• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.1915-1919
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• 2003

A practical method of adaptive rate allocation to source and channel codings for an independent loss channel is proposed for Internet video. It is based on the observations that the values of residual loss probabilities at the optimal code rates for different packet loss probabilities are closely clustered to the average residual loss probability for a transmission frame size n in RS(n,k) code and for a total bit rate R. These observations aye then exploited to find the code rate for maximum PSNR. Simulation results demonstrate that the proposed method achieves a near-optimal bit-rate allocation in the joint source-channel coding of H.263 and RS(n,k) codings.

• Proceedings of the KOSOMBE Conference
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• v.1991 no.11
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• pp.44-47
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• 1991

There exists a transform trellis code that is optimal for stationary Gaussian sources and the squared-error distortion measure at all rates. In this paper, we train an asymptotically optimal version of such a code to obtain one which is matched better to the statistics of real world data. The training algorithm uses the M-algorithm to search the trellis codebook and the LBG-algorithm to update the trellis codebook. To adapt the codebook for the varying input data. we use two gain-adaptive methods. The gain-adaptive scheme 1, which normalizes input block data by its gain factor, is applied to images at rate 0.5 bits/pixel. When each block is encoded at the same rate, the nonstationarity among the block variances leads to a variation in the resulting distortion from one block to another. To alleviate the non-uniformity among the encoded image, we design four clusters from the block power, in which each cluster has its own trellis codebook and different rates. The rate of each cluster is assigned through requiring a constant distortion per-letter. This gain-adaptive scheme 2 produces good visual and measurable quality at low rates.

• Journal of Communications and Networks
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• v.7 no.4
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• pp.489-498
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• 2005

For better performance over a noisy channel, mobile wireless networks transmit packets with forward error correction (FEC) code to recover corrupt bits without retransmission. The static determination of the FEC code size, however, degrades their performance since the evaluation of the underlying channel state is hardly accurate and even widely varied. Our measurements over a wireless sensor network, for example, show that the average bit error rate (BER) per second or per minute continuously changes from 0 up to $10^{-3}$. Under this environment, wireless networks waste their bandwidth since they can't deterministically select the appropriate size of FEC code matching to the fluctuating channel BER. This paper proposes an adaptive FEC technique called adaptive FEC code control (AFECCC), which dynamically tunes the amount of FEC code per packet based on the arrival of acknowl­edgement packets without any specific information such as signal to noise ratio (SNR) or BER from receivers. Our simulation experiments indicate that AFECCC performs better than any static FEC algorithm and some conventional dynamic hybrid FEC/ARQ algorithms when wireless channels are modeled with two-state Markov chain, chaotic map, and traces collected from real sensor networks. Finally, AFECCC implemented in sensor motes achieves better performance than any static FEC algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.3A
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• pp.406-411
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• 2000

In wireless ATM system, powerful FEC code is required for highly reliable data transmission. In this paper, we propose an adaptive hybrid ARQ scheme using RCPSCCC for WATM system. The code rate of RCPSCC is adjusted to match channel conditions and data types. By using only the effective free distances of outer and inner encoders, we derive upper bounds of the bit and word error probabilities over Rayleigh and Rician fading channels. By applying RCPSCC to the adaptive hybrid ARQ protocol, highly reliable data transmission can be achieved.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12A
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• pp.1862-1867
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• 1999

In wireless ATM system, powerful FEC code is required for highly reliable data transmission. In this paper, we propose an adaptive hybrid ARQ scheme using RCPSCCC for WATM system. The code rate of RCPSCCC is adjusted to match channel conditions and data types. By using only the effective free distances of outer and inner encoders, we derive upper bounds of the bit and word error probabilities over Rayleigh and Rician fading channels. By applying RCPSCCC to the adaptive hybrid ARQ protocol, highly reliable data transmission can be achieved.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1132-1136
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• 2005

In this paper, adaptive equalizer receiver for a turbo code direct sequence code division multiple access (DSCDMA) by using least mean square (LMS) adaptive algorithm is presented. The proposed adaptive equalizer is using soft output of decision feedback adaptive equalizer (DFE) to examines the output of the equalizer and the Log- maximum a posteriori (Log-MAP) algorithm for the turbo decoding process of the system. The objective of the proposed equalizer is to minimize the bit error rate (BER) of the data due to the disturbances of noise and intersymbol interference (ISI)phenomenon on the channel of the DS-CDMA digital communication system. The computer program simulation results shown that the proposed soft output decision feedback adaptive equalizer provides a good BER than the others one such as conventional adaptive equalizer, infinite impulse response adaptive equalizer.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.140-143
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• 2000

In this paper, forward error control coding in multicarrier direct sequence code division multiple access (DS/CDMA) systems is considered. In order to accommodate a number of coding rates easily and make the encoder and do-coder structure simple, we use the rate compatible punctured convolutional (RCPC) code. We obtain data throughputs at several coding rates and choose the coding rate which has the highest data throughput in the SINR sense. To achieve maximum data throughput, a rate adaptive system using channel state information (the SINR estimate) is proposed. The SINR estimate is obtain by the soft decision Viterbi decoding metric. We show that the proposed rate adaptive convolutionally coded multicarrier DS/CDMA system can enhance spectral efficiency and provide frequency diversity.

• Journal of Satellite, Information and Communications
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• v.10 no.2
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• pp.19-23
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• 2015

In this paper, we present the return link ACM method to improve the link availability and system throughput for satellite communication service. Also, we describe the optimization of an algorithm for channel prediction using the LMS (Least Mean Square) adaptive filter and the MODCOD (Modulation & Code rate) decision. The simulation results show that the optimized filter taps and step-size of adaptive filter are 2 and 0.00026, respectively. And also confirms the required SNR margin for minimization of MODCOD decision error is 0.3dB.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.252-255
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• 2000

In this paper, a fast pseudo-noise (PN) code acquisition with novel adaptive architecture is presented in direct-sequence spread- spectrum (DS-SS) systems. Since an existing acquisition system has a fixed correlation tap size and threshold value, this system cannot adapt to various mobile communication environments and results in a low detection probability or a high false alarm rate and long acquisition time. Therefore, if a correlation tap size and a threshold value can be controlled adaptively according to received signals, problems of ail existing system will be solved. The system parameter varies adaptively by using constant false alarm rate (CFAR) algorithm well known in a field of detection and proposed signal-to-noise ratio (SNR) measurement system. By deriving formulas of the proposed system, the performance is analyzed.