• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.373-383
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• 2012

The performance of underwater acoustic communication system is sensitive to the Doppler shift and ISI(Inter-Symbol Interference). Therefore, the simulation algorithm needs to consider time-spread due to multipath arrivals which cause the ISI, and time-varying Doppler shift along with moving source and receiver. For this purpose, VirTEX(Virtual Time series EXperiment) based on Ray model has been developed. In this paper, VirTEX is used to compare the characteristics of ocean waveguide from the experimental data and illustrate the performance. The CIR(Channel Impulse Response) that characterizes the multipath arrivals with representative time-spread due to multipath arrivals is compared between numerically simulated and experimental probe signal. Also, the communication performance analysis for BER(Bit Error Rate) is compared between numerically simulated and experimental data signal. As a result, VirTEX can be useful as a simulation tool for evaluating the performance of underwater acoustic communication system.

• The Journal of Pediatrics of Korean Medicine
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• v.28 no.3
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• pp.31-46
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• 2014

Objectives Forsythiae Fructus treatment has been used for inflammatory and allergic diseases in Korean Medicine. Nevertheless, the mechanism of action and the cellular targets are not understood well. The pathogenesis of allergic diseases are associated with Th2 cytokines such as IL-13, MIP-$1{\alpha}$, IL-13, IL-5, GM-CSF, IL-4, TNF-${\alpha}$ and IL-6, which are secreted by the mast cells. This study was conducted to investigate the effects of Forsythiae Fructus extracts (FF) on Th2 cytokines expression and signal transduction in MC/9 mast cells. Methods In the study, MC/9 mast cells were stimulated with DNP-IgE for 24 hours and then treated separately with CsA $10{\mu}g/m{\ell}$ and varying doses of FF for one hour. MC/9 mast cells stimulated with DNP-IgE was the control group, a treatment with CsA was the positive control group and a treatment with varying doses FF was the experimental groups. The mRNA levels of IL-13, IL-5, GM-CSF, IL-4, TNF-${\alpha}$, IL-6 were analyzed with Real-time PCR. The levels of IL-13, MIP-$1{\alpha}$ were measured using enzyme-linked immunosorbent assays(ELISA). NFAT, AP-1 and NF-${\kappa}B$ p65 were examined by Western blot analysis. Results 1. FF were observed to suppress the mRNA expression of IL-13, IL-5, GM-CSF, IL-4, TNF-${\alpha}$, IL-6 in comparison to DNP-IgE control group. 2. FF also has inhibited the IL-13, MIP-$1{\alpha}$ production significantly in comparison to DNP-IgE control group. 3. Western blot analysis of transduction factors involving Th2 cytokines expression has revealed a prominent decrease of the mast cell specific transduction factors including NFAT-1, NFAT-2, c-Jun, and NF-${\kappa}B$ p65 but c-Fos. Conclusions In conclusion, the anti-allergenic activities of FF may be strongly related to the regulation of transcription factors NFAT-1, NFAT-2, c-Jun, and NF-${\kappa}B$ p65 causing inhibition of Th2 cytokines in mast cells.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.12
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• pp.1337-1344
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• 2014

The objective of an echo canceller connected to any end of a communication line such as digital subscriber line (DSL) is to compensate the outgoing transmit signal in the receiving path that the hybrid circuit leaks. The echo canceller working in a full duplex environment is an adaptive system driven by the local signal. Conventional echo canceller that implement the least mean square (LMS) algorithm provides a low computational burden but poor convergence properties. The length of the echo canceller will directly affect both the degree of performance and the convergence speed of the adaptation process. To cancel long time-varying echoes, the number of tap coefficients of a conventional echo canceller must be large, which decreases the convergence speed of the adaptive filter. This paper proposes an alternative technique for the echo cancellation in a telecommunication channel. The new technique employs the bi-directional least mean square (LMS) algorithm for adaptively computing the optimal set of the coefficients of the echo canceller, which is composed of weighted combination of both feedforward and feedback algorithms. Finally, Simulation results as well as mathematical analysis demonstrates that the proposed echo canceller has faster convergence speed than the conventional LMS echo canceller with nearly equivalent complexity of computation.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.10
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• pp.83-88
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• 2012

The pitch searching is a very important process in a vocoder. Generally, the method of pitch searching method is used by highlighting the periodicity, where a correlation is identified with the signal by changing the interval of two pulses. When the correlation value is highest, the pitch can be found by the pulse interval because it is the repetition interval with most striking period. There are many methods to solve this problem and search the pitch by dividing a frame into many subframes, but there is too much calculation to solve. A method in this paper is suggested to vary the number of subframes by predicting the amplitude change rate in a frame. If this method is applied, the general pitch searching performance will be improved because the accuracy may be enhanced without affecting the sound quality in the synthesized signal after parameter transmission; and the pitch searching time may be reduced.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.2
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• pp.218-230
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• 1993

A power line modem(PLM) which transfers data safely through power lines in houses or small offices is considered. When a power line is used for communications, transmitted signals could be affected by the channel characteristics such as frequency-selective fading, interference, and time-varying attenuation. In order to overcome these impairments, a direct sequence(DS) technique which is well known as an effective instrument against a variety of interferences and hostile channel properties is employed. Using a DS technique, however, requires more circuits such as PN code generator circuits, code modification circuits, and complicated synchronization circuits, and it also results in substantial acquisition delay. In this paper, some of these circuits are implemented via software programmed in the system controller, and the complicated synchronization circuits are replaced by simple circuits utilizing a 60 Hz power signal for synchronization. The synchronization ciruits used in this paper virtually eliminate the substantial acquisition delay, and is also designed to free influence of 60 Hz zero crossing jitters which reside in a power signal. As a result, a PLM using a DS technique is realized in the form of wall-socket plug, and the PLM hardware would be very much simplified.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.1
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• pp.80-85
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• 2009

The performance of the robot is very limited in the conventional model-based control methods when the environments around a robot are not structured or are varying dynamically. The reason for that is the methods are based on the model of the environments which is very difficult to match with the real environments and on a path planning which is complex and time-consuming. On the other hand, the behavior-based control methods are not dependant on the model of the environments nor a complex planning. In those methods, a specific behavior is coupled with a specific sensor output, so the response of a robot is quite reactive and timely in dynamic and unstructured environments. In this thesis, we propose a situation dependant behavior based control architecture, in which a robot may behave differently to the same sensor output depending on various situations. We also show some experimental results to show the feasibility of the proposed control architecture.

• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.65-70
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• 2000

It is blown that the loudness is one of the most important metrics in assessing the sound quality and a calculation method for loudness has been standardized for steady sounds. In this study, a new loudness model is suggested for dealing with the transient sound for a unified analysis of various practical sounds. A signal processing technique is introduced for this purpose, which is required for the band subdivision and the prediction of band-level change of transient sounds. In addition, models for the post-masking and the temporal integration are adopted in the analysis of the loudness of transient sounds. In order to solve the problem of the conventional loudness model in the pure-tone signal processing, a critical band filter is employed in the analysis, which consists of 47 critical filters having a filter spacing of a half of the critical bandwidth. For testing the effectiveness of the present model, the predicted responses are compared with the experimental data and it is observed that they are in good agreements.

• Journal of the Institute of Convergence Signal Processing
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• v.18 no.2
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• pp.43-48
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• 2017

The underwater transients signals contain the characteristics of complexity, time varying, nonlinear, and short duration. So it is very hard to model for these signals with reference patterns. In this paper we separate the whole length of signals into some short duration of constant length with overlapping frame by frame. The 20th LPC(Linear Predictive Coding) coefficients are extracted from the original signals using Durbin algorithm and applied to neural network. The 65% of whole signals were learned and 35% of the signals were tested in the neural network with two hidden layers. The types of the whales for sound classification are Blue whale, Dulsae whale, Gray whale, Humpback whale, Minke whale, and Northern Right whale. Finally, we could obtain more than 83% of classification rate from the test signals.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.11
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• pp.2172-2178
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• 2007

In indoor environment, the combination of the two variations, large scale(path loss) and small scale(fading), leads to non-linear variation of RSSI(received signal strength indicator) values as distance varied. This has been one of the difficulties for indoor location estimation. This paper presents new findings on indoor RSSI characterization for more accurate model building. Experiments have been done statistically to find overall trend of RSSI values at different places and times within the same room. From experiments, it has been shown that the variation of RSSI values can be determined by both spatial and temporal factors. These two factors are directly indicated by the two main parameters of path loss model. The results show that all sensor nodes which are located at different places share the same characterization value for the temporal parameter whereas different values for the spatial parameters. The temporal parameter also has a large scale variation effect that is slowly time varying due to environmental changes. Using this relationship, the characterization for location estimation can be more efficient and accurate.

• Journal of IKEEE
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• v.24 no.4
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• pp.1081-1085
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• 2020

Instantaneous frequencies (IF) of the AM-FM signal is estimated based on the inflection point detection (IPD) method. Local maxima/minima are detected using the IPD, and they are exploited to find the IF of AM and FM components, respectively. The envelope of the maxima/minima is obtained to estimate the IF of the AM part. And the distance between neighboring maxima (or minima) is used to estimate the IF of the FM component. Computer simulation shows that the proposed method properly estimates the IF of the AM and FM when the signal has fixed frequencies for both parts. In the case of the time-varying IF of the FM part, the estimated IF shows some deviation from the true IF due to the rough sampling effect of the maximum/minimum points. Thus, the post-processing such as the lowpass filtering of the estimated IF is required to refine the resulting IF estimation.