• 한국통신학회논문지
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• 제30권4C호
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• pp.200-206
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• 2005

적응 전송 기법은 변조방식, 부호율, 전력 등의 전송 매개변수를 채널의 상태에 적응시켜 무선 통신시스템의 성능을 향상시키는 기법이다. OFDM (Orthogonal frequency division multiplexing) 시스템에서는 이러한 적응기법을 부반송파별로 적용시킬 수 있다. 이 논문에서는 각 부채널의 상태에 따라 부반송파에 최적의 부호율을 적응시키는 적응부호율 OFDM 시스템을 고려한다. 성능 분석을 통해 적응부호율 OFDM 시스템이 비트오류율 $10^{-6}$에서 고정부호율 OFDM 시스템에 비해 $3\sim6$ dB의 신호 대 잡음비 이득 또는 $30\sim50\%$의 데이터 전송률 증가를 얻을 수 있음을 보인다.

• 한국통신학회논문지
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• 제17권2호
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• pp.121-131
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• 1992

포스트 필터링(adaptive postfiltering)을 수신단의 마지막에세 수행 하였다. 첫째는, SBC 시스템의 대역 필터를 QMF(Quadrature Mirror Filter) 대신 GQMF(Generalized QMF)를 사용하여 성능을 향상시켰고, 둘째는, 각 대역에 적응 비트 할당을 함으로써 음질의 향상뿐 아니라 variable rate 부호화할 수도 있었다. 세번째로는 APCM(Adaptive Plulse Code Modulation)과 ADPCM(Adaptive Differential Pulse Code Modulation)을 부호화기로 사용하여 각각의 성능을 평가 한 결과, SB-APCM 의 성능이 우수하였다.또한, 수신단의 마지막에서 적응 포스트 필터링을 수행하여 부호화된 음성의 음질을 개선할 수 있었다. 본 논문에서는 두가지의 적응 포스트 필터링 기법을 제안하였는데 낮은 복잡성을 가지고도 부호화된 음성에서 상당량의 잡음 감쇄를 이룰 수 있었다.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 1995년도 학술대회
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• pp.59-61
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• 1995

An adaptive code rate change scheme in DS-SSMA systems is proposed. In the proposed scheme, the error correcting code rate is changed according to the channel state, the effective number of users. The channel state is estimated based on retransmission requests. The criterion for the change of the code rate is to maximize the throughput under given error bound.

• 한국통신학회논문지
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• 제27권3B호
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• pp.251-258
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• 2002

이 논문에서는 여러 반송파 직접수열 부호분할 다중접속 시스템에서 적응 부호율 길쌈부호화 방법을 살펴본다. 여러 가지 부호율을 쉽게 다를 수 있고 부호기와 복호기 얼개가 간단하도록 부호율 호환 구멍 뚫은 길쌈부호를(rate compatible punctured convolutional code: RCPC code) 쓴다. 데이터 처리량이 가장 많아지도록, 신호 대간섭과 잡음비 추정을 바탕으로 하는 적응 부호율 시스템을 제안한다. 제안한 적응 부호율 여러 반송파 직접수열부호분할 다중접속 시스템을 쓰면 주파수 대역 효율을 높이고 주파수 다양성을 얻을 수 있음을 보인다.

• Journal of Communications and Networks
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• 제7권4호
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• pp.462-470
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• 2005

We propose a multi-rate and multi-BEP transmission scheme using adaptive overlapping pulse-position modulator (OPPM) and optical power controller in optical code division multiple access (CDMA) networks. The proposed system achieves the multi-rate and multi-BEP transmission by accommodating users with different values of OPPM parameter and transmitted power in the same network. The proposed scheme has advantages that the system is not required to change the code length and number of weight depending on the required bit rate of a user and the difference of bit rates does not have so much effect on the bit error probabilities (BEPs). Moreover, the difference of transmitted powers does not cause the change of bit rate. We analyze the BEPs of the four multimedia service classes corresponding to the com­binations of high/low-rates and low/high-BEPs and show that the proposed scheme can easily achieve distinct differentiation of the service classes with the simple system configuration.

• Journal of Electrical Engineering and information Science
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• 제1권1호
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• pp.23-28
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• 1996

In this series of two papers, adaptive code rate change schemes in DS-SSMA systems are proposed. In the proposed schemes the error correcting code rate is changed according to the channel states. Two channel states having significant effects on the bit error probability are considered: one is the effective number of users considered in Part 1, and the other is the fading environment considered in Part 2. These channel states are estimated based on retransmission requests. The criterion for the change of the code rate is to maximize the throughput under given error bound. Simulation results show that we can transmit maximum amount of information if we change the code rate based on the channel states.

• Journal of Communications and Networks
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• 제7권3호
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• pp.258-262
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• 2005

An adaptive beamforming system based on code filtering and differential correlation approaches is proposed. The differential correlation method was originally proposed for time delay estimation of direct sequence code division multiple access (DS-CDMA) systems under near-far ratio conditions and the code filtering correlation algorithm, on the other hand, was proposed for array response estimation in DS-CDMA systems under perfect power control. In this paper, by combining differential correlation concept with the code filtering beamforming technology, an accurate estimate of the beam forming weights and an enhanced performance of DS-CDMA systems under sever near-far ratio conditions is achieved. The system performance in terms of beam pattern and bit-error-rate (HER) shows that the proposed adaptive beamformer outperforms the conventional code filtering correlation technique.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2008년도 하계종합학술대회
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• pp.157-158
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• 2008

This paper examines an adaptive coding scheme for flat fading channels to maximize the average code rate of a coded system. The proposed adaptation technique is employed by using the required free distance of a rate compatible code depending on the channel realization. First, the system will calculate the required free distance based on the instantaneous channel gain. Based on this channel gain we will select a set of convolution code to optimize the code rate with a certain performance requirement. Simulation results show that our proposal can achieve a higher code rate.

• Journal of Electrical Engineering and information Science
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• 제1권1호
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• pp.17-22
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• 1996

Adaptive code rate change schemes in DS-SSMA systems are proposed. In the proposed schemes, the error correcting code rate is changed according to the channel states. Two channel states having significant effects on the bit error probability are considered: one is the effective number of users, and the other is the fading environment. These channel states are estimated based on retransmission requests. The criterion for the change of the code rate is to maximize the throughput under given error bound.

• 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.