• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.6
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• pp.1004-1015
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• 1994

Modulation type classifier based on statistical moments has been successfully employed to classify PSK signals. Previously, the classifier developed utilizes the statistical moment of samples of the received signal phase, which may be difficult to extract from received signal. In this paper we propose a new moments-based classifier to classify PSK signals by using the moments of the demodulated signal for PSK. THe demodulated signal can be easily extracted from the conventional demodulation of PSK. The evaluation of the performance of the proposed classifier for PSK signals has been investigated in additive white Gaussian noise environment using the exact distribution of the demodulated signal. The performances of classifier in terms of probability of misclassification were evaluated. We found that the coherent system classifier gave 4dB improvement for BPSK and 3dB for QPSK over noncoherent system classifier, when the probability of misclassification is 10 and m equals to 4.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.16 no.1
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• pp.1-13
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• 2017

A national highway takes on an arterial roadway role between regions. It's traffic volume of minor approach is very lower than mainline because most of the traffic of national highway consists of through traffic. This means an actuated signal operation is more efficient on the national highway than pre-timed signal. actuated signals have detections on some or all movements except the mainline. In spite of these effects have been evaluated in previous studies, widespread propagation of actuated signal operation has not made so much in Korea. Because there are so many problems to use actuated signal as it is in side of roadway facilities. In this study, improved a actuation system for the national highway and evaluated it with field implementation.

• Journal of IKEEE
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• v.3 no.2 s.5
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• pp.186-195
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• 1999

In this paper, the RF-modules composed of front-end, frequency synthesizer, modulator/demodulator and power control multi channel WLL personal system for W-CDMA using 10 [MHz] RF channel bandwidth has been implemented and considered. The measured transmission power is 250 [mW] which is very close to the required value. The measured flatness of power at the final output stage is ${\pm}1.5[dB]$ over the required bandwidth of the receiver. In addition, it is found that the chip rate transmitting spread signal is set to 8.192 [MHz], the required rate. The frequencies of RF_LO signal and LO signal of the modulator and the demodulator measured by a frequency synthesizer are satisfied with design requirements. The operating range of the receiving strength signal indicator and AGC units shows 60 [dB] respectively. Also the measured phasor diagram and eye pattern for deciding the RF modules compatible with baseband digital signal processing part are shown good results.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.5
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• pp.91-97
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• 2016

This paper evaluates the equalization algorithm of FC-MMA (Fast Convergence-Multi Modulus Algorithm) and RMMA (Region based Multi Modulus Algorithm) for the compensation of intersymbol interference which is due to the distortion of communication channel. In order to obtain the error signal for adaptive equalization, the FC-MMA use the modified dispersion constant considering the number of signal symbol, the RMMA separates the 4 region which every symbol points are located, then reduce the symbol point based on this region into constant modulus symbol point. By applying the different principle in order to get the error signal for the updating the tap coefficient of adaptive equalizer, it has the different equalization performance by these error signal. The computer simulation was performed in order to compare the different equalization performance in this paper. The performance index includes the output signal constellation, the residual isi and maximum distortion that is for the convergence characteristics, the SER. As a result of computer simulation, RMMA has more good performance in the residual isi, maximum distortion after in steady state and SER performance than FC-MMA, but not in convergence speed to reach the steady state.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.1
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• pp.85-91
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• 2014

This paper relates with the VSDA (Variable stepsize Square contour Decision directed Algorithm) adaptive equalization algorithm that is used for the minimization of the intersymbol interference due to the distortion which occurs in the time dispersive channel for the transmission of 16-QAM signal.. In the conventional SCA, it is possible to compensates the amplitude and phase in the received signal that are mixed with the intersymbol interference by the constellatin dependent constant by using the 2nd order statistics of the transmitted signal. But in the VSDA, it is possible to the increasing the equalization performance by adding the concept of distance adjusted approach for constellation matching and the cost function of decision directed. We compare the performance of VSDA and SCA algorithm by the computer simulation. For this, the equalizer output signal constellation, residual isi, maximum distortion and MSE were used in the performace index. As a result of computer simulation, the VSDA algorithm has better than the SCA in convergence speed, but it gives nearly same equalization performance in other index.

• Journal of Digital Convergence
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• v.11 no.7
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• pp.209-214
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• 2013

Improve the recognition performance of speech recognition systems as a method for recognizing human listening skills were incorporated into the system. In noisy environments by separating the speech signal and noise, select the desired speech signal. but In terms of practical performance of speech recognition systems are factors. According to recognized environmental changes due to noise speech detection is not accurate and learning model does not match. In this paper, to improve the speech recognition feature extraction using gamma tone and learning model using acoustic model was proposed. The proposed method the feature extraction using auditory scene analysis for human auditory perception was reflected In the process of learning models for recognition. For performance evaluation in noisy environments, -10dB, -5dB noise in the signal was performed to remove 3.12dB, 2.04dB SNR improvement in performance was confirmed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39A no.5
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• pp.251-257
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• 2014

Digital RDA system is retransmit into the opposite direction of the incident signals. Digital RDA system have a disadventage that this system do not signal classification in multipath environment. because multipath signal is shown as vector sum of multipath signal, digital RDA system required complex signal process for multipath signal classification. In this paper, to solve these problem we propose hybrid digital RDA system which combination of the MUSIC algorithm and the digital RDA system. Proposed system has two modes. First mode is digital RDA mode. Secornd mode is digital beamforming mode. Digital RDA mode is used in situations where the less the impact of multipath. Digital beamforming mode is applied to multipath effects is greater. In secornd mode, we find optimal path using MUSIC algorithm. After than the proposed system uses only the optimal path. Through the proposed system in a multipath environment with digital RDA can be used to supplement a disadvantage.

• Composites Research
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• v.34 no.3
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• pp.143-147
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• 2021

Full width half maximum (FWHM) analysis of superimposed terahertz (THz) signals in the glass fiber reinforced polymer (GFRP) was studied to detect fine delamination inside GFRP. The THz signals were measured for each fine delamination size inside the GFRP using the reflection mode of the terahertz time domain spectroscopy (THz-TDS) system. Then, the FWHM of the superimposed THz signal reflected at the fine delamination was extracted. Thereafter, the complex refractive index of the GFRP was measured using transmission mode of the THzTDS system. Based on this, the FWHM of the superimposed THz signal at the fine delamination were calculated and compared with respect to the fine delamination size. From the theoretically calculated superimposed signals, the relationship between the fine delamination size and the FWHM in the superimposed THz signal was derived. Consequently, the fine delamination size could be predicted through the analysis of the FWHM extracted from the THz signal at the fine delamination.

• The Journal of the Korea institute of electronic communication sciences
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• v.17 no.3
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• pp.409-416
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• 2022

This paper relates to the blind noise separation method of time-delayed convolutive mixed signals. Since the mixed model of acoustic signals in a closed space is multi-channel, a convolutive blind signal separation method is applied and time-delayed data samples of the two microphone input signals is used. For signal separation, the mixing coefficient is calculated using an inverse model rather than directly calculating the separation coefficient, and the coefficient update is performed by repeated calculations based on secondary statistical properties to estimate the speech signal. Many simulations were performed to verify the performance of the proposed blind signal separation. As a result of the simulation, noise separation using this method operates safely regardless of convolutive mixing, and PESQ is improved by 0.3 points compared to the general adaptive FIR filter structure.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2
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• pp.148-154
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• 2009

In audio coding, SBR technology synthesizes the high-bands using patched time-frequency information from low-bands and the correction parameters, Since SBR transmits only correction parameters for high-bands, it provides a low-rate coding of high-bands, and is used as a core module of MPEG-4 HE-AAC, SBR was originally designed for audio signal and its performance for speech signal tends to decrease, and the major reason is an excessive noise floor in high-bands which is caused by incorrect tonality computation, In this paper, a new method to determine noise floor level in an adaptive fashion according to the speech characteristics is proposed in order to solve the problem of SBR for speech signal, The proposed method maintains the compatibility with the standard SBR, and the subjective performance evaluation shows that the proposed method improves the SBR performance especially for male speech signal compared with the standard SBR.