• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.198-201
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• 2003

In this paper, we analyze high-rate orthogonal frequency division multiplexing(OFDM) system using time switched time diversity(TSTD) method based on IEEE 802.11a. First, we modeled the transmitter and receiver of OFDM system based on IEEE 802.11a. Then, we applied the TSTD to OFDM system. Finally we analyzed the performance of OFDM system using TSTD method by simulation over the 5GHz realistic channel model, where channel coding and channel equalizer over multipath environment are also applied.

• Journal of IKEEE
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• v.2 no.2 s.3
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• pp.185-193
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• 1998

Fractal-based monochrome image coding method can be easily applied for color image compression by splitting the color image into different primary spectral channels such as RGB, YIQ or $YC_bC_r$, and encoding each channel independently According to this method, it needs to repeat the fractal coding for each channel, so it have the problem of encoding time. In this paper, a fractal-based coder for color still image is proposed which features the enhancement of compression rate and the reduction of coding time. As the result of the experiment where the proposed algorithm is applied far color images, the compression rate is enhanced by 28 : 1 above with average PSNR value $28{\sim}29[dB]$, do not lossless encoding process using JPEG. And the encoding time is reduced by maximum 11.5 %.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2001.11b
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• pp.109-113
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• 2001

최근 들어 디지털 이동통신 시스템은 음성 통신 뿐만 아니라 멀티미디어 통신에도 그 적용범위를 확장시키고 있다. 그러나 이러한 시스템에서 한정된 채널 대역폭의 사용은 멀티미디어 정보의 대용량성을 고려할 때, 그 적용범위를 심각하게 축소시키는 가장 큰 제한요소이다. 일반적인 통신 시스템의 소스 부호화기는 채널 잡음을 고려하지 않고 설계되며, 채널 부호화기는 소스 신호의 특성과 무관하게 채널 환경의 극복에만 중점을 두고 설계된다. 그러나 대역폭 제한적인 통신 환경에서 보다 효율적인 대역폭 사용을 위해서는 채널 환경에 따라 소스 부호율과 채널 부호율을 가변적으로 운용하여야 한다. 본 논문에서는 영상을 부대역 부호화하여 각 부대역 영상이 원래의 영상 재구성에 기여하는 중요도에 따라 한정된 채널 자원을 최적으로 할당하는 공동 소스/채널 부호화(Joint Source-channel Coding)에 관하여 연구하였다. 부대역 영상의 소스 부호화로는 DPCM과 PCM을 사용하였고, 채널 부호화는 TCM 부호화기를 사용하였으며 상이오류보호(Unequal Error Protection)를 위해 3가지 채널 부호율에 따라 각각 TCQPSK, TC8PSK, TC16PSK 변조방식을 사용하였다. 모의실험에 사용된 채널 환경은 랜덤잡음 환경과 이동수신의 경우에 발생하는 Rayleigh 페이딩 환경을 고려하였으며, 각 환경에서의 채널 SNR에 따라 동일오류보호(Equal Error Protection) 시스템과 상이오류보호(Unequal Error Protection) 시스템의 성능을 비교하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2465-2473
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• 2012

Recently, as one of low complexity video encoding methods, DVC (Distributed Video Coding) scheme has been actively studied. Most of DVC schemes exploit feedback channel to achieve better coding performances, however, this causes these schemes to have high decoding delay. In order to overcome these, this paper proposes a new fast DVC decoding method using parity-bit request model, which can be obtained by using bit-error rate, sent by encoder with motion vector, which is transmitted through feedback channel by decoder after generating side information. Through several simulations, it is shown that the proposed method improves greatly the decoding speed, compared to the conventional schemes.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.20 no.9
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• pp.883-889
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• 2009

Since the SRRs(Short Range Radar) load on the GVESs(Ground Vehicle Equipment System) are operated within several hundred meters, it is important to suppress the interferences between the SRRs. These interference are reduced using the frequency separation between channels, however this method isn't a perfect solution owing to a difficulty of a realization for the interference suppression. Thus, a pulse-to-pulse coding used to suppress the interference fundamentally is proposed in this paper. The concept, the application method and the test results of the proposed method have been described in this paper.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.6A
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• pp.782-790
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• 1999

In this paper, we propose an adaptive trellis-coded Multiple Quadrature Amplitude Modulation(ATCMQAM) applying MQAM as the modulation scheme in the conventional ATCMPSK (Adaptive trellis-Coded Multiple Phase Shift Keying) for slowly varying Rayleigh fading channels. The proposed system adaptively controls the coding rate combined with modulation level of pragmatic approach to trellis-coded modulation according to the instantaneous fading channel conditions, and employs MQAM as modulation scheme.Results by computer simulation show that the proposed adaptive model using MQAM can realize higher quality transmission with the improvement more than 0.5 ~ 1bit in average bit rate. And there is a coding gain of 2-5dB, depending on the high SNR value, compared with the conventional adaptive model employing MPSK.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38C no.11
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• pp.1078-1083
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• 2013

At the ubiquitous age, applications of Wireless Personal Area Network (WPAN) technology using LEDs are in progress. However, visible light communication using LED have weakness, which deteriorates performance of communication. To reduce information losses, which is caused by optical noise, such as incandescent lamps, fluorescent lamps, sunbeam etc., proposed channel coding scheme, double binary turbo codes. In this paper, encoding scheme of the proposed system is described and simulation results are analyzed. We had expected improved performance by using double binary turbo codes. Finally, performances of the proposed system came up to our expectations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.3C
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• pp.408-418
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• 2004

In this paper, we propose a novel scalable video coding algorithm based on adaptively weighted multiple reference frame method. To improve the coding efficiency in the enhancement layer, the enhancement frame is predicted by the sum of adaptively weighted double motion compensated frames in the enhancement layer and the current frame in the base layer, according to the input video characteristics. By employing adaptive reference selection scheme at the decoder, the proposed method reduce the drift problem significantly. From the experimental results, the proposed algorithm shows more than 1.0 ㏈ PSNR improvement, compared with the conventional scalable H.263+ for various packet loss rate channel conditions.

• Journal of Communications and Networks
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• v.9 no.2
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• pp.118-127
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• 2007

We propose a link adaptation and selection method for the links constituting an orthogonal frequency division multiplexing (OFDM) based wireless relay network. The proposed link adaptation and selection method selects the forwarding, modulation, and channel coding schemes providing the highest end-to-end throughput and decides whether to use the relay or not. The link adaptation and selection is done for each sub-channel based on instantaneous signal to interference plus noise ratio (SINR) conditions in the source-to-destination, source-to-relay and relay-to-destination links. The considered forwarding schemes are amplify and forward (AF) and simple adaptive decode and forward (DF). Efficient adaptive modulation and coding decision rules are provided for various relaying schemes. The proposed end-to-end link adaptation and selection method ensures that the end-to-end throughput is always larger than or equal to that of transmissions without relay and non-adaptive relayed transmissions. Our evaluations show that over the region where relaying improves the end-to-end throughput, the DF scheme provides significant throughput gain over the AF scheme provided that the error propagation is avoided via error detection techniques. We provide a frame structure to enable the proposed link adaptation and selection method for orthogonal frequency division multiple access (OFDMA)-time division duplex relay networks based on the IEEE 802.16e standard.

• 한국정보통신설비학회:학술대회논문집
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• 2005.08a
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• pp.417-421
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• 2005

In the recent digital communication systems, the performance of Turbo Code using the error correction coding depends on the interleaver influencing the free distance determination and the recursive decoding algorithms that is executed in the turbo decoder. However, performance depends on the interleaver depth that needs many delays over the reception process. Moreover, turbo code has been known as the robust coding methods with the confidence over the fading channel. International Telecommunication Union(ITU) has recently adopted it as the standardization of the channel coding over the third generation mobile communications(IMT-2000). Therefore, in this paper, we proposed the interleaver that has the better performance than existing block interleaver, and modified turbo decoder that has the parallel concatenated structure using MAP algorithm. In the real-time voice and video service over third generation mobile communications, the performance of the proposed two methods was analyzed and compared with the existing methods by computer simulation in terms of reduced decoding delay using the variable decoding method over AWGN and fading channels for CDMA environments.