• Title/Summary/Keyword: channel coding

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Co-channel Interference Mitigation using Orthogonal Transmission Scheme for Cooperative Communication System with Decode-and-Forward Relays (복조후 전송 중계기를 이용한 협력통신 시스템에서 직교 전송 개념을 이용한 동일 채널 간섭 완화)

  • Kim, Eun-Cheol;Seo, Sung-Il;Kim, Jin-Young
    • The Journal of The Korea Institute of Intelligent Transport Systems
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    • v.9 no.1
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    • pp.34-41
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    • 2010
  • In this paper, we analyze and simulate co-channel interference (CCI) mitigation method for cooperative communication systems employing decode-and-forward relays. In co-channel interference mitigation method, A source transmits signals that are encoded by orthogonal code. Then, the receiver can distinguish its own signals form the received signals by using the orthogonal code which is already known to the receiver. The orthogonal codes applied to this paper are orthogonal Gold codes. However, we can employ other codes, which have orthogonality, as the orthogonal code. In addition, we utilize a space time block coding (STBC) scheme for enhancing the system performance by obtaining additional array gain.

Sound Quality Enhancement in MPEG Surround by Using ILD Distortion (ILD DISTORTION을 이용한 MPEG SURROUND의 음질 개선)

  • Chon, Sang-Bae;Choi, In-Yong;Sung, Koeng-Mo
    • Proceedings of the IEEK Conference
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    • 2006.06a
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    • pp.241-242
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    • 2006
  • MPEG Surround is an audio coding technology that represents multi-channel audio signal with downmixed audio signal(s) and very low bitrate side information based on Binaural Cue Coding. The side information consists of Inter-Channel Level Difference, Inter-Channel Correlation, and payloads. These two parameters are correspondent to the well-known spatial parameters in psycho-acoustics, Inter-aural Level Difference (ILD) and Inter-Aural Cross Correlation (IACC). Though ICLD is to provide perceptually equivalent ILD to the listener, however, the ILD of the original multi-channel audio signal and that of the MPEG Surround encoded signal was different. The difference between two ILD values is defined as ILD Distortion (ILDD). This paper provides how ILDD can be applied to enhance sound quality in MPEG Surround and how much ILDD is decreased.

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Channel Error Detwction and Concealment Technqiues for the MPEG-2 Video Standard (MPEG-2 동영상 표준방식에 대한 채널 오차의 검출 및 은폐 기법)

  • 김종원;박종욱;이상욱
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.21 no.10
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    • pp.2563-2578
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    • 1996
  • In this paper, channel error characteristics are investigated to alleviate the channel error propagation problem of the digital TV transmission systems. First, error propagation problems, which are mainly caused by the inter-frame dependancy and variable length coding of the MPEG-2 baseline encoder, are intensively analyzed. Next, existing channel resilient schemes are systematically classified into two kinds of schemes; one for the encoder and the other for the decoder. By comparing the performance and implementation cost, the encoder side schemes, such as error localization, layered coding, error resilience bit stream generation techniques, are described in this paper. Also, in an effort to consider the parcticality of the real transmission situation, an efficient error detection scheme for a decoder system is proposed by employing a priori information of the bit stream syntas, checking the encoding conditions at the encoder stage, and exploiting the statistics of the image itself. Finally, subsequent error concealment technique based on the DCT coefficient recovery algorithm is adopted to evaluate the performance of the proposed error resilience technique. The computer simulation results show that the quality of the received image is significantly improved when the bit error rate is as high as 10$^{-5}$ .

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Study on DC-Offset Cancellation in a Direct Conversion Receiver

  • Park, Hong-Won
    • The Bulletin of The Korean Astronomical Society
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    • v.37 no.2
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    • pp.157.2-157.2
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    • 2012
  • Direct-conversion receivers often suffer from a DC-offset that is a by-product of the direct conversion process to baseband. In general, a basic approach to reduce the DC-offset is to do simple average of the baseband signal and remove the DC by subtracting the average. However, this gives rise to a residual DC offset which degrades the performance when the receiver adopts the coding schemes with high coding rates such as 8-PSK. Therefore, more advanced methods should be additionally required for better performance. While the training sequences are basically designed to have good auto-correlation properties to facilitate the channel estimation, they may be not good for the simultaneous estimation of the channel response and the DC-offset. Also the DC offset compensation under a bad condition does not give good results due to the estimation error. Correspondingly, the proposed scheme employs the two important points. First, the training sequence codes are divided into two groups by MSE(Mean Squared Errors) for estimating the channel taps and then SNR calculated from each group is compared to predefined threshold to do fine DC-offset estimation. Next, ON/OFF module is applied for preventing performance degradation by large estimation error under severe channel conditions. The simulation results of the proposed scheme shows good performances compared to the existing algorithm. As a result, this scheme is surely applicable to the receiver design in many communications systems.

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Performance of MIMO-OFDM Systems for Underwater Communications (수중 통신 환경에서의 MIMO-OFDM 시스템 성능 분석)

  • Han, Dong-Keol;Hui, Bing;Chang, Kyung-Hi;Byun, Sung-Hoon;Kim, Sea-Moon;Lim, Yong-Kon
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2010.10a
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    • pp.597-599
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    • 2010
  • In this paper, by considering the real UWA channel environments, the measured channel data is used to generate the UWA channel model and calculate the relative parameters for underwater OFDM systems. Practical least square (LS) based channel estimation with linear interpolation are adopted to obtain the channel state information (CSI) at receiver side. As multi-input multi-output (MIMO) processing techniques, Alamouti code is implemented and evaluated to perform for space time block coding (STBC) and space frequency block coding (SFBC) for UWA OFDM systems with the MIMO configuration of $2{\times}1$, at the same time, $1{\times}2$ maximum ratio combining (MRC) is performed for the purpose of comparison. The simulation results show that, with perfect channel estimation, SFBC failed to work duo to the serious frequency selectivity of UWA channel environments. When the practical channel estimation is applied, in the case of STBC, the proposed 4-column pilot pattern gives better performance about 7dB than SISO system.

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Efficient Channel Estimation and Packet Scheduling Scheme for DVB-S2 ACM Systems (DVB-S2 ACM 시스템을 위한 효율적인 채널 예측 및 패킷 스케줄링 기법)

  • Kang, Dong-Bae;Park, Man-Kyu;Chang, Dae-Ig;Oh, Deock-Gil
    • Journal of Satellite, Information and Communications
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    • v.7 no.1
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    • pp.65-74
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    • 2012
  • The QoS guarantee for the forward link in satellite communication networks is very important because there are a variety of packets with multiplexing. Especially, the packets are processed depending on the available bandwidth in satellite network changing the wireless channel state in accordance with weather condition. The DVB-S2 increases the transmission efficiency by applying the adaptive coding and modulation (ACM) techniques as a countermeasure of rain attenuations. However, the channel estimation algorithm is required to support the ACM techniques that select the MODCOD values depending on the feedback data transmitted by RCSTs(Return Channel via Satellite Terminal) because satellite communication networks have a long propagation delay. In this paper, we proposed the channel estimation algorithm using rain attenuation values and reference data and the packet scheduling scheme to support the QoS and fairness. As a result of performance evaluation, we showed that proposed algorithm exactly predicts the channel conditions and supports bandwidth fairness to the individual RCST and guarantees QoS for user traffics.

Performance Analysis of WATM-OFDM/l6QAM System in Frequency Selective Rayleigh Fading Channel (주파수 선택성 레일리 페이텅 통신로에서 WATM-OFDM/16QAM 시스템의 성능 분석)

  • 박기식;이영춘;강영흥;김언곤;조성언
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.4 no.3
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    • pp.635-642
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    • 2000
  • We have been derived theoretically the SER's and CLP's of Wireless ATM (WATM) cells employing an OFDM/16QAM modulation scheme in wireless channel modeled as a frequency selective Rayleigh fading channel. The amount of the performance improvement of WATM- OFDM/16QAM systems adopting various coding techniques has been evaluated. In frequency selective Rayleigh fading channel, considering CLP : $10^{-3}$ as a criterion, it is observed that the performance improvement of about 14 dB is obtained in terms of $E_b/N_o$ by employing an OFDM scheme. It is also confirmed that convolutional coding technique gives better performance than the other coding techniques. Especially, when the convolutional codes are adopted to WATM-OFDM/16QAM systems, voice transmission services are sufficiently available with 5 dB of $E_b/N_o$.

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Application of Turbo Code for Digital Audio Broadcasting (DAB) System (디지털 오디오 방송을 위한 터보부호의 응용)

  • 김한종
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.13 no.2
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    • pp.176-187
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    • 2002
  • The digital Audio Broadcasting (DAB) system adopts Coded OFDM(COFDM) for channel coding. The COFDM is a combined technique of multicarrier transmission(OFDM) and punctured convolutional coding with viterbi error correction. Because the channel coding is an important topic for OFDM systems, this paper proposes a new turbo coded OFDM system that replaces the existing RCPC codec by a turbo codec without modifying the puncturing procedure and puncturing vectors defined in the standard DAB system for compatibility. The performance of a new system is compared to that of the conventional system under the frequency selective Rician fading channel and the frequency selective Rayleigh fading channel in conjunction with DAB transmission mode I suitable for the terrestrial single frequency network(SFN) broadcasting. The standard system's performance was improved with the aid of turbo codec.

An Improvement of Still Image Quality Based on Error Resilient Entropy Coding for Random Error over Wireless Communications (무선 통신상 임의 에러에 대한 에러내성 엔트로피 부호화에 기반한 정지영상의 화질 개선)

  • Kim Jeong-Sig;Lee Keun-Young
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.43 no.3 s.309
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    • pp.9-16
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    • 2006
  • Many image and video compression algorithms work by splitting the image into blocks and producing variable-length code bits for each block data. If variable-length code data are transmitted consecutively over error-prone channel without any error protection technique, the receiving decoder cannot decode the stream properly. So the standard image and video compression algorithms insert some redundant information into the stream to provide some protection against channel errors. One of redundancies is resynchronization marker, which enables the decoder to restart the decoding process from a known state in the event of transmission errors, but its usage should be restricted not to consume bandwidth too much. The Error Resilient Entropy Code(EREC) is well blown method which can regain synchronization without any redundant information. It can work with the overall prefix codes, which many image compression methods use. This paper proposes EREREC method to improve FEREC(Fast Error-Resilient Entropy Coding). It first calculates initial searching position according to bit lengths of consecutive blocks. Second, initial offset is decided using statistical distribution of long and short blocks, and initial offset can be adjusted to insure all offset sequence values can be used. The proposed EREREC algorithm can speed up the construction of FEREC slots, and can improve the compressed image quality in the event of transmission errors. The simulation result shows that the quality of transmitted image is enhanced about $0.3{\sim}3.5dB$ compared with the existing FEREC when random channel error happens.

An Efficient SVC Transmission Method in an If Network (IP 네트워크 전송에 적합한 효율적인 SVC 전송 기법)

  • Lee, Suk-Han;Kim, Hyun-Pil;Jeong, Ha-Young;Lee, Yong-Surk
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.4B
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    • pp.368-376
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    • 2009
  • Over recent years, the development of multimedia devices has meant that a wider multimedia streaming service can be supported, and there are now many ways in which TV channels can communicate with different terminals. Generally, scalable video streaming is known to provide more efficient channel capacity than simulcast video streaming. Simulcast video streaming requires a large network bandwidth for all resolutions, but scalable video streaming needs only one flow for all resolutions. On the contrary, to preserve the same video quality, SVC(Sealable Video Coding) needs a higher bit-rate than AVC(non-layered Video Coding) due to the coding penalty($10%{\sim}30%$). In previous research, scalable video streaming has been compared with simulcast video streaming for network channel capacity, in two-user simulation environments. The simulation results show that the channel capacity of SVC is $16{\sim}20%$ smaller than AVC, but scalable video streaming is not efficient because of the limit of the present network framework. In this paper, we propose a new network framework with a new router using EDE(Extraction Decision Engine) and SVC Extractor to improve network performance. In addition, we compare the SVC environment in the proposed framework with previous research on the same way subject. The proposed network framework shows a channel capacity 50%(maximum) lower than that found in previous research studies.