• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.6
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• pp.99-105
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• 2010

The most frequently used noise model for the performance analysis of a communication system is additive white Gaussian noise. However impulsive noise model is more practical for the real communication environments, currently the performance analysis of a communication system in impulsive noise is increasing. In this paper, therefore, the performance of a cooperative system, which is recently one of the most intensive research topics, is derived in impulsive noise. We analytically derive and compare the performance of two opportunistic cooperative diversity systems which have an amplify-and-forward (AF) relaying or a decode-and-forward (DF) relaying. It is noticed that the impulsive noise component is increases with decreasing the average number of impulses in impulsive noise, consequently the performance of two systems is degraded in high SNR region. Also it is shown that the performance of the opportunistic cooperative system with DF relaying is superior to that with AF relaying.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.5
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• pp.1061-1066
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• 2014

Transmission characteristics of the sound propagation is very complicate and sparse in shallow water. To increase the performance of underwater acoustic communication system, lots of channel estimation technique has been proposed. In this paper, we proposed the channel estimation based on LMS(Least Mean Square) algorithm which has faster convergence speed than conventional sparse-aware LMS algorithms. The proposed method combines $L_p$-norm LMS with soft decision process. Simulation was performed by using the sound velocity profile which acquired in real sea trial. As a result, we confirmed that the proposed method shows the improved performance and faster convergence speed than conventional methods.

• Journal of Institute of Control, Robotics and Systems
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• v.19 no.5
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• pp.481-487
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• 2013

This paper presents an FIR (Finite Impulse Response) fixed-interval smoothing filter for fast and exact estimating state variables of a discrete nonlinear system with modeling uncertainty. Conventional IIR (Infinite Impulse Response) filter and smoothing filter can estimate state variables of a system with an exact model when the system is observable. When there is an uncertainty in the system model, however, conventional IIR filter and smoothing filter may cause large errors because the filters cannot estimate the state variables corresponding to the uncertain model exactly. To solve this problem, FIR filters that have fast estimation properties and have robustness to the modeling uncertainty have been developed. However, there is time-delay estimation phenomenon in the FIR filter. The FIR smoothing filter proposed in this paper makes up for the drawbacks of the IIR filter, IIR smoothing filter, and FIR filter. Therefore, the FIR smoothing filter has good estimation performance irrespective of modeling uncertainty. The proposed FIR smoothing filter is applied to the integrated navigation system composed of a magnetic compass based DR (Dead Reckoning) and a GPS (Global Positioning System) receiver. Even when the magnetic compass error that changes largely as the surrounding magnetic field is modeled as a random constant, it is shown that the FIR smoothing filter can estimate the varying magnetic compass error fast and exactly with simulation results.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.1
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• pp.185-190
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• 2020

The rapid aging is increasing as the shortage of medical facilities and the resulting of decline in the quality of public health. In order to ease the burden of rising medical expenses, advanced medical institutions are expanding their remote medical care to lower the cost of services. U-healthcare detects the changes in physical and chemical phenomena occurring in the human body and converts them into electrical signals that can be processed and feeds back to the results through analytical and visualization processes to select only the desired information from the measured signals. The service is provided through a process of providing an alarm to a user. However, traditional biometric methods of attaching sensors directly to the body can be annoying and rejected in daily life. Therefore, there is a need for a method of continuously measuring biometric information without causing inconvenience to daily life. In this paper, we propose an IR-UWB-based non-contact and non-responsive respiratory measurement system that can continuously monitor biological information without any inconveniences to daily life.

• Journal of Digital Contents Society
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• v.9 no.1
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• pp.53-60
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• 2008

Based on similarities between fuzzy set theory and mathematical morphology, Grabish proposed a fuzzy morphology based on the Sugeno fuzzy integral. This paper proposes a fuzzy mathematical morphology based on the defuzzification of the fuzzy measure which corresponds to fuzzy integral. Its process makes a fuzzy set used as a measure of the inclusion of each fuzzy measure for subsets. To calculate such an integral a $\lambda$-fuzzy measure is defined which gives every subsets associated with the universe of discourse, a definite non-negative weight. Fast implementable definitions for erosion and dilation based on the fuzzy measure was given. An application for robust skeletonization of two-dimensional objects was presented. Simulation examples showed that the object reconstruction from their skeletal subsets that can be achieved by using the proposed was better than by using the binary mathematical morphology in most cases.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.1
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• pp.71-77
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• 2005

In this paper we implement a headphone-based 5.1 channel 3-dimensional (3D) sound generation system using HRTF (Head Related Transfer Function). Each mono sound source in the 5.1 channel signal is localized on its virtual location by binaural filtering with corresponding HRTFs, and reverberation effect is added for spatialization. To reduce the computational burden, we reduce the number of taps in the HRTF impulse response and model the early reverberation effect with several tens of impulses extracted from the whole impulse sequences. We modified the spectrum of HRTF by weighing the difference of front-back spec01m to reduce the front-back confusion caused by non-individualized HRTF DB. In informal listening test we can confirm that the implemented 3D sound system generates live and rich 3D sound compared with simple stereo or 2 channel down mixing.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.2
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• pp.15-21
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• 2007

In this paper, we propose a technique of a practical symbol acquisition and tracking using a low complex ADC and simple digital circuits for noncoherent asynchronous impulse-radio-based Ultra Wideband (IR-UWB) receiver based on energy detection. Compared to previous approaches of detecting an exact acquisition time that require much hardware resource, the proposed technique is to detect the target symbol by finding the symbol acquisition interval per symbol with a target symbo, thus the complexity of the complete signal processing and power consumption by ADC are reduced. To do this, we define the bit decision window (BDW) and analyze the relation between SNR, hardware resource, size of BDW and BER(Bit Error Rate). Using the results, the optimum BDW size for the minimum BER with limited hardware resource is selected. The proposed synchronization technique is verified with an aid of a simulator programmed by considering practical impulse channels.

• The Journal of the Acoustical Society of Korea
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• v.9 no.2
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• pp.42-52
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• 1990

A method of measuring impulse response of acoustic system, two kinds of cross-correlation methods (the direct correlation method and the M-sequence modulation correlation method) were applied. According to the direct correlation method, by using stationary random noise source and by calculating the cross-correlation function between the sourece and the output signal, equivalent impulse response can be obtained not being influenced by the back ground noises. By applying this method, the measurement of echo-time patterns in rooms and oblique incident sound absorption characteristics of sound absorbing materials was carried out. In the case of the M-sequence modulation correlation method which was contrived by Aoshima, an intermittent random nosie modulated by M-sequence signal is used as the source signal, and the cross-correlation function between the M-sequence signal and the squared output signal is calculated. According to this method, equivalent energy impulse response (squared impulse response) of te propagation system can be obtained without being influenced by the back ground noises and the air fluctuation caused by wind. As the applcaition of this measuring method to the architectural acoustics, the meaurements of echo-time patterns, reverberation decays and sound pressure lev디 distributions in rooms and sound insulation efficiencies in buildings were carried out. From these experimental studies, it has been found that this M-sequence modulation correlation method is markedly useful especially for the field masurement of sound insulation under high back ground noise condition.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.61-74
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• 2004

In this paper, a contort- and speaker-dependent cepstrum extraction method and a channel normalization method for minimizing the loss of speaker characteristics in the cepstrum were proposed for a robust speaker recognition system over the channel. The proposed extraction method creates a cepstrum based on the pitch synchronous analysis using the inherent pitch of the speaker. Therefore, the cepstrum called the 〃pitch synchronous cepstrum〃 (PSC) represents the impulse response of the vocal tract more accurately in voiced speech. And the PSC can compensate for channel distortion because the pitch is more robust in a channel environment than the spectrum of speech. And the proposed channel normalization method, the 〃formant-broadened pitch synchronous CMS〃 (FBPSCMS), applies the Formant-Broadened CMS to the PSC and improves the accuracy of the intraframe processing. We compared the text-independent closed-set speaker identification on 56 females and 112 males using TIMIT and NTIMIT database, respectively. The results show that pitch synchronous km improves the error reduction rate by up to 7.7% in comparison with conventional short-time cepstrum and the error rates of the FBPSCMS are more stable and lower than those of pole-filtered CMS.

• The Journal of the Acoustical Society of Korea
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• v.36 no.6
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• pp.378-386
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• 2017

Accurate distance measurement between maneuver target in underwater and measuring devices is required to perform quantitative test evaluation in marine weapons system R&D process. In general, the target distance is measured using a one-way ToA (Time of Arrival) method that calculates the time difference between transmitted and received signals from the two accurately synchronized devices. However, the distance estimation performance is degraded because of the multi-path environments. In this paper, the time-variant transfer function of complex underwater environment is estimated from each received data frame using RBD (Ray-based Blind Deconvolution), and the estimated time-variant transfer function is then used to get rid of the effect about complex underwater environment and to recover the data signal using PTRM (Passive Time Reversal Mirror). The result from the simulation and experimental data show that the suggested method improve the distance estimation performance when comparing with the conventional ToA method.