• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.10A
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• pp.839-846
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• 2011

Link Adaptation should select the best modulation and coding scheme (MCS) which gives the highest throughput as channel conditions vary. Several link adaptation algorithms for wireless local area network (WLAN) have been proposed but for the future WLAN systems such as 802.11n system, these algorithms do not guarantee the best performance. In this paper, we propose a new link adaptation algorithm in which an MCS level is chosen by the received SNR plus the offset value obtained from the transmission results. The performance of proposed algorithm is simulated by an IEEE 802.11n system. From the analysis, we conclude the proposed algorithm performs better than the well-known link adaptation algorithms such as auto rate fallback and general SNR-based techniques. Particularly, the proposed algorithm improves throughput when the packet error ratio (PER) is constrained for fast fading channels.

• Aerospace Engineering and Technology
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• v.8 no.1
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• pp.179-186
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• 2009

This paper describes the design and implementation for spread spectrum FTS encoder and decoder. The FTS command format is defined by 64 bit encrypted packet that contains all required information relayed between the ground and the vehicle. Encryption is accomplished using the Tripple-DES encryption algorithm in block encryption form. The proposed FTS encoder and decoder is using the Convolution Encoding and Viterbi Decoding for forward error correction. The Spread Spectrum Modulation is done using a PN code, which is 256 bit gold code. The simulation result shows that the designed FTS decoder is compatible with the designed FTS encoder.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.4B
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• pp.202-209
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• 2005

In this paper, a dynamic TCP-friendly rate control (TFRC) is proposed to adjust the coding rates according to the channel characteristics of the wireless-to-wired network consisting of wireless first-hop channel. To avoid the throughput degradation of multimedia flows traveling through wireless lint the proposed rate control system employs a new wireless loss differentiation algorithm (LDA) using packet loss statistics. This method can produce the TCP-friendly rates while sharing the backbone bandwidth with TCP flows over the wireless-to-wired network. Experimental results show that the proposed rate control system can eliminate the effect of wireless losses in flow control of TFRC and substantially reduce the abrupt quality degradation of the video streaming caused by the unreliable wireless link status.

• Journal of KIISE:Information Networking
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• v.36 no.5
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• pp.425-436
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• 2009

In recent years, the user demands have increased for multimedia service of high quality over the broadband convergence network. These rising demands for high quality multimedia service led the popularization of various user terminals and large scale display equipments, which needs a variety type of QoS (Quality of Service). In order to support demands for QoS, numerous research projects are in progress both from the perspective of network as well as end system; For example, at the network perspective, QoS guaranteeing by improving of internet performance such as Active Queue Management, while at the end system perspective, SVC (Scalable Video Coding) encoding scheme to guarantee media quality. However, existing AQM algorithms have problems which do not guarantee QoS, because they did not consider the essential characteristics of video encoding schemes. In this paper, it is proposed to solve this problem by deploying the TS- AQM (Temporal Scalability Active Queue Management) which employs the differentiated packet dropping for dependency of the temporal level among the frames, based on SVC encoding characteristics by exploiting the TID (Temporal ID) field of the SVC NAL unit header. The proposed TS-AQM guarantees multimedia service quality through video decoding reliability for SVC streaming service, by differentiated packet dropping when congestion exists.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.5
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• pp.435-446
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• 2014

Even if it is important to transmit high rate multimedia information without any transmission errors for surveillance systems, it is difficult to achieve error-free transmission due to infra-less adhoc networks. In order to reduce the transmission errors furthermore, additional signal overheads or retransmission of signals should be required, but they may lead to transmission delay. This paper represents a study on low delay and adaptive video transmission for the unmanned surveillance systems by developing system protocols. In addition, we introduce an efficient and adaptive control algorithm using system parameters for exploiting unmanned surveillance system properly over multi-channels.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8B
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• pp.1426-1437
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• 1999

The introduction of fast packet switching networks such as the ATM leads to the possibility of video services with constant quality. However, the fixed traffic descriptor(TD), which is defined by ATM Forum and ITU-T, can not simultaneously satisfy network efficiency and video quality. This paper presents fast resource reallocation protocol (FRRP) which can provide an effective compromise between network efficiency and video quality. This protocol reallocates the network resources and then adapts the traffic to the reallocated resources. In the proposed method, a video session is divided into several time-intervals according to its generated traffic and TD is calculated which well characterizes the traffic during the specific subdivided interval. In addition, a system model, where MPEG coding algorithm and network control are unified, is proposal and the performance of the FRRP in terms of cell loss rate(CLR) is evaluated though the simulation using real video traffic(Star Wars). Simulation results show that the FRRP has only 5 ~35% CLR of the fixed TD.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.81-89
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• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.