• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.11
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• pp.2264-2275
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• 1994

In this paper, image compression based on neural networks is presented with block classification and coding. Multilayer neural networks with error back-propagation learning algorithm are used to transform the normalized image date into the compressed hidden values by reducing spatial redundancies. Image compression can basically be achieved with smaller number of hidden neurons than the numbers of input and output neurons. Additionally, the image blocks can be grouped for adaptive compression rates depending on the characteristics of the complexity of the blocks in accordance with the sensitivity of the human visual system(HVS). The quantized output of the hidden neuron can also be entropy coded for an efficient transmission. In computer simulation, this approach lie in the good performances even with images outside the training set and about 25:1 compression rate was achieved using the entropy coding without much degradation of the reconstructed images.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.10
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• pp.2447-2453
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• 2013

The digital images are being degraded by noise in the process of acquisition, storage and transmission, Gaussian or impulse noise is the representative noise. Meanwhile, the image has lots of tendency to be degraded by complex noise, so various researches are being conducted for reducing these complex noise. In this paper, to remove complex noise, the algorithm processed by modified switching median filter and modified adaptive weighted filter according to the result after judging the kinds of noise is proposed. In the simulation result, excellent denoising capabilities. Furthermore, we compared proposed algorithm with existing methods for objective judgement, and PSNR(peak signal to noise ratio) is used by the criterion of judgement.

• ETRI Journal
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• v.22 no.2
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• pp.11-19
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• 2000

If the right and left signals of a binaural sound recording are reproduced through loudspeakers instead of a headphone, they are inevitably mixed during their transmission to the ears of the listener. This degrades the desired realism in the sound reproduction system, which is commonly called 'cross-talk.' A 'cross-talk canceler' that filters binaural signals before they are sent to the sound sources is needed to prevent cross-talk. A cross-talk canceler equalizes the resulting sound around the listener's ears as if the original binaural signal sound is reproduced next to the ears of listener. A cross-talk canceler is also a solution to the problem-how binaural sound is distributed to more than 2 channels that drive sound sources. This paper presents an effective way of building a cross-talk canceler in which geometric information, including locations of the listener and multiple loudspeakers, is divided into angular information and distance information. The presented method makes a database in an off-line way using an adaptive filtering technique and Head Related Transfer Functions. Though the database is mainly concerned about the situation where loudspeakers are located on a standard radius from the listener, it can be used for general radius cases after a distance compensation process, which requires a small amount of computation. Issues related to inverting a system to build a cross-talk canceler are discussed and numerical results explaining the preferred configuration of a sound reproduction system for stereo loudspeakers are presented.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.2
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• pp.197-203
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• 2011

Ad-hoc networks do not rely on a preexisting infrastructure such as Access Points(AP) in wireless network infrastructure. Instead each node participates in routing by forwarding data for other nodes. It makes required broadcasting to transmit packets to the whole network. In that part, each node tries to transmit data without any information about the other nodes. Therefore it causes duplication of transmission and waste of power. This paper presents adaptive probabilistic broadcasting schemes based on packet reception counts to alleviate the broadcast storm problem for wireless ad hoc networks. In this algorithm, each node calculates efficiency broadcast probability. Simulation results for the proposed flood algorithm are also presented.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.2
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• pp.637-658
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• 2015

The development of multimedia applications has followed the development of high-speed networks. By improving the performance of mobile devices, it is possible to provide high-transfer-speed broadband and seamless mobile multicasting services between indoor and outdoor environments. Multicasting services support efficient group communications. However, mobile multicasting services have two constraints: tunnel convergence and handoff latency. In order to solve these problems, many protocols and handoff methods have been studied. In this paper, we propose inter local mobility anchor (inter-LMA) optimized handoff model for mobile multicasting services in proxy mobility IPv6 based (PMIPv6-based) networks. The proposed model removes the tunnel convergence issue and reduces the router processing costs. Further, it the proposed model allows for the execution of fast handoff operations with adaptive transmission mechanisms. In addition, the proposed scheme exhibits low packet delivery costs and handoff latency in comparison with existing schemes and ensures fast handoff when moving the inter-LMA domain.

• Proceedings of the KSME Conference
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• 2004.11a
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• pp.585-590
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• 2004

Many machine failures are not detected well in advance due to the masking of background noise and attenuation of the source signal through the transmission mediums. Advanced signal processing techniques using adaptive filters and higher order statistics have been attempted to extract the source signal from the measured data at the machine surface. In this paper, blind deconvolution using the eigenvector algorithm (EVA) technique is used to recover a damaged bearing signal using only the measured signal at the machine surface. A damaged bearing signal corrupted by noise with varying signal-to-noise (s/n) was used to determine the effectiveness of the technique in detecting an incipient signal and the optimum choice of filter length. The results show that the technique is effective in detecting the source signal with an s/n ratio as low as 0.21, but requires a relatively large filter length.

• Journal of Satellite, Information and Communications
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• v.10 no.2
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• pp.10-13
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• 2015

This paper proposed a novel user's information feedback algorithm for improving network energy harvesting rate. The proposed algorithm is focused on determining energy harvesting users comparing increasing ratio of the amount of harvesting energy versus emitted energy and network threshold ${\alpha}$, which is critical harvesting parameter. Using this method, we can increase the rate of network energy harvesting preventing emitted energy from wasting inefficiently. The result of experiment in this paper shows that user's information feedback algorithm makes network energy harvesting rate more efficiently when it uses threshold ${\alpha}=15%$ to determine energy harvesting users.

• Journal of Advanced Navigation Technology
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• v.5 no.2
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• pp.165-174
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• 2001

In this paper, we investigated performance for 5.8 GHz dedicated short range communication system using OFDM which will be applied to Intelligent transportation system services. We analyzed modulation technique and interference cancellation method to improve performance in physical layer. We presented channel model to estimate performance between Roadside Unit and Onboard Unit. finally, We calculated the link budget of entire system which analyzed performance of channel according to transmission power, presented the adaptive modulation using pilot signal and RSSI algorithm is proposed.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.10a
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• pp.468-472
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• 2000

This paper discusses the development of a acoustic vector channel simulator for the performance analysis of a acoustic digital communication system. The channel simulator consists of transmission module, acoustic channel model, receiver, beamformer, and adaptive equalizer. QPSK source signal is generated by the parameters specified by a user and transmitted. The transmitted signal generate multipath signals which have a different delay, amplitude, and dopper Sequency. The multipath signals with the acoustic noises is the received signal. We can analysis the communication system performance according to the antenna structure, beamforming algorithm, and equalization algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4B
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• pp.255-269
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• 2006

The differentiated services(DiffServ) architecture provides packet level service differentiation through the simple and predefined Per-Hop Behaviors(PHBs). The Assured Forwarding(AF) PHB proposed as the assured services uses the RED-in/out(RIO) approach to ensusre the expected capacity specified by the service profile. However, the AF PHB fails to give good QoS and fairness to the TCP flows. This is because OUT(out- of-profile) packet droppings at the RIO buffer are unfair and sporadic during only network congestion while the TCP's congestion control algorithm works with a different round trip time(RTT). In this paper, we propose an Adaptive Regulating Drop(ARD) marker, as a novel dropping strategy at the ingressive edge router, to improve TCP fairness in assured services without a decrease in the link utilization. To drop packets pertinently, the ARD marker adaptively changes a Temporary Permitted Rate(TPR) for aggregate TCP flows. To reduce the excessive use of greedy TCP flows by notifying droppings of their IN packets constantly to them without a decrease in the link utilization, according to the TPR, the ARD marker performs random early fair remarking and dropping of their excessive IN packets at the aggregate flow level. Thus, the throughput of a TCP flow no more depends on only the sporadic and unfair OUT packet droppings at the RIO buffer in the core router. Then, the ARD marker regulates the packet transmission rate of each TCP flow to the contract rate by increasing TCP fairness, without a decrease in the link utilization.