• ETRI Journal
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• v.29 no.3
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• pp.292-299
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• 2007

In this paper, we present a speech recognition system using a throat microphone. The use of this kind of microphone minimizes the impact of environmental noise. Due to the absence of high frequencies and the partial loss of formant frequencies, previous systems using throat microphones have shown a lower recognition rate than systems which use standard microphones. To develop a high performance automatic speech recognition (ASR) system using only a throat microphone, we propose two methods. First, based on Korean phonological feature theory and a detailed throat signal analysis, we show that it is possible to develop an ASR system using only a throat microphone, and propose conditions of the feature extraction algorithm. Second, we optimize the zero-crossing with peak amplitude (ZCPA) algorithm to guarantee the high performance of the ASR system using only a throat microphone. For ZCPA optimization, we propose an intensification of the formant frequencies and a selection of cochlear filters. Experimental results show that this system yields a performance improvement of about 4% and a reduction in time complexity of 25% when compared to the performance of a standard ZCPA algorithm on throat microphone signals.

• Journal of Biomedical Engineering Research
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• v.22 no.3
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• pp.241-247
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• 2001

ABR(auditory brainstem response) is one of the audiometry which measures objective hearing threshold level by acquiring electric evoked potentials emanated from auditory nerve system responding to an auditory stimulation. However, the obtained potentials which are largely interfered by power line noise, have extremely low SNR, thus ensemble average algorithm is generally used. The purpose of this study was to investigate the effect of iteration number in ensemble average on the reduction of the power line noise. The power line noise was modeled to be a 60 Hz sinusoidal signal and the energy of the modeled signal was calculated when it was averaged. It was verified by simulation that the energy had the periodic zero points for each stimulation rate, and 60 Hz signal induced by the power line was applied to the developed ABR system to confirm that the period of zero energy point was the same with that of the simulation. By the properly selected iteration number, power line noise could be reduced and more reliable ABR could be acquired.

• The Journal of the Korea Contents Association
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• v.11 no.4
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• pp.27-32
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• 2011

This paper presents a methodology for classifying sleep disturbance using electroencephalogram (EEG) signal at sleep stage 2 and principal component analysis. For extracting initial features, fast Fourier transforms(FFT) were carried out to remove some noise from EEG signal at sleep stage 2. In the second phase, we used principal component analysis to reduction from EEG signal that was removed some noise by FFT to 5 features. In the final phase, 5 features were used as inputs of NEWFM to get performance results. The proposed methodology shows that accuracy rate, specificity rate, and sensitivity were all 100%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.5C
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• pp.673-682
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• 2004

The adaptive beam-forming is promising approach for noise reduction in hearing aids. This approach has come in the focus of interest only recently, because of the availability of new and powerful digital signal processors. The adaptation U using usually a Least Mean Squares algorithm, updates the weight vector. In this Paper, we propose a fast wavelet based adaptive algorithm using variable step size algorithm which varies adaptive constant by the change of signal environment. We compared the performance of the proposed algorithm with the known adaptive algorithm using computer simulation of multi channel adaptive bemformer in hearing aids. As the result the proposed algorithm is suitable for adaptive signal processing area using hearing aids and has advantages reducing computational complexity. And we show the beam-forming system using proposed algorithm converges stably in a sudden change of system environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1325-1331
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• 1996

In this paper, a new current-memory circuit is proposed that reduces the clock feedthrough(CFT) error voltage causing total harmonic distortion(THD) increment in switched-current(SI) systems. Using PMOS transistor in CMOS complementary, the proposed one reduces output distortion current due to the CFT errorvoltage. A proposed current-memory is designed using a 1.2.mu.m CMOS process anda 1MHz sinusoidal signal having a 68.mu.A amplitude current is applied as input (sampling frequency:20MHz). It hasbeen shown from the simulation that the output distortion current effected by the CFT error voltage is reduced by approximately 10 times the error voltage of conventional one, THD is -57dB in case ofappling 1kHz frequency input signalwith 0.5 peak signal-to-bias current ratio.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.756-763
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• 2009

The Least Mean Square (LMS) algorithm is often used to restore signal corrupted by additive noise. A major defect of this algorithm is that the excess Mean Square Error (EMSE) increases linearly according to speech signal power. This result reduces the efficiency of performance significantly due to the large EMSE around the optimum value. Choosing a small step size solves this defect but causes a slow rate of convergence. The step size must be optimized to satisfy a fast rate of convergence and minimize EMSE. In this paper, the Instantaneous Gain Control (IGC) algorithm is proposed to deal with the situation as it exists in speech signals. Simulations were carried out using a real speech signal combined with Gaussian white noise. Results demonstrate the superiority of the proposed IGC algorithm over the LMS algorithm in rate of convergence, noise reduction and EMSE.

• Journal of Korea Society of Digital Industry and Information Management
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• v.12 no.4
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• pp.25-31
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• 2016

In this research, to analyze the concentration of heavy metal ions in water, we tried to find the measuring time at which the faradaic electric currents flowing by the pure oxidation-reduction reaction from the pushing up mercury electrode of the stripping scan square wave voltammetry(SV+SWV) methods system becomes larger than the capacitance electric current. In order to do this, a method for analyzing signals using FPGA has been designed and we conducted 120 experiments using it. As a result, when the frequency of the square wave is 40Hz, The valid potential-current signal was measured from 96.6667% to 96.7155% of the end of the pulse of the forward and reverse, and the optimal signal was measured at 96.6667%. In addition, the experiment was carried out 40 times by changing the pulse height of the square wave from 10Mv to 40Mv. As a result, at a size smaller than 40Mv, there is little change in the magnitude of the potential-current, and an invalid signal was generated when it is out of this size.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.3
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• pp.494-507
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• 2011

In this paper, we propose two schemes to reduce the complexity of fixed-complexity sphere decoder (FSD) algorithm in the ordering and tree-search stages, respectively, while achieving quasi-ML performance. In the ordering stage, we propose a QR-decomposition-based FSD signal ordering based on the zero-forcing criterion (FSD-ZF-SQRD) that requires only a few number of additional complex flops compared to the unsorted QRD. Also, the proposed ordering algorithm is extended using the minimum mean square error (MMSE) criterion to achieve better performance. In the tree-search stage, we introduce a threshold-based complexity reduction approach for the FSD depending on the reliability of the signal with the largest noise amplification. Numerical results show that in 8 ${\times}$ 8 MIMO system, the proposed FSD-ZF-SQRD and FSD-MMSE-SQRD only require 19.5% and 26.3% of the computational efforts required by Hassibi's scheme, respectively. Moreover, a third threshold vector is outlined which can be used for high order modulation schemes. In 4 ${\times}$ 4 MIMO system using 16-QAM and 64-QAM, simulation results show that when the proposed threshold-based approach is employed, FSD requires only 62.86% and 53.67% of its full complexity, respectively.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2002.10a
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• pp.471-474
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• 2002

A fuel tank contains water at the bottom under the fuel. The water comes from humidity by temperature change of inside and outside of tank. So it is necessary to measure both level to check precise amount of oil. But measuring instrument for level of water and fuel is not available yet. Since the fuel is inflammable, the sensor system must not include any electric circuits in the fuel tank. Optical cable sensor can satisfy this non-explosive condition. The displacement of a float changing by water level makes bending curvature of optical cable different. As the float rise up, the optical cable is bent more and the light signal in the cable decreases. The reduction of light signal is detected and it is converted into the change of water level. The output signal from a photo diode shows the proportional relation of water level. The increase of sensor voltage as a unit of ㎷ follows the level position of the float that is located between water and gasoline in the tank.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.05a
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• pp.346-349
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• 2017

Digital image processing is widely used in a variety of areas, and noise elimination is used as the preprocessing in all the image processing processes. Degradation is occurred in the image data due to multiple reasons. Degradation is to add the noise in the image signal, and salt and pepper noise is the representative one to cause degradation. Therefore, image restoration algorithm was proposed to process with histogram weight filter and median filter by the noise density of local mask to restore the damaged image in the salt and pepper noise environment, in this article. In addition, it was compared with the existing methods using peak signal to noise ratio(PSNR) as the objective determination factor of improvement effect.