• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.5C
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• pp.384-392
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• 2012

Recently, much research on blind estimation of the interleaver parameters has been performed by using Gauss-Jordan elimination to find the linearity of the block channel code. When using Gauss-Jordan elimination, the input data to be calculated needs to run as long as the square multiple of the number of the interleaver period. Thus, it has a limit in estimating the interleaver parameters with insufficient input data. In this paper, we introduce and analyze an estimation algorithm which can estimate interleaver parameters by using only 15 percent of the input data length required in the above algorithm. The shorter length of input data to be calculated makes it possible to estimate the interleaver parameters even when limited data is received. In addition, a 80 percent reduction in the number of the interleaver period candidates increases the efficiency of analysis. It is also feasible to estimate both the type and size of the interleaver and the type of channel coding.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.12
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• pp.2719-2730
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• 2009

In this paper, we implement the measurement functionality for performance measurements of EDGE (Enhanced Data Rates for GSM Evolution) terminal by using software. Generally speaking, the receiving algorithms in normal MODEM cannot be used directly to a measurement system due to the lack of accuracy. Therefore, we propose a new receiver algorithm for precise EDGE signal measurements. In the proposed algorithm, 2-stage (coarse stage, fine stage) parameters estimation (symbol-timing, frequency offset, carrier phase) scheme is used. To improve the estimation accuracy, we increase the number of the received signal samples by interpolation. The proposed EDGE signal measurement algorithm can be used for verifying the hardware measurement system, and also can be used for the commercial systems through software optimization.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.711-716
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• 2003

This paper proposes the method to select the number of optimal mixtures in each state in Continuous Density HMM (Hidden Markov Models), Previously, researchers used the same number of mixture components in each state of HMM regardless spectral characteristic of speaker, To model each speaker as accurately as possible, we propose to use a different number of mixture components for each state, Selection of mixture components considered the probability value of mixture by each state that affects much parameter estimation of continuous density HMM, Also, we use PCA (principal component analysis) to reduce the correlation and obtain the system' stability when it is reduced the number of mixture components, We experiment it when the proposed method used average 10% small mixture components than the conventional HMM, When experiment result is only applied selection of mixture components, the proposed method could get the similar performance, When we used principal component analysis, the feature vector of the 16 order could get the performance decrease of average 0,35% and the 25 order performance improvement of average 0.65%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.3
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• pp.217-223
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• 2013

A digital retrodirective antenna system is easy to modify and upgrade because it can control the phase information of the output signal toward opposite direction to input signal without a priori knowledge of the arrival direction. Due to this advantage, it is possible to perform fast beam tracking. In this paper, a design digital retrosirective array antenna system according to the number of antenna array by using only one digital PLL which finds angle of delayed phase and we test BER performance of this system. When we transmit data at actual communication system, the data modulated onto carrier frequency in order to shift spectrum from base band to another band. So we simulate system considering carrier frequency according to the number of antenna array. As a result, carrier frequency has no impact on the performance.

• Journal of IKEEE
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• v.27 no.4
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• pp.624-629
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• 2023

Estimation of signal parameters via rotational invariance techniques (ESPRIT) is an algorithm that estimates the angle of a signal arriving at an array antenna using the shift invariance property of an array antenna. ESPRIT offers the good trade-off between performance and complexity. However, the ESPRIT algorithm still requires high-complexity operations such as covariance matrix and eigenvalue decomposition, so implementation with a hardware processor is essential to estimate the angle of arrival in real time. In addition, ESPRIT processors should have high performance. The performance is related to the number of antennas, and the number of antennas required for each application are different. Therefore, we proposed an ESPRIT processor that provides 2 to 8 variable antenna configurations to meet the performance and complexity requirements according to the applied field. The proposed ESPRIT processor was designed using the Verilog-HDL and implemented on a field programmable gate array (FPGA).

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.3
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• pp.291-297
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• 2016

In this paper, we are comparative analysis of exit algorithm and proposal algorithm for desired target direction of arrival estimation in correlation signal system. Proposed algorithm in this paper is to decrease target direction of arrival an estimation error probability using bayesian, maximum posterior, and MUSIC algorithm in order to decrease direction of arrival error probability as optimize and use linear constrained minimum variance to update weight value. Through simulation, we were comparative analysis proposed algorithm and exit MUSIC algorithm. In case SNR is 10dB and antenna element arrays are 9 and 12, We show the superior performance of the proposed method relative to the class method to decrease of signal estimation error probability about 11% and 13%, respectively.

• Proceedings of the Korea Multimedia Society Conference
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• 2012.05a
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• pp.353-354
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• 2012

광대역 VoIP 네트워크 환경에서는 불안정한 네트워크 환경으로 인해 음성패킷이 불규칙적으로 수신되어 음성데이터의 재생이 원활하지 못하다. 이러한 문제점을 해결하기 위해 본 논문에서는 네트워크 상태에 따라 원활하게 음성패킷을 재생시키는 스케줄링 방식을 제안한다. 제안하는 방식은 수신단에 도착한 패킷 헤더정보를 이용해 네트워크 지터를 추정하고, 추정된 지터와 지터버퍼와 음성프레임버퍼에 존재하는 패킷수 및 음성프레임 개수, 음성클래스정보에 따라 음성프레임의 길이를 변화시켜 재생시킴으로써 수신단의 버퍼링 지연을 줄이고 출력신호의 음성왜곡을 최소화한다. 제안하는 스케줄링 방식의 성능측정을 위해 버퍼링 지연과 PESQ를 기존 음성패킷 재생 스케줄링 방식과 비교한다.

• The Journal of the Acoustical Society of Korea
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• v.38 no.6
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• pp.678-686
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• 2019

Underwater communication performance is degraded by the influence of Inter-Symbol Interference (ISI) due to multipath. Passive time reversal processing is the most effective technique for mitigating multipath, and the diversity combining method can be used to improve its performance. This paper analyzed communication performance using the beam diversity combining method, which combines signals obtained through the beam steering to various angles. Directions of arrival were estimated through the beam-time migration, which, in turn, was estimated from probe signals received by a vertical line array. The performance was analyzed based on the number and type of combinations among the estimated angles. In this paper, the data obtained from the Biomimetic Long range Acoustic Communications 2018 (BLAC18) experiment, which was conducted in the East sea, ~50 km east of Pohang, in October 2018, were used for the analysis. The output Signal to Noise Ratio (SNR) was used as communication indicators.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.1
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• pp.31-36
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• 2014

A study on the indoor positioning system has been active recently for the location based service indoors. In the 3 dimensional positioning system based on the acoustic signal and TDOA technology, the error characteristics of the estimated source position would be changed depending on the number of microphones and the pattern of the microphone array. In this paper, the estimated position error according to the measured distance error between the microphones and the signal source is analyzed, and the optimum microphone array is decided considering the estimated position error patterns and the total amount of the estimated position error.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.7
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• pp.1239-1247
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• 2015

In frequency-division duplex (FDD) massive multiple-input multiple-output (MIMO) system, the computational complexity of downlink channel estimation is proportional to the number of antennas at a base station. Therefore, effective channel estimation techniques may have to be studied. In this paper, novel channel estimation algorithms using adaptive techniques such as Kalman and least mean square (LMS) filters are proposed in a channel model with temporal and spatial correlation.