• 한국정보통신학회논문지
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• 제14권2호
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• pp.501-506
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• 2010

본 논문에서는 AMC(Adaptive Modulation and Coding) 기법에 MIMO(Multiple Input Multiple Output) 기법이 결합된 Adaptive-MCM(Modulation, Coding and MIMO) 시스템을 제안하고, 모의실험을 통해 그 성능을 분석한다. AMC 기법은 채널 환경에 따라서 변조방식, 코드율을 적응적으로 변화시켜 전송률과 신뢰도의 향상을 가져온다. 제안된 Adaptive-MCM 시스템은 채널 환경에 따라서 MIMO 기법을 적응적으로 활용함으로써 기존의 변조방식과 코드율의 조합에 국한된 AMC 기법보다 더욱 향상된 전송률과 에러 성능을 가져 오게 한다. 제안된 시스템의 성능 분석은 Adaptive-MCM 시스템을 적용한 시스템과 적용하지 않은 시스템의 각 전송률 결과를 바탕으로 비교 수행되었다. 실험결과, Adaptive-MCM 시스템을 적용함으로써 각 MCMS(Modulation, Coding, and MIMO Scheme) 레벨에 해당하는 조합을 고정적으로 사용하는 시스템에서 나타났던 SNR(Signal to Noise Ratio) 확보와 데이터율 증가 사이의 Trade-Off 관계 개선이 이루어졌으며, 이를 통해 평균 데이터율의 향상을 가져올 수 있었다.

• 인터넷정보학회논문지
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• 제4권5호
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• pp.1-9
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• 2003

본 논문에서는 CDMA 시스템에서 적응 변조 및 부호화 (AMC) 기법의 성능을 향상시키기 위한 방안으로서 기지국과 단말기 간에 개루프 및 폐루프 전송 다이버시티를 적용하였다. 이는 전송 다이버시티를 적용하여 수신 단의 신호 대 잡음비 (SNR)를 개선함으로써, 보다 높은 전송률을 지원하는 변조 및 부호화 레벨이 선택될 확률을 증가시키고 이에 따른 평균 전송률의 이득을 얻고자 하는 것이다. 개루프 전송 다이버시티로는 시공간 블록 부호화 기반 전송 다이버시티 (STTD) 기법을, 폐루프 전송 다이버시티로는 선택적 전송 다이버시티 (STD) 기법을 적용하였다. 모의 실험 결과, AMC 기법과 전송 다이버시티를 결합함으로써 기존 시스템에 비하여 향상된 전송률을 얻을 수 있었다. 단일 경로 레일리 페이딩 채널 환경에서 두 개의 전송 안테나를 사용한 AMC-STTD는 9dB신호 대 잡음비에 대하여 기존의 AMC 기법보다 약 250kbps의 전송률 이득을 보였다. 또한 두 개의 전송 안테나를 사용한 AMC-STD의 경우는 역시 9dB 신호 대 잡음비에서 약 420kbps의 전송률 이득이 얻어짐을 확인하였다.

• Journal of Communications and Networks
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• 제7권3호
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• pp.307-312
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• 2005

In this work, we study the performance of a serial concatenated scheme comprising a convolutional code (CC) and an orthogonal space-time block code (STBC) separated by an inter-leaver. Specifically, we derive performance bounds for this concatenated scheme, clearly quantify the impact of using a CC in conjunction with a STBC, and compare that to using a STBC code only. Furthermore, we examine the impact of performing antenna selection at the receiver on the diversity order and coding gain of the system. In performing antenna selection, we adopt a selection criterion that is based on maximizing the instantaneous signal-to­noise ratio (SNR) at the receiver. That is, we select a subset of the available receive antennas that maximizes the received SNR. Two channel models are considered in this study: Fast fading and quasi-static fading. For both cases, our analyses show that substantial coding gains can be achieved, which is confirmed through Monte-Carlo simulations. We demonstrate that the spatial diversity is maintained for all cases, whereas the coding gain deteriorates by no more than $10\;log_{10}$ (M / L) dB, all relative to the full complexity multiple-input multiple-output (MIMO) system.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• 전자공학회논문지B
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• 제31B권4호
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• pp.88-96
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• 1994

In this paper, we propose a subband pyramid image coding scheme that uses ECVQ (ntropy Constrained Vector Quantizer). In subband pyramid image coding, each subband can be encoded with a coder matched to the statistics of that particular subband, and available versions of the original image at different resolution are easily obtained. ECVQ, aiming at the minimization of the distortion for a fixed entropy of the quantizer output, is well combined with the subband pyramid image coding which yields high compression ratio and good image quality. The optimum bit allocation to each subbands corresponds to the points where the individual distortion rate curves are of particular slope, weighted to the number of samples in that subband, in designing ECVQ.

• 대한전기학회논문지:시스템및제어부문D
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• 제52권4호
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• pp.238-245
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• 2003

This paper presents a novel method for searching stable solution using DNA coding scheme. Often there are more than one solutions that satisfy the system requirements. These solutions can be viewed as extremes in multimodal function. All extremes are not the same in that some of them are using sensitive to noise or perturbation. This paper address the method that selects a solution that meets the system requirements in terms of output performance and is tolerant to the perspective noise or perturbation. A new method, called a Gradient DNA coding is proposed to achieve such objectives several numerical examples and presented and comparing DNA coding with genetic algorithm is also given.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1998년도 하계학술대회 논문집 G
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• pp.2480-2482
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• 1998

In this paper, the implementation of dual rate ADPCM using G.723 16Kbps and 40Kbps speech codec algorithm is handled. For small signals, the low rate 16Kbps coding algorithm shows the same SNR as the high rate 40Kbps coding algorithm, while the low rate 16Kbps coding algorithm shows the lower SNR than the high rate 40Kbps coding algorithm for large signal. To obtain the good trade-off between the data rate and synthesized speech quality, we applied low rate 16Kbps for the small signal and high rate 40Kbps for the large signal. Various threshold values determining the rate are tested for good trade off data rate and speech quality. Also the low pass filter effect of speech input and output devices is simulated at several cut-off frequencies. To simulation result shows the good speech quality at a low rate comparing with 16Kbps & 40Kbps.

• 한국컴퓨터정보학회논문지
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• 제21권12호
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• pp.219-226
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• 2016

In this paper, we propose an efficient repository model for online software education. The software education of app development consists of 7 stages: coding & debugging, submit, collaboration, review, validate, deployment, certification. Proposed repository model supports all 7 stages efficiently. In the coding & debugging stage, the students repeat coding and debugging of source. In the submit stage, the output of previous process such as source codes, project, and videos, are uploaded to repository server. In the collaboration stage, other students or experts can optimize or upgrade version of source code, project, and videos stored in the repository. In the review stage, mentors can review and send feedbacks to students. In the validate stage, the specialists validate the source code, project, and the videos. In the deployment stage, the verified source code, project, and videos are deployed. In the certification stage, the source code, project, and the videos are evaluated to issue the certificate.

• Journal of electromagnetic engineering and science
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• 제9권3호
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• pp.124-129
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• 2009

Although the maximum sum-rate capacity of multiple-input multiple output(MIMO) broadcast channels(BCs) can be achieved by dirty-paper coding(DPC), the results were obtained without fairness considerations in uncorrelated MIMO channels. In this paper, we propose new multiuser scheduling algorithms, which find a best user set for approaching the maximum sum-rate capacity while maintaining fairness among users. We analyze the performance of the proposed algorithms using zero-forcing dirty paper coding(ZF-DPC) in the correlated MIMO BCs for throughput and delay fairness, respectively. Numerical results demonstrate that a large time window can reduce the average throughput difference between users, but it increases head-of-line(HOL) delay jitters in the case of delay fairness.

• 대한원격탐사학회:학술대회논문집
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• 대한원격탐사학회 2004년도 Proceedings of ISRS 2004
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• pp.457-459
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• 2004

The Channel Coding Unit (CCU) is an integral component of Payload Data Transmission System (PDTS) for the Multi-Spectral Camera (MSC) data. The main function of the CCU is channel coding and encryption. CCU has two channels (I & Q) for data processing. The input of CCU is the output of DCSU (Data Compression & Storage Unit). The output of CCU is the input of QTX which modulate data for RF communication. In this paper, there are the overview, short H/W description and operation concept of CCU.