• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.3
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• pp.167-167
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• 2004

The system combined layered space-time processing and space-time trellis codes (STTC) provide high transmission rate as well as diversity and coding gain without bandwidth expansion. In this paper, two layered receiver structures are proposed. One is the LSTT-MMSE in which received bit streams are decoupled by interference nulling and then decoded by separate STTC decoders. The decoded outputs are cancelled from the received signal before advancing to the next layer detection. The other is LSTT-Whitening employing whitening rather than nulling. The receiver employing whitening process shows several advantages on diversity gain and the required number of receive antennas compare to the convolutional coded space-time processing. The proposed receivers use different decoding order scheme according to each interference suwression. The (4, 3) LSTT-Whitening receiver still achieves 1㏈ gain over the (4, 4) LSTT-MMSE and the (4, 4) coded layered space-time processing.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.3
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• pp.9-14
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• 2004

The system combined layered space-time processing and space-time trellis codes (STTC) provide high transmission rate as well as diversity and coding gain without bandwidth expansion. In this paper, two layered receiver structures are proposed. One is the LSTT-MMSE in which received bit streams are decoupled by interference nulling and then decoded by separate STTC decoders. The decoded outputs are cancelled from the received signal before advancing to the next layer detection. The other is LSTT-Whitening employing whitening rather than nulling. The receiver employing whitening process shows several advantages on diversity gain and the required number of receive antennas compare to the convolutional coded space-time processing. The proposed receivers use different decoding order scheme according to each interference suwression. The (4, 3) LSTT-Whitening receiver still achieves 1㏈ gain over the (4, 4) LSTT-MMSE and the (4, 4) coded layered space-time processing.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.1
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• pp.189-195
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• 2008

The OFDMA Physical layer in the WiMAX standard of IEEE 802.16e adopts 114 LDPC codes with various code rates and block sizes as a channel coding scheme to meet varying channel environments and different requirements for transmission performance. In this paper, the performances of the LDPC codes are evaluated according to various code rates and block-lengths throueh simulation studies using min-sum decoding algorithm in AWGN chamois. As the block-length increases and the code rate decreases, the BER performance improves. In the cases with code rates of 2/3 and 3/4, where two different codes ate specified for each code rate, the codes with code rates of 2/3A and 3/4B outperform those of 2/3B and 3/4A, respectively. Through the statistical analyses of the number of decoding iterations the decoding complexity and the word error rates of LDPC codes are estimated. The results can be used to trade-off between the performance and the complexity in designs of LDPC decoders.

• Journal of Communications and Networks
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• v.10 no.4
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• pp.371-383
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• 2008

Application of network coding in wireless two-way relay channels (TWRC) has received much attention recently because its ability to improve throughput significantly. In traditional designs, network coding operates at upper layers above (including) the link layer and it requires the input packets to be correctly decoded. However, this requirement may limit the performance and application of network coding due to the unavoidable fading and noise in wireless networks. In this paper, we propose a new wireless network coding scheme for TWRC, which is referred to as soft network coding (SoftNC), where the relay nodes applies symbol-by-symbol soft decisions on the received signals from the two end nodes to come up with the network coded information to be forwarded. We do not assume further channel coding on top of SoftNC at the relay node (channel coding is assumed at the end nodes). According to measures of the soft information adopted, two kinds of SoftNC are proposed: amplify-and-forward SoftNC (AF-SoftNC) and soft-bit-forward SoftNC (SBF-SoftNC). We analyze the both the ergodic capacity and the outage capacity of the two SoftNC schemes. Specifically, analytical form approximations of the ergodic capacity and the outage capacity of the two schemes are given and validated. Numerical simulation shows that our SoftNC schemes can outperform the traditional network coding based two-way relay protocol, where channel decoding and re-encoding are used at the relay node. Notable is the fact that performance improvement is achieved using only simple symbol-level operations at the relay node.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.8 no.6
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• pp.63-69
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• 2008

In this paper, our goal is to find a power-effective protocol that improves the accuracy of transmission in sensor networks. Therefore we propose a cooperative communication protocol based on MRC(Maximal Ratio Combining) and cyclic code. In our proposal, one sensor node assists two others to communicate with a clusterhead that can get diversity effect and MRC can improve diversity effect also. The proposed protocol with cyclic code can correct error up to 3-bit and reduce decoding complexity compared with convolutional code. Simulation results reveal proposed protocol can save the network energy up to 6dB over single-hop protocol at BER(Bit Error Rate) of $10^{-2}$.

• Journal of IKEEE
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• v.2 no.2 s.3
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• pp.285-293
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• 1998

This paper presents unequal error protection and error concealment techniques far robust H.263 video transmission over mobile channels. The proposed error protection scheme has three major features. First, it has the capability of preventing the loss of synchronization information in H.263 video stream as much as possible that the H.263 decoder can resynchronize at the next decoding point, if errors are occurred. Secondly, it employs an unequal error protection scheme to support variable coding rates using rate compatible punctured convolutional (RCPC) codes, dividing the encoded stream into two classes. Finally, a macroblock-interleaving scheme is employed in order to minimize the corruption of consecutive macroblocks due to burst errors, which can make a proper condition for error concealment. In addition, to minimize the spatial error propagations due to the variable length codes, a fast resynchronization scheme at the group of block layer is developed for recovering subsequent error-free macroblocks following the damaged macroblock. futhermore, error concealment techniques based on both side match criterion and overlapped block motion compensation (OBMC) are employed at the source decoder so that it can not only recover the lost macroblock more accurately, but also reduce blocking artifacts. Experimental results show that the proposed scheme can be an effective error protection scheme since proper video quality can be maintained under various channel bit error rates.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.11B
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• pp.1151-1168
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• 2009

Improving packet loss does not necessarily coincide with the improvement in user perceivable QoS because each frame carries different degree of importance. We propose Significance-aware packet scheduling (SAPS) to maximize user perceivable QoS. SAPS carries out two fundamental issues of packet scheduling: "What to transmit" and "When to transmit?" To adapt to the available bandwidth, it is necessarily to transmit the subset of the data packets if the entire set of packets can not be transmitted. "Packet Significance" quantifies the importance of the frame by elaborately incorporating frames' dependency. Greedy approach is used in selecting packets and transmission schedule is determined based on the Packet Significance. The proposed scheme is tested using publicly available MPEG-4 video clips. Decoding engine is embedded in the simulation software and user perceivable QoS is exposeed in termstermiSNR. Throughout the simulation based experiment, the performance of the proposed scheme is compared two other schemes: Size-based packet scheduling and Bit-rate based best effort packet scheduling. SAPS successfully incorporates the semantics of a packet and improves user perceivable QoS significantly. It successfully provides unequal protection to more important packets.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2E
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• pp.91-97
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• 1995

This paper describes the implementation of a multi-channel audio codec for HETV. This codec has the features of the 3/2-stereo plus low frequency enhancement, downward compatibility with the smaller number of channels, backward compatibility with the existing 2/0-stereo system(MPEG-1 audio), and multilingual capability. The encoder of this codec consists of 6-channel analog audio input part with the sampling rate of 48 kHz, 4-channel digital audio input part and three TMS320C40 /DSPs. The encoder implements multi-channel audio compression using a human perceptual psychoacoustic model, and has the bit rate reduction to 384 kbit/s without impairment of subjective quality. The decoder consists of 6-channel analog audio output part, 4-channel digital audio output part, and two TMS320C40 DSPs for a decoding procedure. The decoder analyzes the bit stream received with bit rate of 384 kbit/s from the encoder and reproduces the multi-channel audio signals for analog and digital outputs. The multi-processing of this audio codec using multiple DSPs is ensured by high speed transfer of date between DSPs through coordinating communication port activities with DMA coprocessors. Finally, some technical considerations are suggested to realize the problem of real-time operation, which are found out through the implementation of this codec using the MPEG-2 layer II sudio coding algorithm and the use of the hardware architecture with commercial multiple DSPs.