• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.1
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• pp.30-36
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• 2016

In this paper, an Rx-ordering aided sphere decoding algorithm for generalized spatial modulation (GSM) systems is proposed. In the proposed sphere decoding algorithm, to efficiently reduce the search region, the received signals are optimally ordered, which results in the reduction of computational complexity. To evaluate the performance and the computational complexity of the proposed Rx-ordered sphere decoding algorithm, the simulations are performed. In the simulation results, it is observed that in GSM systems, the proposed decoding algorithm achieves the same error performance with the conventional SD, whereas it efficiently decreases the computational complexity for symbol detection.

• The Transactions of the Korean Institute of Power Electronics
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• v.7 no.6
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• pp.554-562
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• 2002

This paper presents a new sensorless control scheme using simple duty signal feedback technique in DC/DC converters. The proposed Duty Feedback Control(DFC) has the characteristics that they show the same as operation performance of current mode control by using duty feedback technique without current sensor as well as present faster dynamic response performance than conventional Sensorless Current Mode(SCM) control in case that input source is perturbed by step change or DC input source includes the harmonics. Also, the proposed control scheme has good noise immunity and simple control circuits since they have one feedback loop, and can be applied to all DC/DC converters. The concept and control principles of the proposed control scheme are explained in detail and the validity of the proposed control scheme is verified through several interesting simulated and experimental results.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.3
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• pp.516-522
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• 2017

In this paper, we propose a maximum likelihood signal detection scheme for a generalized spatial modulation system that activates only a subset of transmit antennas among multiple antennas and transmits information through the indexes of active antennas as well as through the transmit symbols. The proposed maximum likelihood receiver extracts a set of candidate solutions based on their a posteriori probabilities to lower the computational load of the robust receiver under channel information errors. Then, the chosen candidate solutions are exploited to estimate the covariance matrix of effective noise. Simulation results show that the proposed maximum likelihood detection scheme achieves better error performance than a receiver that does not take into account the channel information errors. It is also seen that it reduces the computational complexity with the same bit error rate performance as the conventional robust maximum likelihood receiver.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.5 s.311
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• pp.52-61
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• 2006

Orthogonal wavelet tansform which is generally used in image and signal processing applications has limited performance because of lack of shift invariance and low directional selectivity. To overcome these demerits complex wavelet transform has been proposed. In this paper, we present an efficient image denoising method using dual-tree complex wavelet transform and Bernoulli-Gauss prior model. In estimating hyper-parameters for Bernoulli-Gaussian model, we present two simple and non-iterative methods. We use hypothesis-testing technique in order to estimate the mixing parameter, Bernoulli random variable. Based on the estimated mixing parameter, variance for clean signal is obtained by using maximum generalized marginal likelihood (MGML) estimator. We simulate our denoising method using dual-tree complex wavelet and compare our algorithm to well blown denoising schemes. Experimental results show that the proposed method can generate good denoising results for high frequency image with low computational cost.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2
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• pp.40-49
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• 1995

Application of the filtered-x LMS adaptive filter to active noise cancellation requires to estimate the transfer charactersitics between the output and the error signal of the adaptive canceler. In this paper, we derive the filtered-x adaptive noise cancellation algorithm and analyze its convergence behavior when the acoustic noise consists of multiple sinusoids. The results of the convergence analysis of the filtered-x LMS algorithm indicate that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct components : Phase estimation error and estimated gain. In particular, the convergence is shown to strongly affected by the accuracy of the phase response estimate. Simulation results are presented to support the theoretical convergence analysis.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.693-704
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• 2011

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music contents. MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved Study on DIS at the 96th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC, ACELP, and TCX) for low frequency regions, SBR for high frequency regions, the MPEG Surround for stereo information, and window transition technology for smoothing transition between various core coder. USAC can provide consistent sound quality for both speech and music contents and can be applied to various applications such as multi-media download to mobile devices, digital radio, mobile TV and audio books.

• The Journal of the Acoustical Society of Korea
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• v.13 no.1E
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• pp.11-19
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• 1994

Hidden Markov Models (HMM's) with explicit state duration density (HMM/SD) can represent the time-varying characteristics of speech signals more accurately. However, such an advantage is reduced in relatively smooth state duration densities or ling bounded duration. To solve this problem, we propose HMM's with global path constraint (HMM/GPC) where the transition between states occur only within prescribed time slots. HMM/GPC explicitly limits state durations and accurately describes the temproal structure of speech simply and efficiently. HMM's formed by combining HMM/GPC with HMM/SD are also presented (HMM/SD+GPC) and performances are compared. HMM/GPC can be implemented with slight modifications to the conventional Viterbi algorithm. HMM/GPC and HMM/SD_GPC not only show superior performance than the conventional HMM and HMM/SD but also require much less computation. In the speaket independent isolated word recognition experiments, the minimum recognition eror rate of HMM/GPC(1.6%) is 1.1% lower than the conventional HMM's and the required computation decreased about 57%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7C
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• pp.678-687
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• 2003

8SQAM(8-state Superposed Quadrature Amplitude Modulation) being a new modem technique for use in power and bandwidth limited digital communication system generates output signals which have a mか and continuous phase transition and a reduced envelope fluctuation by keeping correlation between amplitudes and phases of two subsequent symbols. Also, 8SQAM signal is free of inter-symbol interference(ISI), and has a compact power spectrum. Accordingly 8SQAM, as compared with a conventional 8PSK, is influenced a little by inter-modulation(IM), inter-symbol interference(ISI) and adjacent channel interference(ACI) in a nonlinearly amplified multi-channel(SCPC-FDMA) environment. In this paper, the performance of 8SQAM system in a nonlinearly amplified multi-channel interference environment is analyzed via computer simulation The simulation result shows that 8SQAM outperforms 8PSK with roll-off value of $\alpha$ = 0.3 by 2.7dB in CNR to maintain BER=1$\times$10$^{-4}$ when input back-off(IBO) of HPA is 1dB and channel space is 41.7% of the data bit rate(i.e., spectral efficiency = 2.40b/s/Hz).

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.38 no.1
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• pp.55-68
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• 2001

This paper presents a new ALU architecture to minimize glitching power consumption which is appeared in the conventional one with P(carry propagation)/G(carry generation) blocks. In general, A lot of glitches generated once are propagating into the next stage of circuits to make unnecessary power dissipation. Therefore, a new ALU architecture which removes the glitches at the output of P/G blocks is presented in this paper. If a lot of glitches at the output of P/G blocks are removed, then the signal transitions caused by glitches are reduced in the sum generation block and hence power consumption is also reduced. A latch is inserted into the conventional P/G blocks to remove the glitches at the output of P/G blocks. Latch enable signal can make a role in eliminating a lot of glitches at the P/G's outputs by controlling output enable time. Experimental results from HSPICE simulations with implementing 16-b ALU show 28% reduction in glitching power consumption with negligible delay penalty.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.15 no.10 s.89
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• pp.952-960
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• 2004

PSK(Phase Shift Keying) is useful because of the power and spectral efficient modulation. In this paper, no additional hardware will be needed to support various transmit mode in the suggested DSP scheme. We design and implement the synchronization algorithm for M-ary PSK(M=2, 4) demodulator based on DSP scheme, instead of complex analog PSK demodulator. TMS320C6203 is used as DSP. We check the all kinds of waveforms via the graph view window after software programming the emulation on the DSP tool. The result of implementation proves that demodulator using the suggested algorithm has equal performance with demodulator using analog circuits.