• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.9A
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• pp.1290-1298
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• 2000

Turbo Code decoder is an iterate decoding technology, which extracts extrinsic information from the bit to be decoded by calculating both forward and backward metrics, and uses the information to the next decoding step Turbo Code shows excellent performance, approaching Shannon Limit at the view of BER, when the size of Interleaver is big and iterate decoding is run enough. But it has the problems which are increased complexity and delay and difficulty of real-time processing due to Interleaver and iterate decoding. In this paper, it is analyzed that MAP(maximum a posteriori) algorithm which is used as one of Turbo Code decoding, and the factor which determines its performance. MAP algorithm proceeds iterate decoding by determining soft decision value through the environment and transition probability between all adjacent bits and received symbols. Therefore, to improve the performance of MAP algorithm, the trust between adjacent received symbols must be ensured. However, MAP algorithm itself, can not do any action for ensuring so the conclusion is that it is needed more algorithm, so to decrease iterate decoding. Consequently, MAP algorithm and Turbo Code performance are analyzed in the nongaussian channel applying Robust equalization technique in order to input more trusted information into MAP algorithm for the received symbols.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.2 no.1
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• pp.45-51
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• 1998

In the realization of a high speed digital transmission over several tens kbps in mobile communications, it follow delay distortion on a count of transmission path variation by multipath interference waves. Therefore it is inevitably necessary to establish the techniques overcoming the frequency selective fading. In this paper, a feedback running RLS algorithm which is improved the RLS algorithm using exponential weighting factor, is proposed for the control algorithm of DFE and by the computer simulation method, and the performance characteristics of DFE is analyzed for the one-ray and two-ray waves model under the random Gaussian noise. The computer simulation results of proposed RLS algorithm for the $\pi$/4QPSK signal with one symbol time delayed interference signal of S/I = 20［dB］, under the S/N = 10［dB］ with $\lambda$= 0.9 showed excellent following characteristics and equalization characteristics.

• 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.

• Journal of Navigation and Port Research
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• v.36 no.3
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• pp.169-173
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• 2012

In this paper, we propose an equalizer to minimize the inter-symbol interference when PSSK(Phase Silence Shift Keying) technique is applied to the low power underwater acoustic communication. PSSK is a QPSK(Quadrature Phase Shift Keying) modulation combined with PPM(Pulse Position Modulation), and it was proposed for low power communication. However, it has poor performance due to delay spread of underwater channel. In this paper, we propose a decision feedback equalizer to minimize the error in PSSK receiver. The sea trial was performed to evaluate the performance of the proposed method. In the result, the BER of PSSK was $4.36{\times}10^{-2}$ before the equalizer was applied, but the BER of PSSK was $3.95{\times}10^{-4}$ after the proposed equalizer was applied.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10A
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• pp.1034-1041
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• 2007

In this paper, we propose a FD USE (frequency domain minimum mean square error) equalizer algorithm for MBOK DS-UWB (M-ary bi-orthogonal keying direct sequence UWB) systems considered as a PHY proposal for high-speed wireless communication in IEEE 802.15.TG3a. The conventional FD MMSE equalization scheme has a structural limit due to insertion of the cyclic prefix (CP) in all transmit packets, but the proposed scheme is able to equalize the channel effect without CP. In order to overcome channel estimation error by multipath delay, we introduce a moving FFT and a moving average scheme. Compared with conventional FD MMSE equalizer and the traditional TD (time domain) MMSE-RAKE receiver, the proposed FD MMSE equalizer has better BER performance and we demonstrate this result by computer simulation.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.3
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• pp.99-105
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• 2015

This paper deals with the mMMA (modified MMA) which possible to improving the convergence speed that employing the serial connecting form of two stage digital filter instead of signal filter of MMA adaptive equalizer without applying the variable step size for compensates the intersymbol interference by channel distortion in the nonconstant modulus signal. The adaptive equalizer can be implemented by signal digital filter using the finite order tap delay line. In this paper, the equalizer is implemented by the two stage serial form and the filter coefficient are updated by the error signal using the same algorithm of MMA in each stage. The fast convergence speed is determined in the first stage, and the residual isi left at the output of first stage output is minimized in the second stage filter. The same digital filter length was considered in single stage and two stage system and the performance of these systems were compared. The performance index includes the output signal constellation, the residual isi and maximum distortion, MSE that is measure of the convergence characteristics, the SER. As a result of computer simulation, mMMA that has a FIR structure of two stage, has more good performance in every performance index except the constellation diagram due to equalization noise and improves the convergence speed about 1.5~1.8 time than the present MMA that has a FIR structure of single stage.

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.167-174
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• 2016

Underwater communication is difficult to increase the communication capacity because the carrier frequency is lower than that of radio communications on land. This is limited to the bandwidth of the signal under the influence of the characteristics of an ocean medium. As the high transmission speed and large transmission capacity have become necessary in the limited frequency range, the studies on MIMO (Multiple Input Multiple Output) communication have been actively carried out. The performance of the MIMO communication is lower than that of the SIMO (Single Input Multiple Output) communication because cross-talk occurs due to multiusers along with inter symbol interference resulting from the channel characteristics such as delay spread and doppler spread. Although the adaptive equalizer considering multi-channels is used to mitigate the influence of the cross-talk, the algorithm is normally complicated. In this paper, time reversal mirror technique with the characteristic of a self-equalization will be applied to simplify the compensation algorithm and relieve the cross-talk in order to improve the communication performance when the signal transmitted from two channels is received over interference on one channel in the same time. In addition, the performance of the MIMO communication based on the time reversal mirror is verified using data from the SAVEX15(Shallow-water Acoustic Variability Experiment 2015) conducted at the northern area of East China Sea in May 2015.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.19 no.12
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• pp.569-573
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• 2018

Music is composed of the sound of many instruments. Among them, the sound of the human voice naturally stands out to us and immediately connects with the listener. However, A lot of different steps go into perfectly mixing a vocal, but I'm going to focus on the most important step, equalization. In this paper, starting with the concept and the type of EQ for the requirements associated with the EQ's work and will know about when and how to use subtractive EQ, additive EQ during the recording and mixing process. EQ is one if the most important tools for mixing, especially when dealing with vocals. The control that EQ's offer allows you work, boosting and cutting to fit the vocal perfectly into the mix. The key to get a professional sounding vocal every time is to always keep in mind what you're trying to achieve stylistically and for it, using reference track is very effective. In addition to EQing, there are a variety of complex working steps such as compression, reverb, chorus, delay, adjusted for the effects of the work and harmonies of backing vocals and that are also very important task. The work of EQing is the beginning of the mixing process, among other things, need to be a detailed work throughout the consideration of the above points to its importance is greater relationship.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.1
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• pp.138-145
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• 2007

Wireless transmission system using the multipath channel is affected ISI due to the delay spread. So we use a decision feedback equalizer which consist of decision part and feedback filter for remove the ISI effectively. In this paper, we propose a improved adaptive decision feedback equalizer to mitigate ISI effectively. The proposed adaptive decision feedback equalizer is construct by using soft decision device and hierarchical feedback filter based on MMSE sub-optimal equalizer using the LMS algorithm. Soft decision device mitigate the error propagation in feedback filter by incorrectly detected decision symbol and feedback filter which is divided two step independently mitigate the ISI by using a adaptive algorithm. As a result this structure shows better performance than conventional decision feedback equalizer by mitigating the error propagation in filter cause incorrectly detecting symbol. and we get the MSE more rapidly by using larger step-size due to reduce the number of feedback filter tap. In computer simulation, we compare the bit error rate performance of proposed decision feedback equalizer with conventional one on the S-V channel model for UWB system.