• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.4
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• pp.506-519
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• 1999

An adaptive array antennal and a CCI canceller have been considered as techniques for cancelling Multi-User Interference(MUI) in Direct Sequence Code Division Multiple Access(DS-CDMA) system. These techniques have different problems respectively in the process of cancelling MUI as the number of users increases. For that reason, the scheme which can cancel MUI effectively by compensating for the problems of each of the techniques has been required. For the scheme, the technique to connect an adaptive array antenna and a CoChannel Interference(CCI) canceller in cascade form has been studied. In the existing study about the cascade connection method, the effect of cancelling MUI about two interference signals is analyzed, but the analysis for the quantitative BER(Bit Error Rate) improvement according to the number of users is not considered. Therefore, in this paper, we have analyzed the degree of BER performance improvement quantitatively according to the number of users by introducing the receiving system, which connects an adaptive array antenna and a CCI canceller to a DS-CDMA system in cascade form. For the method of analyzing the performance, we have performed the theoretical analysis and the simulation, considering the case of adopting only an adaptive array antenna and of cascade connection respectively, and having compared and analyzed the results. From the results, it is confirmed that in the case of adopting only an adaptive array antenna, the problems occur in the process of cancelling MUI according to the number of users and the receiving direction of interference signals, and can be compensated by the cascade connection method. In conclusion, we have known that MUI is cancelled effectively by using the cascade connection method, and the much better BER performance improvement is obtained.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.19 no.5
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• pp.47-54
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• 2019

In this paper, we consider a constant modulus (CM) based blind adaptive receiver design for downlink multi-carrier code-division multiple access (MC-CDMA) systems employing simple space-time block coding (STBC). In the paper, filter weight vectors used for the detection of the transmitted symbols are partitioned into its subvectors and then, special relations among the optimal subvectors minimizing the CM metric are derived. Using the special relations, we present a modified CM metric and propose a new blind adaptive stochastic-gradient CM algorithm (SG-CMA) by minimizing the modified CM metric. The proposed blind adaptive SG-CMA has faster convergence rate than the conventional SG-CMA because the filter weight vectors of the proposed scheme are updated in the region of satisfying the derived special relations. Computer simulation results are given to verify the superiority of the proposed SG-CMA.

• Journal of KIISE:Information Networking
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• v.35 no.3
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• pp.227-234
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• 2008

The development of wireless network technology allows high data rate seamless communication irrespective of the place and time in various emerging mobile service environment. Unlike wired networks, however, wireless networks utilize expensive limited bandwidth. MBS(Multicast Broadcast Service), which is supported by mobile WiMAX system based on IEEE802.16e, overcomes this problem using a shared downlink channel for efficiently supporting a number of users. However. the coverage and throughput of the system are significantly affected by the channel condition. In this paper we propose on MBS system employing Mobile Multi-Hop Relay(MMR) and adaptive modulation and coding(AMC) scheme. The result of NS-2 computer simulation shows that the throughput and transmission time are substantially improved by the proposed approach compared to the existing MBS system.

• Journal of Digital Contents Society
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• v.15 no.1
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• pp.93-99
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• 2014

Recently Wireless communication among short-distance devices has come to notice due to smart phone generalization recently. However, instead of setting up additional H/W, communication technology providing wireless communication based on S/W is in need due to limited availability of NFC's use. Accordingly, short-distance wireless communication technology that makes great use of mike and speaker which installed in every device draws attention. Our thesis suggests improvement of acoustic transmission speed by applying multi-channel parallel transmission and advancement of transmission rate that differed from each mike's own characteristics through optimizing adaptive threshold. The study is not only just applied in specific and limited conditions such as promoting corporation and payments system but also fast and convenient data transmit system general users-oriented.

• Journal of KIISE
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• v.43 no.2
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• pp.246-255
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• 2016

The orthogonal frequency-division multiple access (OFDMA) systems are well suited to multi-rate multicast transmission, as they allow flexible resource allocation across both frequency and time, and provide adaptive modulation and coding schemes. Unlike layered video coding, the multiple description coding (MDC) enables flexible decomposition of the raw video stream into two or more substreams. The quality of the video stream is expected to be roughly proportional to data rate sustained by the receiver. This paper describes a mathematical model of resource allocation and throughput in the multi-rate video multicast for the OFDMA wireless and mobile networks. The impact on mean opinion score (MOS), as a measurement of user-perceived quality (by employing a variety of scheduling disciplines) is discussed in terms of utility maximization and proportional fairness. We propose a pruning algorithm to ensure a minimum video quality even for a subset of users at the resource limitation, and show the optimal number of substreams and their rates can sustain.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.63-71
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• 2003

This paper suggests the packet loss recovery method to communicate in real time in the Internet. To reduce the effects of packet loss, Forward Error Correction (FEC) that adds redundant information to voice packets can be used. Adaptive Multi Rate Wideband(AMR-WB) codec which is recently selected by the Third Generation Partnership Project(3GPP) for GSM and the third generation mobile communication WCDMA system and has also been standardized in ITU-T for providing wideband speech services is used. The major cause for speech qualitly degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme if evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.

• Journal of Internet Computing and Services
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• v.21 no.4
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• pp.35-41
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• 2020

This paper proposes an algorithm to improve the compression performance of the adaptive multi-rate (AMR) speech coding using the context tree weighting (CTW) method. AMR is the voice encoding standard adopted by IMT-2000, and supports 8 transmission rates from 4.75 kbit/s to 12.2 kbit/s to cope with changes in the channel condition. CTW as a kind of the arithmetic coding, uses a variable-order Markov model. Considering that CTW operates bit by bit, we propose an algorithm that re-orders AMR data and compresses them with CTW. To verify the validity of the proposed algorithm, an experiment is conducted to compare the proposed algorithm with existing compression methods including ZIP in terms of compression ratio. Experimental results indicate that the average additional compression rate in AMR data is about 3.21% with ZIP and about 9.10% with the proposed algorithm. Thus our algorithm improves the compression performance of AMR data by about 5.89%.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.34-39
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• 2001

An adaptive multi-rate (AMR) speech coder was adopted as a standard of W-CDMA by 3GPP and ETSI. The AMR coder is based on the CELP algorithm operating at rates ranging from 12.2 kbps down to 4.75 kbps, and it is a source controlled codec according to the channel error conditions and the traffic loading. In this paper, we implement the DSP S/W of the AMR coder using OakDSPCore. The implementation is based on the CSD17C00A chip developed by C&S Technology, and it is tested using test vectors, for the AMR speech codec, provided by ETSI for the bit exact implementation. The DSP B/W requires 20.6 MIPS for the encoder and 2.7 MIPS for the decoder. Memories required by the Am coder were 21.97 kwords, 6.64 kwords and 15.1 kwords for code, data sections and data ROM, respectively. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• KIPS Transactions on Computer and Communication Systems
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• v.1 no.1
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• pp.29-34
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• 2012

Wireless multicast service can be used for video streaming service to save the network resources by sending the same popular multimedia contents to a group of users at once. For better multimedia streaming multicast service, we propose a quality-aware rate adaptation (QARA) scheme for scalable video multicast in rate adaptive wireless networks. In QARA, transmission rate is determined depending on the content's type and users' channel conditions. First, the base layer is transmitted by a low rate for high reliability. That means we provide basic service quality to all users. On the contrary, the transmission rate for enhancement layer is adapted by using channel condition feedback from a randomly selected node. So, the enhancement layer frames in a multimedia content is sent with various transmission rates. Therefore, each node can be provided with differentiated quality services. Consequently, QARA is capable of serving heterogeneous population of mobile nodes. Moreover, it can utilize network resources more efficiently. Our simulation results show that QARA outperforms utilization of the available transmission rate and reduces the data transmission time.

• Journal of Information Processing Systems
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• v.11 no.2
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• pp.295-309
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• 2015

Beacon scheduling is considered to be one of the most significant challenges for energy-efficient Low-Rate Wireless Personal Area Network (LR-WPAN) multi-hop networks. The emerging new standard, IEEE802.15.4e, contains a distributed beacon scheduling functionality that utilizes a specific bitmap and multi-superframe structure. However, this new standard does not provide a critical recipe for superframe duration (SD) allocation in beacon scheduling. Therefore, in this paper, we first introduce three different SD allocation approaches, LSB first, MSB first, and random. Via experiments we show that IEEE802.15.4e DSME beacon scheduling performs differently for different SD allocation schemes. Based on our experimental results we propose an adaptive SD allocation (ASDA) algorithm. It utilizes a single indicator, a distributed neighboring slot incrementer (DNSI). The experimental results demonstrate that the ASDA has a superior performance over other methods from the viewpoint of resource efficiency.