• ETRI Journal
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• 제39권4호
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• pp.535-545
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• 2017

This paper presents a novel K-band (18 GHz) 16-quadrature amplitude modulation (16-QAM) orthogonal frequency-division multiplexing (OFDM)-based $2{\times}2$ line-of-sight multi-input multi-output communication system. The system can deliver 356 Mbps on a 56 MHz channel. Alignment mismatches, such as amplitude and/or phase mismatches, between the transmitter and receiver antennas were examined through hardware experiments. Hardware experimental results revealed that amplitude mismatch is related to antenna size, antenna beam width, and link distance. The proposed system employs an alignment mismatch compensation method. The open-loop architecture of the proposed compensation method is simple and enables facile construction of communication systems. In a digital modem, 16-QAM OFDM with a 512-point fast Fourier transform and (255, 239) Reed-Solomon forward error correction codecs is used. Experimental results show that a bit error rate of $10^{-5}$ is achieved at a signal-to-noise ratio of approximately 18.0 dB.

• ETRI Journal
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• 제36권2호
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• pp.206-213
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• 2014

This paper presents a novel 90 GHz band 16-quadrature amplitude modulation (16-QAM) orthogonal frequency-division multiplexing (OFDM) communication system. The system can deliver 6 Gbps through six channels with a bandwidth of 3 GHz. Each channel occupies 500 MHz and delivers 1 Gbps using 16-QAM OFDM. To implement the system, a low-noise amplifier and an RF up/down conversion fourth-harmonically pumped mixer are implemented using a $0.1-{\mu}m$ gallium arsenide pseudomorphic high-electron-mobility transistor process. A polarization-division duplex architecture is used for full-duplex communication. In a digital modem, OFDM with 256-point fast Fourier transform and (255, 239) Reed-Solomon forward error correction codecs are used. The modem can compensate for a carrier-frequency offset of up to 50 ppm and a symbol rate offset of up to 1 ppm. Experiment results show that the system can achieve a bit error rate of $10^{-5}$ at a signal-to-noise ratio of about 19.8 dB.

• 한국음향학회지
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• 제20권7호
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• pp.76-82
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• 2001

본 논문에서는 7 kHz 대역의 광대역 신호 압축기를 위한 새로운 양자화 방법을 제안한다. 일반적인 광대역 신호 압축기는 입력 신호를 주파수 영역으로 변환하고 청각 모델을 적용하여 주파수 대역별로 양자화하여 Huffman 코딩하는 구조를 가진다. 그러나, 주파수 대역별로 신호의 특성이 일정하지 않으므로 모든 대역을 동일한 방법으로 양자화하면 각 주파수 대역의 특성에 적합한 양자화를 하지 못하므로 전체 압축기의 성능이 저하된다. 따라서 본 논문에서는 각 주파수 대역별로 특성을 분석하여 주파수 영역 또는 시간 영역 중에서 양자화에 효율적인 영역을 선택하여 양자화 하는 새로운 방법을 제안한다. 제안한 양자화 방법의 성능을 측정하여 ITU G.722.1 표준 압축기의 양자화 방법보다 우수한 성능을 가지는 것을 확인하였다.

• 대한전자공학회논문지SP
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• 제43권3호
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• pp.54-60
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• 2006

본 논문에서는 EVRC와 G.729A 음성부호화기를 위한 상호부호화 알고리듬을 제안한다. 다른 음성 표준을 사용하는 통신망간에 음성신호를 연동시키는 가장 간단한 방법은 이중 부/복호화 (tandem coding) 방법이다. 이 방법은 두 번의 부/복호화 과정을 거치기 때문에 많은 계산량이 요구되며 아울러 음성 지연이 발생하게 된다. 이러한 문제점들을 개선하기 위하여 상호부호화를 사용한다. 상호부호화는 LSP (Line Spectral Pair) 변환과 피치 지연 변환 그리고 지연 시간 단축 알고리듬을 통하여 수행한다. 제안된 알고리듬은 $18{\sim}22%$의 적은 계산량과 $5{\sim}10ms$의 짧은 지연으로 상호 부/복호화에 상응하는 음성 품질을 제공함을 실험을 통해 확인할 수 있다.

• Journal of Advanced Marine Engineering and Technology
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• 제38권9호
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• pp.1125-1130
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• 2014

본 논문에서는 디지털 신호처리 기술에 기반하여 주변잡음에 강건한 선박용 음향수신장치를 제안한다. 원하지 않는 주변 잡음을 억제하기 위하여 70Hz-820Hz의 통과대역을 갖는 디지털 대역통과 필터를 적용한다. 4개의 마이크로폰으로부터 입력된 음향 신호들이 디지털 대역통과 필터를 거친 후 8 방향의 방향 탐지가 가능한 방향 탐지 알고리즘을 제안한다. 또한, DSP 칩과 오디오 코덱을 이용한 프로토타입 시스템을 구현하여 본 알고리즘의 동작을 테스트한다.

• JSTS:Journal of Semiconductor Technology and Science
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• 제13권5호
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• pp.430-442
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• 2013

This paper presents a fast multi-reference frame integer motion estimator for H.264/AVC. The proposed system uses the previously proposed fast multi-reference frame algorithm. The previously proposed algorithm executes a full search area motion estimation in reference frames 0 and 1. After that, the search areas of motion estimation in reference frames 2, 3 and 4 are minimized by a linear relationship between the motion vector and the distances from the current frame to the reference frames. For hardware implementation, the modified algorithm optimizes the search area, reduces the overlapping search area and modifies a division equation. Because the search area is reduced, the amount of computation is reduced by 58.7%. In experimental results, the modified algorithm shows an increase of bit-rate in 0.36% when compared with the five reference frame standard. The pipeline structure and the memory controller are also adopted for real-time video encoding. The proposed system is implemented using 0.13 um CMOS technology, and the gate count is 1089K with 6.50 KB of internal SRAM. It can encode a Full HD video ($1920{\times}1080P@30Hz$) in real-time at a 135 MHz clock speed with 5 reference frames.

• ETRI Journal
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• 제34권5호
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• pp.649-654
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• 2012

This paper presents a novel 16-quadrature-amplitude-modulation (QAM) E-band communication system. The system can deliver 10 Gbps through eight channels with a bandwidth of 5 GHz (71-76 GHz/81-86 GHz). Each channel occupies 390 MHz and delivers 1.25 Gbps using a 16-QAM. Thus, this system can achieve a bandwidth efficiency of 3.2 bit/s/Hz. To implement the system, a driver amplifier and an RF up-/down-conversion mixer are implemented using a $0.1{\mu}m$ gallium arsenide pseudomorphic high-electron-mobility transistor (GaAs pHEMT) process. A single-IF architecture is chosen for the RF receiver. In the digital modem, 24 square root raised cosine filters and four (255, 239) Reed-Solomon forward error correction codecs are used in parallel. The modem can compensate for a carrier-frequency offset of up to 50 ppm and a symbol rate offset of up to 1 ppm. Experiment results show that the system can achieve a bit error rate of $10^{-5}$ at a signal-to-noise ratio of about 21.5 dB.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제9권1호
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• pp.242-259
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• 2015

Three-dimensional video coding is one of the main challenges restricting the widespread applications of 3D video and free viewpoint video. In this paper, a novel fractal coding algorithm with motion-vector-field-based motion estimation for depth map sequence is proposed. We firstly add pre-search restriction to rule the improper domain blocks out of the matching search process so that the number of blocks involved in the search process can be restricted to a smaller size. Some improvements for motion estimation including initial search point prediction, threshold transition condition and early termination condition are made based on the feature of fractal coding. The motion-vector-field-based adaptive hexagon search algorithm on the basis of center-biased distribution characteristics of depth motion vector is proposed to accelerate the search. Experimental results show that the proposed algorithm can reach optimum levels of quality and save the coding time. The PSNR of synthesized view is increased by 0.56 dB with 36.97% bit rate decrease on average compared with H.264 Full Search. And the depth encoding time is saved by up to 66.47%. Moreover, the proposed fractal depth map sequence codec outperforms the recent alternative codecs by improving the H.264/AVC, especially in much bitrate saving and encoding time reduction.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.258-262
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• 2009

In this paper, we propose an effective memory reduction algorithm to reduce the amount of reference frame buffer and memory bandwidth in video encoder and decoder. In general video codecs, decoded previous frames should be stored and referred to reduce temporal redundancy. Recently, reference frames are recompressed for memory efficiency and bandwidth reduction between a main processor and external memory. However, these algorithms could hurt coding efficiency. Several algorithms have been proposed to reduce the amount of reference memory with minimum quality degradation. They still suffer from quality degradation with fixed-bit allocation. In this paper, we propose an adaptive block-based min-max quantization that considers local characteristics of image. In the proposed algorithm, basic process unit is $8{\times}8$ for memory alignment and apply an adaptive quantization to each $4{\times}4$ block for minimizing quality degradation. We found that the proposed algorithm could improve approximately 37.5% in coding efficiency, compared with an existing memory reduction algorithm, at the same memory reduction rate.

• 한국통신학회논문지
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• 제34권6B호
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• pp.575-584
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• 2009

본 논문에서는 IP 기반 콜 센터에서 통화 처리 모의 실험을 위한 VoIP 트래픽 발생기를 설계하고 구현한다. 구현된 트래픽 발생기(VoIPTG)는 H.323과 SIP 기반의 VoIP 콜 생성과 G.711 코덱을 사용하는 RTP 트래픽을 발생시킴으로써 다수의 대화자들이 음성 통화하는 상황을 모의 실험을 할 수 있도록 해준다. VoIPTG를 이용하면 0.323 또는 SIP 세션 제어 프로토콜 선택, 사용자(call)수 변화, 시간 변화, 음성코덱의 선택 등 여러 가지 조합을 통해 다양한 모의실험 환경을 연출 할 수 있다. 이러한 트래픽 발생기는 IP 기반 컨택 센터의 전반적인 기능 검사 및 성능평가를 위해 유용하게 사용될 수 있으며, 특히 IP 기반 녹취 시스템의 성능 평가를 위해서 필수적이다.