• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.548-555
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• 2013

The localization of sources has a numerous number of applications. To estimate the position of sources, the relative delay between two or more received signals for the direct signal must be determined. Although the generalized cross-correlation method is the most popular technique, an approach based on eigenvalue decomposition (EVD) is also popular one, which utilizes an eigenvector of the minimum eigenvalue. The performance of the eigenvalue decomposition (EVD) based method degrades in the low SNR and the correlated environments, because it is difficult to select a single eigenvector for the minimum eigenvalue. In this paper, we propose a new adaptive algorithm based on Canonical Correlation Analysis (CCA) in order to extend the operation range to the lower SNR and the correlation environments. The proposed algorithm uses the eigenvector corresponding to the maximum eigenvalue in the generalized eigenvalue decomposition (GEVD). The estimated eigenvector contains all the information that we need for time delay estimation. We have performed simulations with uncorrelated and correlated noise for several SNRs, showing that the CCA based algorithm can estimate the time delays more accurately than the adaptive EVD algorithm.

• Proceedings of the KIPE Conference
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• 1998.07a
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• pp.85-89
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• 1998

A switching power stage is a very nonlinear system because it has two or more operation modes in one switching cycle. To model a switching power stage, the state space averaging method has been developed. Though it allows a unified treatment of a large variety of switching power stages, the model it yields is always very nonlinear. So, it is required to linearize the averaged model. But it is well known that a controller for a nonlinear plant designed by the linearization frequently fails in showing satisfactory control performance. Hence it is very natural to try to design a nonlinear controller for a switching power stage. In design of a switching power system, nonlinear control approaches such as adaptive control and fuzzy control have been widely studied so far. In this research, a recently developed control method, time delay control is briefly studied and a design example for a ZVS PWM half bridge converter is given. The performance of the time delay controller is compared to its conventional counterpart, PI controller by computer simulations.

• Journal of Communications and Networks
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• v.5 no.1
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• pp.65-72
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• 2003

Internet roundtrip delay/time (RTT) prediction plays an important role in detecting packet losses in reliable transport protocols for traditional web applications and determining proper transmission rates in many rate-based TCP-friendly protocols for Internet-based real-time applications. The widely adopted autoregressive and moving average (ARMA) model with fixed-parameters is shown to be insufficient for all scenarios due to its intrinsic limitation that it filters out all high-frequency components of RTT dynamics. In this paper, we introduce a novel parameter-varying RTT model for Internet roundtrip time prediction based on the information theory and the maximum entropy principle (MEP). Since the coefficients of the proposed RTT model are updated dynamically, the model is adaptive and it tracks RTT dynamics rapidly. The results of our experiments show that the MEP algorithm works better than the ARMA method in both RTT prediction and RTO estimation.

• Journal of Information Processing Systems
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• v.19 no.6
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• pp.817-829
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• 2023

Variable wireless channel is a big challenge for real-time video applications, and the rate adaptation of realtime video streaming becomes a hot topic. Intra-video coding is important for high-quality video communication and industrial video applications. In this paper, we proposed a novel adaptive scheme for real-time video transmission with intra-only coding over a wireless network. The key idea of this scheme is to estimate the instantaneous remaining capacity of the network to adjust the quality of the next several video frames, which not only can keep low queuing delay and ensure video quality, but also can respond to bandwidth changes quickly. We compare our scheme with three different schemes in the video transmission system. The experimental results show that our scheme has higher bandwidth utilization and faster bandwidth change response, while maintaining low queuing delay.

• Journal of Communications and Networks
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• v.15 no.5
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• pp.496-503
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• 2013

In an orthogonal frequency division multiplexing system, conventional interpolation techniques cannot correctly balance performance and overhead when estimating dynamic long-delay channels in single frequency networks (SFNs). In this study, classical filter analysis and design methods are employed to derive a complex interpolator for maximizing the resistible echo delay in a channel estimator on the basis of the correlation between frequency domain interpolating and time domain windowing. The coefficient computation of the complex interpolator requires a key parameter, i.e., channel length, which is obtained in the frequency domain with a tentative estimation scheme having low implementation complexity. The proposed complex adaptive interpolator is verified in a simulated digital video broadcasting for terrestrial/handheld receiver. The simulation results indicate that the designed channel estimator can not only handle SFN echoes with more than $200{\mu}s$ delay but also achieve a bit-error rate performance close to the optimum minimum mean square error method, which significantly outperforms conventional channel estimation methods, while preserving a low implementation cost in a short-delay channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.4B
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• pp.675-682
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• 2000

In this paper, we propose a new scheduling algorithm called the Adaptive Weighted Round Robin with Delay Tolerance (AWRR/DT). The proposed scheme is based on the per-class queueing mechanism in which a number of connections of similar characteristics are multiplexed into one class-queue. Traffic classes of the proposed method are classified into a single non-real-time traffic class and other real-time traffic classes. The proposed scheme determines the weights of classes according to the input traffic and delay characteristics of each class at the beginning of every cycle. Furthermore, this scheme incorporates a cell discarding method to reduce the QoS degradation that may be incurred by congestion of networks. We have evaluated the proposed scheme through discrete-event simulation. Simulation results indicate that the proposed scheme can reduce the average delay of non-real-time class while maintaining the QoS of real-timeclasses. The proposed algorithm can be effectively applied to high-speed networks such as ATM networks.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.6 s.348
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• pp.30-38
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• 2006

In this paper, an algerian that provisions absolute and proportional differentiation of packet delays is proposed with an objective for enhancing quality of service (QoS) in future packet networks. It features a scheme that compensates the deviation for prediction on the traffic to be arrived continuously It predicts the traffic to be arrived at the beginning of a time slot and measures the actual arrived traffic at the end of the time slot and derives the difference between them. The deviation is utilized to the delay control operation for the next time slot to offset it. As it compensates the prediction error continuously, it shows superior adaptability to the bursty traffic as well as the exponential traffic. It is demonstrated through simulation that the algorithm meets the quantitative delay bounds and shows superiority to the traffic fluctuation in comparison with the conventional non-adaptive mechanism.

• Journal of KIISE
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• v.42 no.12
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• pp.1495-1502
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• 2015

Data compression involves a trade-off between delay time and data size. Greater delay times require smaller data sizes and vice versa. There have been many studies performed in the field of wireless sensor networks on increasing network life cycle durations by reducing data size to minimize energy consumption; however, reductions in data size result in increases of delay time due to the added processing time required for data compression. Meanwhile, as energy generation occurs periodically in solar energy-based wireless sensor networks, redundant energy is often generated in amounts sufficient to run a node. In this study, this excess energy is used to reduce the delay time between nodes in a sensor network consisting of solar energy-based nodes. The energy threshold value is determined by a formula based on the residual energy and charging speed. Nodes with residual energy below the threshold transfer data compressed to reduce energy consumption, and nodes with residual energy above the threshold transfer data without compression to reduce the delay time between nodes. Simulation based performance verifications show that the technique proposed in this study exhibits optimal performance in terms of both energy and delay time compared with traditional methods.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.8
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• pp.3542-3566
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• 2018

In this paper, we investigate a buffer-aided wireless powered cooperative communication network (WPCCN), in which the source and relay harvest the energy from a dedicated power beacon via wireless energy transfer, then the source transmits the data to the destination through the relay. Both the source and relay are equipped with an energy buffer to store the harvested energy in the energy transfer stage. In addition, the relay is equipped with a data buffer and can temporarily store the received information. Considering the buffer-aided WPCCN, we propose two buffer-aided relaying protocols, which named as the buffer-aided harvest-then-transmit (HtT) protocol and the buffer-aided joint mode selection and power allocation (JMSPA) protocol, respectively. For the buffer-aided HtT protocol, the time-averaged achievable rate is obtained in closed form. For the buffer-aided JMSPA protocol, the optimal adaptive mode selection scheme and power allocation scheme, which jointly maximize the time-averaged throughput of system, are obtained by employing the Lyapunov optimization theory. Furthermore, we drive the theoretical bounds on the time-averaged achievable rate and time-averaged delay, then present the throughput-delay tradeoff achieved by the joint JMSPA protocol. Simulation results validate the throughput performance gain of the proposed buffer-aided relaying protocols and verify the theoretical analysis.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.28-36
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• 1995

In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.