• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.1
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• pp.33-38
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• 2017

In this paper, a 4-channel common-cathode VCSEL diode driver array with 3.125 Gb/s per channel operation speed is realized. In order to achieve faster speed of the switching main driver with relatively large transistors, the transmitter array chip consists of a pre-amplifier with active inductor stage and also an input buffer with modified equalizer, which leads to bandwidth extension and reduced current consumption. The utilized VCSEL diode provides inherently 2.2 V forward bias voltage, $50{\Omega}$ resistance, and 850 fF capacitance. In addition, the main driver based upon current steering technique is designed, so that two individual current sources can provide bias currents of 3.0 mA and modulation currents of 3.3 mA to VCSEL diodes. The proposed 4-channel VCSEL driver array has been implemented by using a $0.11-{\mu}m$ CMOS technology, and the chip core occupies the area of $0.15{\times}0.18{\mu}m^2$ and dissipates 22.3 mW per channel.

• Journal of KIISE:Computer Systems and Theory
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• v.32 no.2
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• pp.55-67
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• 2005

Separation between computation and communication in system design allows the system designer to explore the communication architecture independently of component selection and mapping. In this paper we present an iterative two-step exploration methodology for bus-based on-chip communication architecture and memory allocation, assuming that memory traces from the processing elements are given from the mapping stage. The proposed method uses a static performance estimation technique to reduce the large design space drastically and quickly, and applies a trace-driven simulation technique to the reduced set of design candidates for accurate Performance estimation. Since local memory traffics as well as shared memory traffics are involved in bus contention, memory allocation is considered as an important axis of the design space in our technique. The viability and efficiency of the proposed methodology arc validated by two real -life examples, 4-channel digital video recorder (DVR) and an equalizer for OFDM DVB-T receiver.

• Journal of Advanced Marine Engineering and Technology
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• v.40 no.1
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• pp.45-49
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• 2016

The performance of underwater acoustic communication systems is sensitive to inter-symbol interference caused by delay spread developed from multipath signal propagation. The multipath nature of underwater channels causes signal distortion and error floor. In order to improve the performance, it is necessary to employ an iterative coding scheme. Of the various iterative coding schemes, turbo code and convolutional code based on the BCJR algorithm have recently dominated this application. In this study, the performance of iterative codes based on turbo equalizers with equivalent coding rates and similar code word lengths were analyzed. Underwater acoustic communication system experiments using these two coding techniques were conducted on Kyeong-chun Lake in Munkyeong City. The distance between the transmitter and receiver was 400 m, and the data transfer rate was 1 Kbps. The experimental results revealed that the performance of turbo codes is better for channeling than that of convolutional codes that use a BCJR decoding algorithm.

• Journal of Korea Multimedia Society
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• v.10 no.10
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• pp.1284-1294
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• 2007

In this paper, a blind nonlinear channel equalization is implemented by using a Modified Fuzzy C-Means (MFCM) algorithm. The proposed MFCM searches the optimal channel output states of a nonlinear channel from the received symbols, based on the Bayesian likelihood fitness function instead of a conventional Euclidean distance measure. Next, the desired channel states of a nonlinear channel are constructed with the elements of estimated channel output states, and placed at the center of a Radial Basis Function (RBF) equalizer to reconstruct transmitted symbols. In the simulations, binary signals are generated at random with Gaussian noise. The performance of the proposed method is compared with that of a hybrid genetic algorithm (GA merged with simulated annealing (SA): GASA), and the relatively high accuracy and fast searching speed are achieved.

• Journal of Korea Multimedia Society
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• v.4 no.3
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• pp.230-237
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• 2001

Adaptive filters are widely used for acoustic echo canceler, adaptive equalizer and adaptive noise canceler. Coefficients of adaptive filters are updated by NLMS algorithm. However, Coefficients are misaligned by ambient noises when they are adapted by NLMS algorithm. In this Paper, a method determined the adaptation constant by low-pass filtered instantaneous gradient vector of LMS algorithm using orthognality principles of optimal filter is proposed. At initial states, instantaneous gradient vector, that is the cross-correlation of input signals and estimation error signals, has large value because input signals are remained in estimation error signals. When an adaptive filter is conversed, the cross-correlation will be close to zero. It isn's affected by ambient noises because ambient noises are uncorrelated with input signals. Determining adaptation constant with the cross-correlation, adaptive filters can be robust to ambient noises and the convergence rate doesn't slower As results of computer simulations, it is shown that the performance of proposed algorithm is betted than that of conventional algorithms.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.22-33
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• 2005

In this paper, an efficient signal interference control technique to improve the high convergence speed of LMS algorithms is introduced in the adaptive transversal filter of DS/SS. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, is analyzed to prove theoretically the improvement of high convergence speed. According as the step-size parameter ${\mu}$ is increased, the rate of convergence of the algorithm is controlled. Also, an increase in the stop-size parameter ${\mu}$ has the effect of reducing the variation in the experimentally computed learning curve. Increasing the eigenvalue spread has the effect of controlling which is downed the rate of convergence of the adaptive equalizer. Increasing the steady-state value of the average squared error, proposed algorithm also demonstrate the superiority of signal interference control to the filter algorithm increasing convergence speed by (B+1) times due to the data-recycling LMS technique.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.6
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• pp.1231-1243
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• 1997

Digital satellite communication channels have nonlinearities with memory due to saturation characteristics of traveling wave tube amplifier in the satellite and transmitter/receiver linear filters. In this paper, we propose a network structure and a learning algorithm for complex pi-sigma network (CPSK) and exploit CPSN in the problem of equalization of nonlinear satellite channels. The proposed CPSN is a complex-valued extension of real-valued pi-sigma network that is a higher-order feedforward network with fast learning while greatly reducing network complexity by utilizing efficient form of polynomials for many input variables. The performance of the proposed CPSN is demonstrated by computer simulations on the equalization of complex-valued QPSK input symbols distorted by a nonlinear channel modeled as a Volterra series and additive noise. The results indicate that the CPSN shows good equalization performance, fast convergence, and less computations as compared to conventional higher-order models such as Volterra filters.

• Proceedings of the Korea Inteligent Information System Society Conference
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• 2007.05a
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• pp.382-388
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• 2007

In this paper, a modified Fuzzy C-Means (MFCM) algorithm is presented for nonlinear blind channel equalization. The proposed MFCM searches the optimal channel output states of a nonlinear channel from the received symbols, based on the Bayesian likelihood fitness function instead of a conventional Euclidean distance measure. Next, the desired channel states of a nonlinear channel are constructed with the elements of estimated channel output states, and placed at the center of a Radial Basis Function (RBF) equalizer to reconstruct transmitted symbols. In the simulations, binary signals are generated at random with Gaussian noise. The performance of the proposed method is compared with that of a hybrid genetic algorithm (GA merged with simulated annealing (SA): GASA), and the relatively high accuracy and fast searching speed are achieved.

• Journal of Navigation and Port Research
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• v.37 no.3
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• pp.269-273
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• 2013

Underwater acoustic channels are generally recognized as one of the most difficult communication media because of the multipath propagation, dispersion, and so on. MIMO (Multiple-input multiple-output) techniques have been actively pursued in underwater acoustic communications recently to increase the data rate over the bandwidth-limited channels. The transmit diversity techniques can be applied in this case, and one of them is Alamouti's scheme. In this paper the performances of the transmit diversity technique are evaluated via experiment. Two transmitters and two receivers were used in experiment, and the experiment was performed in indoor water tank. The error rate 5~8% was confirmed in experimental results, and these are the improved values than the error rate 14.8% for SISO(Single-input single-output) channel under same data rate condition.

• Journal of Satellite, Information and Communications
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• v.1 no.2
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• pp.63-70
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• 2006

This paper presents the results of Ka band qualification model multiplexers for COMS Payload to be launched in 2008. These are the input and output multiplexers of the satellite transponder to use available frequency resources effectively and the diplexer of the satellite antenna to use the same reflector for both transmitting and receiving frequency bands, respectively. The input multiplexer with four frequency channels has four(4) independent channel filters which consist of an 8-pole elliptic band-pass filter for high frequency selectivity and a 2-pole equalizer for group delay equalization. For low insertion loss, mass and volume reduction, manifold type os employed for output multiplexer. E-plane T-junction is used for either splitting or combining a frequency band into two sub-bands. Asymmetric inductive irises are used to tune the receiving filter easily. The electrical performance and environmental test such as vibration test, mechanical shock test, thermal vacuum test and EMC test are performed and the results of all qualification model multiplexers are compliant to the requirement of each multiplexer. Followed by this qualification, the flight model equipment will be developed.