• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.12C
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• pp.1273-1279
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• 2006

In this paper, we address the design of adaptive handover schemes based on the signal strength measurement for cellular communications systems. Conventional handover algorithms, which are based only on the downlink measurement, cannot guarantee the required uplink quality because uplink channel can differ greatly from downlink channel quality. Therefore, we proposes a new suboptimal adaptive handover algorithm that considers both the uplink and downlink channel quality in order to achieve the best cell selection gains when there is a wide difference between the uplink and downlink signal quality. Simulation results show that the proposed scheme achieves better performance compared to conventional handover schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.8
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• pp.1548-1557
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• 1994

In this paper, the dual mode error correcting adaptive decoding algorithm which is adapted to the continuous phase frequency shift keying(CPFSK) modulation is presented as a technique for overcoming the distortion that reveals from the Rayleigh fading channel. The dual mode adaptive decoder nominally operates as a Viterbi decoder and switches to the burst error correcting mode, whenever the decoder detects an uncorrectable burst error pattern. Under the fading channel environment and when the usable memory quantity is restricted, the dual mode adaptive decoding algorithm shows an advantage in the BER performance over the interleaving technique, and also obtains the merit of not needing the large time delay that the interleaving technique requires. The experimental results from the computer simulation demonstrate the performance of the algorithm and verify the theoretical results.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.6
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• pp.1152-1168
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• 2010

Recently, due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services such as the mobile IPTV. In such a wireless broadband network, transmitting a continuous stream of multimedia data is difficult to achieve due to mobile stations (MSs) movement. Providing Quality of Service (QoS) for multimedia video streaming applications requires the server and/or client to be network-aware and adaptive. Therefore, in order to deploy a mobile IPTV service in wireless broadband networks, offering users efficient wireless resource utilization and seamlessly offering user perceived QoS are important issues. In this paper, we propose a new adaptive streaming scheme, called MARC (Mobile-aware Adaptive Rate Control), which adjusts the quality of bit-stream and transmission rate of video streaming based on the wireless channel status and network status. The proposed scheme can control the rate of multimedia streaming to be suitable for the wireless channel status by using awareness information of the wireless channel quality and the mobile station location. The proposed scheme can provide a seamless multimedia playback service in wireless broadband networks in addition to improving the QoS of multimedia streaming services. The proposed MARC scheme alleviates the discontinuity of multimedia playback and allocates a suitable client buffer to the wireless broadband network. The simulation results demonstrate the effectiveness of our proposed scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.12
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• pp.1219-1228
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• 1991

In the actual case, a model of noise canceller using adaptive filter has both a channel transfer function from noise source to main signal input and to noise canceller input. The previous models of noise canceller have been considered to be only one side channel transfer function. Therefore, it is proposed that a new model has two channel transfer functions and derives an optimal tranfer function of adaptive noise canceller. The adaptive filter is using the joint process lattice filter that has fast adaptive speed. The signal noise radio has been improved by a model of ANC-ALE and it is confirmed with computer simulation. Beside, a dc bias is very effective for noise cancelling, especially to the particular signal.

• Journal of electromagnetic engineering and science
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• v.16 no.2
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• pp.80-86
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• 2016

This paper proposes the algorithm for forward link adaptive coding and modulation (ACM) and the detailed design for a satellite communication system to improve network reliability and system throughput. In the ACM scheme, the coding and modulation schemes are changed by as much as the channel can provide depending on the quality of the communication link. To implement the forward link ACM system in the Ka-band, channel prediction and modulation/coding decision methods are proposed and simulated. The parameters of the adaptive filter predictor based on the least mean square are optimized, the minimum mean square error of the channel predictor is 0.0608 when step size and the number of filter tap are 0.0001 and 4, respectively. A test-bed is set up to verify the forward link ACM system, and a test is performed using a Ka-band satellite (i.e., Communication, Ocean, and Meteorological Satellite [COMS]). This test verifies that the ACM scheme can increase the system throughput.

• Journal of Broadcast Engineering
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• v.8 no.2
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• pp.204-208
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• 2003

In this paper, a new matrix vector formulation for a wavelet-based subband decomposition is introduced. This formulation provides a means to compute a regular multi-resolution analysis over many levels of decomposition. With this approach. any single channel linear space-invariant filtering problem can be cast into a multi-channel framework. This decomposition Is applied to the linear space-invariant image restoration problem and propose a frequency-adaptive constrained least squares(CLS) filter. In the proposed filter, we use different parameters adaptively according to subband characteristics. Experimental results are presented for the proposed frequency-adaptive CLS filter These experiments show that if accurate estimates of the subband characteristics are available, the proposed frequency adaptive CLS filter provides significant improvements over the traditional single channel filter.

• Journal of the Korea Society of Computer and Information
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• v.13 no.5
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• pp.111-116
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• 2008

Radar system must estimate exactly quickness and target in interference channel. Because interference of radio channel is multipath channel by artificial structure and nature structure. signal estimation is difficult. As long as, get rid of interference signal have been study digital beamforming, adaptive array antenna and so on. In this paper, proposed SPT-SALCMV beamforming algorithm get rid of coherent interference algorithm and adaptive array antenna. Adaptive array forms null pattern and reduces gains for direction of interference signal. And estimate signal that want by keeping gains of beam pattern changelessly to target signal direction. In this paper, proposed SPT-SALCMV algorithm was exactly received position of target. But general SPT-LCMV algorithm resulted beam error about 30degrees. Therefore, proved that SPT-SALCMV algerian that propose in this paper is more excellent than genaral SPT-LCMV algorithm.

• The KIPS Transactions:PartC
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• v.16C no.4
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• pp.477-486
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• 2009

This paper proposes a dynamic bandwidth allocation (DBA) algorithm with adaptive maximum transmission window (DBA-AMTW) to increase channel utilization in Ethernet passive optical networks (EPONs). A polling mechanism in EPON determines channel utilization and puts constraints on DBA algorithm and scheduling. DBA algorithms based on interleaved polling with stop (IPS) allocate transmission windows to optical network units (ONU) considering requests of all ONUs. However channel idle time when any ONU does not transmit packets decreases channel utilization. Proposed DBA-AMTW improves efficiency of a network and allocates transmission windows effectively by appropriate DBA computation from REPORT messages of all ONUs. An adaptive maximum transmission window for each ONU determined by a DBA computation in the previous scheduling cycle. Simulation results show that the proposed DBA algorithm improves performance of throughput and average delay time.

• Journal of Korea Multimedia Society
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• v.13 no.7
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• pp.973-982
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• 2010

In this paper, we proposed the single-channel adaptive noise canceller for the enhancement of heart sound (HS) in the auscultation signal. In case of either normal or emergency, a HS diagnosis is difficult due to the various signal source in the chest. Therefore, the HS enhancement is necessary. The conventional active noise canceller(ANC) has two channel, main signal and reference signal. For signal channel, the reference signal in ANC was generated by the proposed HS analyser and BS-Gate based on the characteristic of HS. This reference signal is suitable to the ANC condition. Experimental data were acquisited from MP36, SS30L in BIOPAC Inc., By the experiment, we confirmed that the proposed single-channel ANC was efficient for HS enhancement. And by the comparison with active linear enhancement, it was validate that the proposed ANC is not affected by the variation of a heartbeat.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9A
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• pp.830-837
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• 2005

ISI is caused by delay spread in the multipath channel environment. There are two kinds of channel equalizer: Linear and Non-Linear type according to the structures. In this paper, we propose an improved adaptive linear equalizer to mitigate ISI. The proposed adaptive equalizer is constructed by using asymmetrical Dsmvenu filter based on USE sub-optimal receiver. Asymmetrical structure of the transversal filter is realized by moving the main tap position from center to side. If this structure is used, we can divide ISI to precusor and postcusor. As a result the proposed equalizer has a larger extended compensation range than conventional adaptive linear equalizer. In computer simulation, we compare the bit error rate performance of the proposed linear equalizer with the conventional one on the S-V channel which is modeled for WB systems.