• 한국마린엔지니어링학회:학술대회논문집
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• 한국마린엔지니어링학회 2005년도 후기학술대회논문집
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• pp.65-66
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• 2005

This paper presents an adaptive satellite communication systems to adapt channel environment. High performance coding and modulation techniques are applied to poor channel condition, otherwise, bandwidth efficient coding and modulation techniques are applied to good channel condition to obtain high transmission rate.

• IEIE Transactions on Smart Processing and Computing
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• 제6권4호
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• pp.276-280
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• 2017

This paper presents computationally efficient image compression for wireless transmission of high-definition video, to adaptively utilize available channel bandwidth and improve image quality. The method indirectly predicts an unknown available channel bandwidth by monitoring encoder buffer status, and adaptively controls a quantization parameter to fully utilize the bandwidth. Experimental results show that the proposed method is robust to variations in channel bandwidth.

• IEIE Transactions on Smart Processing and Computing
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• 제5권5호
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• pp.303-309
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• 2016

This paper presents a channel-adaptive rate control algorithm for low delay video coding. The main goal of the proposed method is to adaptively use the unknown available channel bandwidth while reducing the end-to-end delay between encoder and decoder. The key idea of the proposed algorithm is for the status of the encoder buffer to indirectly reflect the mismatch between the available channel bandwidth and the generated bitrate. Hence, the proposed method fully utilizes the unknown available channel bandwidth by monitoring the encoder buffer status. Simulation results show that although the target bitrate mismatches the available channel bandwidth, the encoder efficiently adapts the given available bandwidth to improve the peak signal-to-noise ratio.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 하계종합학술대회 논문집(1)
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• pp.73-76
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• 2001

A key requirement for packet based wireless communication systems is to provide a high data rate packet service and improved throughput. To achieve a high throughput, adaptive methods for adjustment of the modulation and coding can be used. In this paper, we propose and analyze a scheme which is a combination of an adaptive modulation and coding(AMC) and transmit diversity(TD). Two different TD schemes are analysed: STTD and STD. Proposed system provides significant improvement in the average throughput.

• Journal of electromagnetic engineering and science
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• 제16권2호
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• pp.80-86
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• 2016

This paper proposes the algorithm for forward link adaptive coding and modulation (ACM) and the detailed design for a satellite communication system to improve network reliability and system throughput. In the ACM scheme, the coding and modulation schemes are changed by as much as the channel can provide depending on the quality of the communication link. To implement the forward link ACM system in the Ka-band, channel prediction and modulation/coding decision methods are proposed and simulated. The parameters of the adaptive filter predictor based on the least mean square are optimized, the minimum mean square error of the channel predictor is 0.0608 when step size and the number of filter tap are 0.0001 and 4, respectively. A test-bed is set up to verify the forward link ACM system, and a test is performed using a Ka-band satellite (i.e., Communication, Ocean, and Meteorological Satellite [COMS]). This test verifies that the ACM scheme can increase the system throughput.

• 한국콘텐츠학회논문지
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• 제8권4호
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• pp.55-62
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• 2008

AMC 기술은 고속의 데이터 전송율의 요청을 지원하는 유망한 기술이므로 4세대 이동통신 시스템의 표준으로 제안되고 있다. 본 논문에서는 단일 사용자에 초점을 맞추어 OFDM 시스템 기반 하에 AMC 기술을 SISO-OFDM과 SFBC-OFDM을 비교하여 시뮬레이션을 수행하였다. 서로 다른 성상도의 크기 하에 다중 경로 페이딩 채널을 겪는 다운링크 시스템 환경 아래 채널 용량의 측면에서 성능 분석을 하였다. 채널 상태가 예측 가능하다는 전제 하에 SFBC-OFDM이 채널 용량 측면에서 더 나은 성능을 보였다.

• Journal of Communications and Networks
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• 제7권3호
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• 한국정보과학회논문지:컴퓨팅의 실제 및 레터
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• 제16권10호
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• pp.990-994
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• 2010

모바일 와이맥스(Mobile WiMAX)는 계층적인 인코딩 기법을 사용하는 비디오 부호화 기술의 본질적인 특성을 고려하지 못하여 미디어의 품질을 보장하지 못하는 문제점을 가지고 있다. 이러한 문제점을 해결하기 위해 본 논문에서는 미디어 우선순위 기반의 채널 적응적인 스트리밍 기법인 PC-MCA(Priority-based Combining adaptive Modulation and Coding with ARQ)를 제안하였다 PC-MCA는 미디어 프레임의 우선순위에 따라 스케줄링을 하며, 무선 채널 상태와 미디어 프레임의 우선순위를 고려하여 변조 방식 및 부호화율, 그리고 ARQ 재전송 시간을 차등적으로 조절한다. 이를 통해 안정된 프레임 복호화를 제공함으로써, 멀티미디어 서비스 품질을 보장하였다.

• 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.

• 한국통신학회논문지
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• 제29권5A호
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• pp.492-501
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• 2004

압축된 동영상을 이동통신 채널과 같은 다경로 페이딩 채널을 통해서 전송할 때 전송오류에 의해서 전송 신호의 왜곡이 발생한다. 이러한 전송 오류를 줄이기 위한 한 가지 방법으로 오류정정 부호를 사용할 수 있다. 본 논문에서는 전송되는 정보의 단위인 프레임별로 채널의 상태를 예측하고 예측된 정보를 이용하여 RCPC(rate compatible punctured convolutional) 오류정정 부호의 부호화율을 적응적으로 변화시키는 방법을 제안한다. 이를 위하여 시변 페이딩 채널을 모델링하고 채널예측을 위한 3가지 방법을 제안하여 기존의 채널 예측 방법과 비교하고 성능을 분석하였다. 성능평가 결과 제안하는 적응적 오류정정 부호화 기법이 고정 부호화율을 갖는 오류정정 부호화 기법에 비해서 우수한 성능을 보임을 입증하였다.