• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.1
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• pp.121-127
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• 2010

This paper relates with the improved CMA blind adaptive equalization algorithm which uses the constellation matching method that improve the inverse modelling efficiency of a communication channel compared to the present CMA blind adaptive equalizer. The amplitude distortion can be compensated in the present CMA blind adaptive equalizer which is used for the reduction of intersymbol interference by distortion that generate such as a band limited wireless mobile channel, but in the improved adaptive alogorithm operates with the minimize the amplitude phase distortion in the output of equalizer by applying the cost function that is composition of additional signal constellation matching error terms. In order to evaluation of the inverse modeling efficiency of improved algorithm, the residual intersymbol interference and recovered signal constellation were compared by computer simulation. As a result of comparion of computer simulation, the improved algorithm has a good stability in the residual intersymbol interference in the steady state, but it has a slow convergence rate in the adaptation state in initial state.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.5
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• pp.41-46
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• 2011

Mobile Ad-Hoc networks (MANET) has been getting the spotlight in the area of the wireless networks because of its flexibility and salability, and developing of many applications with it. As a consequence, MANETs are required to process not only simple data traffics, but also multimedia traffics, which has some constraints on the performances. However, unlike wired-networks, the wireless networks have frequent route-disconnections due to some reasons such as channel errors and mobility. Such frequent link disconnections makes the delivery of multimedia traffics difficult. Even though there many researches on multimedia transmissions on various environments, the performances of multimedia transmissions over fading channel and mobility even using various routing protocols have not been evaluated. Therefore, this paper extensively simulates the MPEG-4 video transmission over fading channel environment with node's mobility. In addition, the performances with two routing protocols such as AODV and DSR are compared. Finally, the simulation results shows that DSR protocol has the better performances to deliver MPEG-4 traffic.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.20 no.5
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• pp.39-43
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• 2020

In this paper, we propose the multi-channel interference cancellation algorithm for visible light communication (VLC) in smart building. The VLC system is communication technology using visible rays that come out in Light Emitting Diode (LED) device. It has energy curtailment effect and possible to use in ubiquitous network service applications. When a large number of users communicate indoors, the performance can be reduced due to channel interference. To remove interference, at the first, the minimum mean square error (MMSE) scheme as interference cancellation methods used, and then the successive interference cancellation (SIC) is applied to obtain additional diversity gain and improve interference cancellation performance. Indoor VLC channel model is employed. The performance is evaluated in terms of bit error rate (BER). From the simulation results, it is confirmed that the proposed scheme has better BER performance compared to the previous systems. As a result, the proposed interference cancellation improves the signal quality of VLC systems by effectively removing the channel noise. The results of the paper can be applied to VLC for smart building and general communication systems.

• Journal of Broadcast Engineering
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• v.17 no.4
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• pp.640-658
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• 2012

One of the existing terrestrial multi-channel DTV service frameworks, called KoreaView, provides four programs, composed of MPEG-2 based one HD video and H.264/AVC based three SD videos within one single 6MHz frequency bandwidth. However the additional 3 SD videos can not provide enough quality due to its reduced spatial resolution and low target bitrates. In this paper, we propose a framework, which is called a terrestrial multi-channel high quality hybrid DTV service, to overcome such a weakness of KoreaView services. In the proposed framework, the three additional SD videos are encoded based on an H.264/SVC Spatial Base layer, which is compliant with H.264/AVC, and are delivered via broadcasting networks. On the other hand, and the corresponding three additional HD videos are encoded based on an H.264/SVC Spatial Enhancement layer, which are transmitted over broadband networks such as Internet, thus allowing the three additional videos for users with better quality of experience. In order to verify the effectiveness of the proposed framework, various experimental results are provided for real video contents being used for DTV services. First, the experimental results show that, when the SD sequences are encoded by the H.264/SVC Spatial Base layer at a target bitrate of 1.5Mbps, the resulting PSNR values are ranged from 34.5dB to 42.9dB, which is a sufficient level of service quality. Also it is noted that 690kbps-8,200kbps are needed for the HD test sequences when they are encoded in the H.264/SVC Spatial Enhancement layer at similar PSNR values for the same HD sequences encoded by MPEG-2 at a target bitrate of 12 Mbps.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.2
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• pp.167-174
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• 2012

The circuits including with Optical transceiver and clock data recovery, in this paper, SERDES (SERializer-DESerializer) are implemented to construct a GE-PON burst-mode transceiver supporting IEEE 802.3ah and a jig for measuring the burst-mode characteristics, that is to say, PON upstream optical transmission environment are manufactured to evaluate the performance of transceiver. we verified that the limiting amplifier compensated the gap of max. 26dB optical power by experiments. The startup acquisition lock time is 670ns in case of using VSC7123 and 2300ns in case of S2060 and the data acquisition lock time were measured to be 400ns and 600ns, respectively, in the upstream channel transmission in this work. While on the other, VSC7123 is satisfied with IEEE802.3ah recommendations.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.4
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• pp.15-22
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• 2012

This paper deals with the performance comparison and evaluation of blind adaptive equalizer, the PMCMA-MDD and DW-MCMA, that is used for compensation of the amplitude and phase distortion which occurs in the time dispersive channel. Basically, these algorithms are modification of MCMA cost function in order to obtain the fast convergence speed and reduced residual isi by taking the parallel and serial double structured and the combination of the concept of RCA for the updating the tap coefficient. We implements the algorithm of it and compare the recovered constellation, residual isi, MSE characteristics curve and SER in the signal to noise ratio given the time dispersive channel. As a result of simulation, the PMCMA-MDD has a good in recovered constellation than DW-MCMA. But in the SER, the DW-MCMA has a good than PMCMA-MDD.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.91-97
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• 2016

This paper evaluates the equalization algorithm of FC-MMA (Fast Convergence-Multi Modulus Algorithm) and RMMA (Region based Multi Modulus Algorithm) for the compensation of intersymbol interference which is due to the distortion of communication channel. In order to obtain the error signal for adaptive equalization, the FC-MMA use the modified dispersion constant considering the number of signal symbol, the RMMA separates the 4 region which every symbol points are located, then reduce the symbol point based on this region into constant modulus symbol point. By applying the different principle in order to get the error signal for the updating the tap coefficient of adaptive equalizer, it has the different equalization performance by these error signal. The computer simulation was performed in order to compare the different equalization performance in this paper. The performance index includes the output signal constellation, the residual isi and maximum distortion that is for the convergence characteristics, the SER. As a result of computer simulation, RMMA has more good performance in the residual isi, maximum distortion after in steady state and SER performance than FC-MMA, but not in convergence speed to reach the steady state.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.17 no.1
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• pp.83-88
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• 2017

This paper related with the performance evaluation of H-MMA (Hybrid-MMA) which is applying the adaptive modulus and adaptive step size concept to MMA adaptive equalization algorithm in order to reduce the intersymbol interference that is occurred in communication channel for digital code transmission. In the conventional MMA adaptive equalizer, the coefficient is updated by using the equalizer output and possible to compensation of amplitude and phase in 2nd dimensional QAM signal, the equalization performance were degraded due to fixed modulus and step size. For the overcomming the abovemensioned problem, it is possible to improving the equalization performance in the 2nd dimensional QAM signal by applying the adaptive modulus and adaptive step size propotional to equalizer output signal to the conventional MMA algorithm. The computer simulation was performed in the same channel for the compare the performance of MMA and proposed H-MMA which is proposed in this paper. As a result of simulation, the proposed H-MMA has slower convergence time in order to arriving the steady state than MMA. But after the steady state, H-MMA has more superior to the MMA in every performance index and the equalization noise was reduced.

• The Journal of the Acoustical Society of Korea
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• v.29 no.7
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• pp.458-467
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• 2010

In this paper, a new audio reproduction system was developed in which the cross-talk signals would be reasonably cancelled at an arbitrary listener position. To adaptively remove the cross-talk signals according to the listener's position, a method of tracking the listener position was employed. This was achieved using the two microphones, where the listener direction was estimated using the time-delay between the two signals from the two microphones, respectively. Moreover, room reverberation effects were taken into consideration where linear prediction analysis was involved. To remove the cross-talk signals at the left-and right-ears, the paths between the sources and the ears were represented using the KEMAR head-related transfer functions (HRTFs) which were measured from the artificial dummy head. To evaluate the usefulness of the proposed listener tracking system, the performance of cross-talk cancellation was evaluated at the estimated listener positions. The performance was evaluated in terms of the channel separation ration (CSR), a -10 dB of CSR was experimentally achieved although the listener positions were more or less deviated. A real-time system was implemented using a floating-point digital signal processor (DSP). It was confirmed that the average errors of the listener direction was 5 degree and the subjects indicated that 80 % of the stimuli was perceived as the correct directions.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.10 no.1
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• pp.44-53
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• 1999

In this paper, we propose a pilot based time-averaging channel estimation method and analyze error performances for efficient transmission of OFDM(Orthogonal Frequency Division Multiplexing) in multipath fading environment. Frequency domain channel estimations have been used in OFDM systems to compensate signal distortions due to fading on each subcarrier. The frequency domain estimation scheme uses scattered pilot to estimate channel response by simple interpolation. This implies that the estimated channel response includes signal distortions due to the noise. In this paper, we propose time-averaged channel estimation method to remove the distortion by noise. The proposed scheme can effectively remove noise components by taking time-average of the estimated channel response after estimating frequency domain channel. The computer simulations were performed to evaluate the performance of the proposed channel estimator. For the Rician channel, we compared the performance of the proposed method to that of a conventional one using channel estimation by gaussian interpolation when SER(Symbol Error Rate) = $10^{-4}$, and compared to perfect channel estimation case. The proposed method showed differences of 0.07 dB, 0.6 dB compared to perfect channel estimation and improvements of 1.7 dB, 1.9 dB for 16 QAM, 64 QAM respectively compared to conventional method.