• 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.

• Transactions of the Society of Information Storage Systems
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• v.1 no.2
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• pp.155-160
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• 2005

In this paper, the channel decoder promising reliable data retrieving in noisy holographic channel has been developed for holographic WORM(write once read many) system. It covers various DSP(digital signal processing) blocks, such as align mark detector, adaptive channel equalizer, modulation decoder and ECC(error correction code) decoder. The specific schemes of DSP are designed to reduce the effect of noises in holographic WORM(H-WORM) system, particularly in prototype of DAEWOO electronics(DEPROTO). For real time data retrieving, the channel decoder is redesigned for FPGA(field programmable gate array) based hardware, where DSP blocks calculate in parallel sense with memory buffers between blocks and controllers for driving peripherals of FPGA. As an input source of the experiments, MPEG2 TS(transport stream) data was used and recorded to DEPROTO system. During retrieving, the CCD(charge coupled device), capturing device of DEPROTO, detects retrieved images and transmits signals of them to the FPGA of hardware channel decoder. Finally, the output data stream of the channel decoder was transferred to the MPEG decoding board for monitoring video signals. The experimental results showed the error corrected BER(bit error rate) of less than $10^{-9}$, from the raw BER of DEPROTO, about $10^{-3}$. With the developed hardware channel decoder, the real-time video demonstration was possible during the experiments. The operating clock of the FPGA was 60 MHz, of which speed was capable of decoding up to 120 mega channel bits per sec.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.1_2
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• pp.1-14
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• 2006

Sort-last parallel rendering using a cluster of GPUs has been widely used as an efficient method for visualizing large- scale volume datasets. The performance of this method is constrained by load balancing when data parallelism is included. In previous works static partitioning could lead to self-balance when only task level parallelism is included. In this paper, we present a load balancing scheme that adapts to the characteristic of volume dataset when data parallelism is also employed. We effectively combine the hierarchical data structures (octree and BSP tree) in order to skip empty regions and distribute workload to corresponding rendering nodes. Moreover, we also exploit a 3D clustering method to determine visibility order and save the AGP bandwidths on each rendering node. Experimental results show that our scheme can achieve significant performance gains compared with traditional static load distribution schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.5C
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• pp.398-405
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• 2005

The multiuser detection (MUD) based wireless receiver requires more accurate channel estimation than the single user detection (SUD) schemes such as Rake receiver, and hence the post processing is required for MUD to clean up the estimated channel coefficients by eliminating the noise only coefficients. The adaptive post processing method is proposed in order to provide more accurate channel responses and the power level of the background noise and interferences at the cost of the negligible processing delay compared to the conventional method based on the threshold test with the threshold value relative to the noise variance. The simulations are performed in 3GPP-TDD mode environment. The results show that the noise estimation error of the proposed method is maximum $10\%$, which is much smaller than $50\%$ maximum error of the conventional method.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.6
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• pp.205-212
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• 2003

Because a wireless ad hoc network does not have fixed infrastructure, a call admission control approach researched in a wireless network is not feasible to this network. In this paper, we propose call admission control scheme to support this problem and the burst handoff traffic due to group mobility in a wireless ad hoc network. This scheme is an adaptive guard channel scheme which adapt the number of guard channels in each MBS(mobile base station) according to the current estimate of the potential handoff call rate derived from the number of ongoing calls within the coverage area of an MBS that initiate group handoff in a wireless ad hoc network. Our simulation studies are performed for comparisons of the proposed scheme with the other channel allocation schemes. Simulation results show that the proposed scheme efficiently reduces handoff call blocking probability in wireless ad hoc networks.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.5 s.305
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• pp.115-122
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• 2005

There are some problems in directly applying the conventional MPEG bit-stream based watermarking schemes to the bit-stream of a new compression standard, H.264/AVC. In this paper we analyze the effects of the conventional DCT-based watermarking scheme to H.264/AVC, especially in terms of image distortions and bit-rate changes. It turns out that the intra-frame prediction md CAVLC of H.264/AVC with the watermarking worsen the image distortions and bit-rate changes. The experiment results show on average 28.17dB decrease in PSNR and 56.71% increase in bit-rate over all QPs.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.1
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• pp.1-13
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• 2001

In this paper, we propose a new dynamic bandwidth allocation and call admission control for VBR video sources with QoS constraints to support an efficient resource management and at the same Lime to satisfy the user's quality or service requirements. For the dynamic bandwidth allocation, first the next amount of traffic is predicted using a modified adaptive linear prediction method that considers abrupt scene change effects. And then, we dynamically allocate the necessary bandwidth to each connection based on the currently provided quality degree by the network with respect to the user's QoS requirements in terms of average delay and loss ratio. For the admission control, we determine the acceptance or rejection or a new connection based on the quality satisfaction degrees of the existing connections. Simulation results show that our proposed dynamic schemes are able to provide a stable service, which well meets the user's quality requirements.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.60 no.1
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• pp.205-213
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• 2011

The Long Term Evolution (LTE) system is designed to provide a high quality data service for fast moving mobile users. It is based on the Orthogonal Frequency Division Multiplexing (OFDM) and relies its channel estimation on the training samples which are systematically built within the transmitting data. Either a preamble or a lattice type is used for the distribution of training samples and the latter suits better for the multipath fading channel environment whose channel frequency response (CFR) fluctuates rapidly with time. In the lattice-type structure, the estimation of the CFR makes use of the least squares estimate (LSE) for each pilot samples, followed by an interpolation both in time-and in frequency-domain to fill up the channel estimates for subcarriers corresponding to data samples. All interpolation schemes should rely on the pilot estimates only, and thus, their performances are bounded by the quality of pilot estimates. However, the additive noise give rise to high fluctuation on the pilot estimates, especially in a communication environment with low signal-to-noise ratio. These high fluctuations could be monitored in the alternating high values of the first forward differences (FFD) between pilot estimates. In this paper, we analyzed statistically those FFD values and propose a postprocessing algorithm to suppress high fluctuations in the noisy pilot estimates. The proposed method is based on a localized adaptive moving-average filtering. The performance of the proposed technique is verified on a multipath environment suggested on a 3GPP LTE specification. It is shown that the mean-squared error (MSE) between the actual CFR and pilot estimates could be reduced up to 68% from the noisy pilot estimates.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2042-2051
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• 1998

VBR compressed, pre-recorded video is known to exhibit significant, multiple time-scale bit rate variability. To reduce the variability and burstiness of the aggregated transmission rate as low as possible, in this paper, we present an adaptive bandwidth smoothing algorithm, that can be effectively applicalble for VoD services over a shared-medium channel. For these environments where many clients are connected to a single server, by introducing the conventional MVBA(minimum variability bandwidth allocation) algorithm and controlling adaptively the conventional MVBA(minimum variability bandwidth asllocation) algorithm and controlling adaptively the aggregated transmission rate whenever a new clients request is arrived at the server side, the proposed algorithm effectively reduces the burstiness and variability of the aggregated transmission rate. Through computer experiments, it is shown that the proposed method perporms better than the convertional non-aggregated bandwidth smoothing schemes in terms of the peak rate, standard deviation, number of rate changes for the aggregated traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.10A
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• pp.1139-1146
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• 2004

An abrupt increase of traffic-load in cellular CDMA systems can cause overload, as a result, degrade the quality of service (QoS) and the spectrum utilization due to lacking radio resources in base stations. In this paper, we propose a joint coverage and resource management (JCRM) scheme which can improve the QoS degradation and the spectrum utilization. The JCRM scheme hands over the overloaded traffic to neighboring cells by virtually reducing overloaded cell coverage and extending neighboring cell coverage, as well the scheme allocates radio resources based on the necessary handover probability. The proposed scheme can be applied to the existing cellular CDMA systems as well as adaptive coverage management schemes for next generation mobile communication systems.