• Title/Summary/Keyword: Adaptive transmission

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Implementation of Co-Channel Radio Relay System and Its Performance Evaluation with Synchronous Digital Hierarchy (동기식 디지틀 계위의 동일채널 무선 전송장치구현 및 성능분석)

  • 서경환
    • Journal of the Korean Institute of Telematics and Electronics D
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    • v.35D no.11
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    • pp.10-22
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    • 1998
  • In this paper, as a way of improving the availability and spectral efficiency of radio relay system, a co-channel radio relay system based upon the synchronous digital hierarchy is developed and its performance measured by 64-QAM with a never-seen multi-purpose ASIC chip is illustrated. This system provides a couple of transmission capacity compared with alternative channel arrangement. By adopting a powerful complex 13-tap adaptive time domain equalizer and cross-pol interference canceller, improvement of more than 1.5 ~ 2.0 dB in signature is obtained in comparison to 9 or 11-tap adaptive time domain equalizer, and about 22.5 dB in improvement factor of cross-pol interference canceller is achieved at C/N of 24.5 dB. In addition, digital filter makes it possible to optimize the occupied bandwidth by selecting an appropriate roll-off factor externally. It is expected that co-channel radio relay system with the powerful multi-purpose ASIC chip plays a key role in creating a value-added product, reliability, and reducing the outage time.

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Speech Packet Transmission Using the AMR-WB Coder with FEC (FEC기능을 추가한 AMR-WB 음성 부호화기를 이용한 음성 패킷 전송)

  • 황정준;이인성
    • 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.

Improvement of AMR Data Compression Using the Context Tree Weighting Method (Context Tree Weighting을 이용한 AMR 음성 데이터 압축 성능 개선)

  • Lee, Eun-su;Oh, Eun-ju;Yoo, Hoon
    • 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%.

A Performance Comparison of VSCA and VSDA Adaptive Equalization Algorithm using Distance Adjusted Approach in QAM Signal (QAM 신호에서 Distance Adjusted Approach를 이용한 VSCA와 VSDA 적응 등화 알고리즘의 성능 비교)

  • Lim, Seung Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.15 no.1
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    • pp.139-145
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    • 2015
  • This paper compare the VSCA (Variable stepsize Square Contour Algorithm) and VSDA (Variable stepsize Square contour Decision directed Algorithm) adaptive equalization algorithm that is used for the minimization of the intersymbol interference which occurs in the time dispersive channel for the transmission of 16-QAM signal.. In the SCA, it is possible to compensates the amplitude and phase in the received signal that are mixed with the intersymbol interference by the constellatin dependent constant by using the 2nd order statistics of the transmitted signal. But in the VSCA and VSDA, it is possible to the improving the equalization performance by varing the stepsize using the concept of distance adjusted approach for constellation matching. We compare the performance of the VSCA and VSDA algorithm by the computer simulation. For this, the equalizer output signal constellation, residual isi, maximum distortion and MSE were used in the performace index. As a result of computer simulation, the VSCA algorithm has better than the VSDA in every performance index.

Unproved AMC-MIMO Multiplexing Systems Using Selection Transmit Diversities (선택적 전송다이버시티를 사용한 개선된 AMC-MIMO 다중화시스템)

  • Kang, Min-goo
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.7 no.4
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    • pp.652-657
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    • 2003
  • In this paper, Adaptive Modulation and Coding (AMC) is combined with Multiple Input Multiple Output (MIMO) multiplexing to improve the throughput performance of AMC. In addition, a system that adopts Selection Transmit Diversity (STD) in the AMC-MIMO multiplexing system is proposed. The received SNR is improved by adopting STD techniques. And it increases probability of selecting MCS (Modulation and Coding Scheme) level that supports higher data rate. This leads to an increased throughput of the AMC-MIMO system. STD in our simulation selects 2 transmission antennas from 4 antennas and AMC-MIMO multiplexing process operates with the selected antennas. The computer simulation is performed in flat Rayleigh fading channel. The results show that the proposed system achieves a gain of 1Mbps over the AMC-MIMO multiplexing system with the same number of antennas at 15dB SNR.

An Adaptive Cell Selection Scheme for Ultra Dense Heterogeneous Mobile Communication Networks (초밀집 이종 이동 통신망을 위한 적응형 셀 선택 기법)

  • Jo, Jung-Yeon;Ban, Tae-Won;Jung, Bang Chul
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.19 no.6
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    • pp.1307-1312
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    • 2015
  • As smart-phones become popular, mobile data traffic has been dramatically increasing and intensive researches on the next-generation mobile communication network is in progress to meet the increasing demand for mobile data traffic. In particular, heterogeneous network (HetNet) is attracting much interest because it can significantly enhance the network capacity by increasing the spatial reuse with macro and small cells. In the HetNet, we have several problems such as load imbalance and interference because of the difference in transmit power between macro and small cells and cell range expansion (CRE) can mitigate the problems. In this paper, we propose a new cell selection scheme with adaptive cell range expansion bias (CREB) for ultra dense HetNet and we analyze the performance of the proposed scheme in terms of average cell transmission rate through system-level simulations and compare it with those of other schemes.

Autonomic Multimedia Transcoding Algorithm for Seamless Streaming Media of Mobile Clients (이동 단말에서 끊임없는 스트리밍 미디어를 위한 오토노믹 멀티미디어 트랜스코딩 알고리즘)

  • Han, Woo-Ram;Heo, Nan-Sook;Park, Chong-Myung;Seo, Dong-Mahn;Jung, In-Bum
    • Journal of KIISE:Computing Practices and Letters
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    • v.13 no.5
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    • pp.260-270
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    • 2007
  • Owing to the improved wireless communication technologies, it is possible to provide streaming service of multimedia with PDAs and mobile phones in addition to desktop PCs. Since mobile client devices have low computing power and low network bandwidth due to wireless network, the transcoding technology to adapt media for mobile client devices considering their characteristics is necessary. However, since mobile clients use wireless network and the wireless network have less stable bandwidth according to distance from AP and environments, it is hard to support stable QoS to mobile clients. In this paper, we propose a dynamic bit rate control method for transcoding in order to supprot stable QoS of streaming media service to mobile clients via wireless network. The proposed method is shown to serve seamless streaming media service with adaptive bit rate control according to state of wireless network in real time and to reduce transmission failure from experiment.

An Adaptive Neighbor Discovery for Tactical Airborne Networks with Directional Antenna (지향성 안테나 기반 공중전술네트워크를 위한 적응적 이웃노드 탐색기법)

  • Lee, Sung-Won;Yoon, Sun-Joong;Ko, Young-Bae
    • Journal of KIISE:Information Networking
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    • v.37 no.1
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    • pp.1-7
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    • 2010
  • Network Centric Warfare(NCW) is becoming a prominent concept in the current trend of warfare. To support high quality communication between strategic/tactical units in the concept of NCW, Tactical Airborne Networks are likely to be constructed in the near future to take part in the NCW. In these Tactical Airborne Networks with dynamic topology variations due to very high mobility of participants nodes, more efficient and reliable neighbor discovery protocols are needed. This paper presents the adaptive HELLO message scheduling algorithm for Tactical Airborne Network using directional antennas. The purposed algorithm can reduce the overhead of periodic HELLO message transfer, while guaranteeing successful data transmission. We concluded a mathematical analysis and simulation studies using Qualnet 4.5 for evaluation the performance and efficiency of the proposed scheme.

Performance Analysis of Target Adapted RED Algorithm on TCP/IP based GEO Satellite Communication Network (TCP/IP 기반의 정지 위성 궤도 통신망에서 TARED 알고리즘 성능 분석)

  • 서진원;김덕년
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.6A
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    • pp.667-667
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    • 2004
  • We must design the buffer algorithm that protects traffic congestion and decreasing throughput at satellite communication network. It is important that buffer algorithm is satisfied with the good performance of transmission packet, responsibility of many connecting traffic and the QOS for connecting character. Old buffer algorithms are not the suitable algorithms when we have the satellite communication network environment. RED buffer algorithm is proposed by Floyd. It has a better performance than old buffer algorithm. But this algorithm is not well adapted a number of connecting TCP packet and changing network, so this algorithm has a bad performance on satellite communication network that is many of connecting user at same time. This paper propose the TARED(Target Adaptive RED). It has a good performance, adaptation and stability on satellite communication network and has not overflow and underflow of the buffer level.

Performance Analysis of Target Adapted RED Algorithm on TCP/IP based GEO Satellite Communication Network (TCP/IP 기반의 정지 위성 궤도 통신망에서 TARED 알고리즘 성능 분석)

  • 서진원;김덕년
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.6A
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    • pp.666-676
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    • 2004
  • We must design the buffer algorithm that protects traffic congestion and decreasing throughput at satellite communication network. It is important that buffer algorithm is satisfied with the good performance of transmission packet, responsibility of many connecting traffic and the 005 for connecting character. Old buffer algorithms are not the suitable algorithms when we have the satellite communication network environment. RED buffer algorithm is proposed by Floyd. It has a better performance than old buffer algorithm. But this algorithm is not well adapted a number of connecting TCP packet and changing network, so this algorithm has a bad Performance on satellite communication network that is many of connecting user at same time. This paper Propose the TARED(Target Adaptive RED). It has a good performance, adaptation and stability on satellite communication network and has not overflow and underflow of the buffer level.