoding input data of CS/CDMA amplitude of the output symbol can be made constant. With this coding scheme, level clipping is not necessary and the output signal can be BPSK modulated for transmission. In this paper we show that the constant amplitude coded(CA-) CS/CDMA transmitter can be implemented using only logic gates, and the hardware complexity is very low. In the proposed transmitter architecture there is no apparent redundant encoder block which plays a major role in the constant amplitude coded CS/CDMA.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.199-203
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• 1999

This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.10
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• pp.2071-2078
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• 2009

The first frame is encoded in intra mode which generates a larger number of bits. In addition, the first frame is used for the inter mode encoding of the following frames. Thus the intial QP (Quantization Parameter) for the first frame affects the first frame as well as the following frames. Traditionally, the initial QP is determined among four constant values only depending on the bpp. In the case of low bit rate video coding, the initial QP value is fixed to 35 regardless of the output bandwidth. Although this initialization scheme is simple, yet it is not accurate enough. An accurate intial QP prediction scheme should not only depends on bpp but also on the complexity of the video sequence and the output bandwidth. In the proposed scheme, we use a linear model because there is a linear inverse proportional relationship between the output bandwidth and the optimal intial QP. Model parameters of the model are determined depending on the spatial complexity of the first frame. It is shown by experimental results that the new algorithm can predict the optimal initial QP more accurately and generate the PSNR performance better than that of the existing JM algorithm.

• Journal of Digital Contents Society
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• v.18 no.1
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• pp.63-70
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• 2017

Recently, as user feedback such as social reading become active, demand has been increased on e-book contents making which is based on user feedback. However, to reflect the user feedback onto the e-book, direct coding is required, which was difficult to the author who was not good at programming. To resolve this problem, Block assembly style feedback editor system, using Blockly was developed. This editor enables to reflect the user feedback by area allocation, component allocation, block editing, and code generating insertion, contrary to the existing way of programming realization in which direct coding was required for input, processing and output separately. This system was developed by using HTML. Javascript, PHP, and Codeigniter. Block editing is enabled to do provision and assembly of blocks by Blockly. The function of code generation & insertion allows to insert the Library function code. Through this system, the general users who are not capable of coding also can reflect feedback without doing actual coding.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.2
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• pp.439-446
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• 2011

In this paper, we proposed and implemented a dedicated hardware for Huffman coding which is a method of entropy coding to use compressing multimedia data with video coding. The proposed Huffman codec consists Huffman encoder and decoder. The Huffman encoder converts symbols to Huffman codes using look-up table. The Huffman code which has a variable length is packetized to a data format with 32 bits in data packeting block and then sequentially output in unit of a frame. The Huffman decoder converts serial bitstream to original symbols without buffering using FSM(finite state machine) which has a tree structure. The proposed hardware has a flexible operational property to program encoding and decoding hardware, so it can operate various Huffman coding. The implemented hardware was implemented in Cyclone III FPGA of Altera Inc., and it uses 3725 LUTs in the operational frequency of 365MHz

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.52-57
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• 2000

In this paper, we designed the 16 Kbps speech coder that has compatibility with CS-ACELP algorithm(G.729). The speech signal is sampled at rate of 16 KHz, divided into two narrowband signal by QMF filterbank, and decimated to rate of 8 KHz. The lower-band signal is encoded by CS-ACELP and the upper-band signal is encoded by Adaptive Transform Coding(ATC) algorithm. At the receiver, two band signals are synthesized by decoder of CS-ACELP and ATC, respectively. The reconstructed output is obtained by passing the QMF synthesis bank. The proposed wideband coder is evaluated with ITU-T G.722 coder through the Mean Opinion Score(MOS) test.

• Fashion & Textile Research Journal
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• v.22 no.1
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• pp.55-65
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• 2020

Smart photonic clothing integrates light emitting technology inside and outside of the garment and integrates it as a fashion product. It expresses digital color that radiates light outside the body that expands the functionality of the clothing as well as makes new and various attempts visually. It is also is gradually expanding into a new area of fashion. LED, one of the digital color output devices, is a light emitting device that is suitable for presenting consumer customized designs in that the patterns and colors of clothes can be modified as desired by utilizing computer technology such as program coding. LED technology that can realize various digital colors is actively applied in various industrial design fields, but there are few previous studies on smart clothes using LED color in Korean fashion fields. Therefore, this study develops a prototype of a customized LED smart photonic garment that allows the user to directly participate in the color implementation of clothing and select a digital color suitable for the desired function. The LED module was designed to be detachable from clothing and made using a 256-pixel LED matrix. Various coding patterns of the LED were designed using the coding change of Arduino program.

• Journal of Korea Society of Digital Industry and Information Management
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• v.10 no.1
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• pp.99-104
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• 2014

Nowadays, various researches fulfill in many communication engineering area for B4G (Beyond Forth Generation). Next LTE-A (Long Term Evolution Advanced), MU-MIMO (Multi-User Multi Input Multi Output) method raises to upgrade throughput performance. However, the method of user selection is not decided because of many types and discussions in MU-MIMO system. Many existing methods are powerful for enhancing performance but have various restrictions in practical implementation. Fairness problem is primary restriction in this area. Existing papers emphasis algorithm to increase sum-rate but we introduce an algorithm about dealing with fairness problem for real commercialization implementation. Therefore, this paper introduces new user selection method in MU-MIMO system. This method overcomes a fairness problem in SUS (Semiorthogonal User Selection) algorithm. We can use the method to get a similar sum-rate with SUS and a high fairness performance. And this paper uses a hybrid method with SC-SUS (Superposition Coding SUS) algorithm and SUS algorithm. We find a threshold value of optimal performance by experimental method. We show this performance by computer simulation with MATLAB and analysis that results. And we compare the results with another paper's that different way to solve fairness problem.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.2
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• pp.44-51
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• 2008

In this paper, we propose a modified clipping scheme for iterative decoding of superposition coding system by losing less power than clipping scheme. Our proposed scheme in superposition coding system shows good performance in peak-to-average power ratio(PAPR) and system performance with the same Clipping Ratio especially in low Clipping Ratio case. Finally in order to alleviate the performance degradation due to clipping noises, we combine a soft compensation algorithm that is combined with soft-input-soft-output(SISO) decoding algorithms in an iterative manner proposed by [1][2]. Simulation results show that with the proposed scheme, most performance loss can be recovered.