• Journal of Advanced Navigation Technology
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• v.22 no.4
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• pp.271-278
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• 2018

For ionospheric correction of low earth orbiter (LEO) satellites using single frequency global navigation satellite system (GNSS) receiver, ionospheric scale factor should be applied to the ground-based ionosphere model. The ionospheric scale factor can be calculated by using a NeQuick model, which provides a three-dimensional ionospheric distribution. In this study, the ionospheric scale factor is calculated by using NeQuick G model during 2015, and it is compared with the scale factor computed from the combination of LEO satellite measurements and international GNSS service (IGS) global ionosphere map (GIM). The accuracy of the ionospheric delay calculated by the NeQuick G model and IGS GIM with NeQuick G scale factor is analyzed. In addition, ionospheric delay errors calculated by the NeQuick G model and IGS GIM with the NeQuick G scale factor are compared. The ionospheric delay error variations along to latitude and solar activity are also analyzed. The mean ionospheric scale factor from the NeQuick G model is 0.269 in 2015. The ionospheric delay error of IGS GIM with NeQuick G scale factor is 23.7% less than that of NeQuick G model.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.12
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• pp.2394-2402
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• 2017

The frequency selective fading by multipaths determines a performance of underwater acoustic communication system in shallow littoral ocean. In this study, a characteristics of underwater acoustic channel and performance of multi-carrier system is evaluated in littoral ocean with a 50m deep water, an effective wave height of 0.5m and sandy mud bottom near Busan city. A multipath delay spread and time and frequency domain are presented as a function of a transmitter-to-receiver range. A bit-error-rate of a 5 channel 4FSK(Frequency Shift Keying) with a transmission rate of 1kbps, is examined and RS(Reed-Solomon) code is also adopted to remove a burst error due to time domain fading. A number of multipath are less than four and a bit-error-rate is decreased as an increase of a transmitter-to-receiver range which gives a congestion of multi-paths resulting in a decrease of time and frequency domain fading. The measured bit-error-rate is about 10-4 at greater than 600m of transmitter-to-receiver range.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.1978-1982
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• 2001

In a Multi-carrier modulation system, CP (Cyclic prefix) is inserted in the transmit tame in order to eliminate the ISI (Intersymbol Interference) and ICI (Interchannel Interference) caused by delay spread of a received signal, which in rum degrades the throughput of the system. TEQ (Time-domain equalizer) improves the system throughput by shortening the CIR (Channel Impulse Response) time and maintaining the CP length to the minimum regardless of the channel condition. In this paper, a new MMSE (Minimum Mean Square Error) TEQ algorithm is proposed and its performance is analyzed in order to speed up computing the optimum tap coefficients of the equalizer by employing Newton method.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.26 no.1
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• pp.89-96
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• 2016

TDOA(time delay of arrival) position estimate from acoustic measurement of artillery shell impact is studied in order to develop a targeting evaluation system. Impact position is calculated from the intersections of hyperbolic estimates based on the least square Taylor series method. The mathematical process of Taylor series estimation is known to be robust. However, the concern lays with the accuracy because it is sensitive to the bias caused by the randomness of measurement situation. The measurement error typically occurs from the distortion of waveform, change of travelling path, and sensor position error. For outdoor measurement, a consideration should be made on the atmospheric condition such as temperature and wind which can possibly change the trajectories of rays of sound. It produces wrong propagation time events accordingly. Ray tracing and optimization techniques are introduced in this study to minimize the bias induced by the ray of sound. The numerical simulation shows that the atmospheric correction improves the estimation accuracy.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.24-29
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• 2004

In ocean acoustic tomography (OAT), the pulse compression techniques using M-sequences are employed in the many studies for investigating the ocean structures. M-sequences can provide the good time and Doppler resolution in the process of demodulation using matched-filter. The signal-to-noise (SNR) performance at the output of receiver may be improved by manipulating received signal, i. e. coherently averaging. The processing time can be significantly reduced by using fast hadarmard transform (FHT) or fast Fourier transform (FFT). In this paper, we estimate the multipath arrival structures and delay times using the East Korean Sea experiment data and explore the compensation method for the detrimental effects on performance due to sampling rate error. We also analyze the characteristics of the ocean acoustic channels through scattering function, delay power profile, and time dispersions.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.12A
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• pp.1136-1144
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• 2010

In this paper, the effects of HARQ feedback error are evaluated in IEEE 802.16m system when the HARQ and ARQ interactions that utilize the HARQ feedback information is used. Also, the HARQ and ARQ interaction scheme considering HARQ feedback errors are proposed. The HARQ and ARQ interaction scheme improve the system throughput by using the HARQ feedback information instead of the ARQ feedback message, which reduce retransmission time. However, errors in the HARQ feedback information generate severe performance degradation. Especially, the local NAK errors between HARQ feedback error critically degrade the performance, because the local NAK errors lead the loss of ARQ blocks. We propose a channel state-based schemes for HARQ and ARQ interactions to mitigate the throughput degradation due to HARQ feedback errors. Simulation results show that the proposed scheme improves the throughput and the delay performance.

• 제어로봇시스템학회:학술대회논문집
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• 1992.10a
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• pp.652-657
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• 1992

This paper presents to accomplish successfully a multi-input real time control by applying control hierarchy for sliding mode of multi-joint manipulators whose nonlinear terms are regarded as disturbances. We- could simplify the dynamic equations of a manipulator and servo system, which are composed of linear elements and nonlinear elements, by assuming that nonlinear terms, which are Inertia term, gravity force term, Coriolis force term and centrifugal force term, are external disturbance. By simplifying that equation, we could easily obtain a control input which satisfy sliding mode of multi-input system. We proposed a new control input algorithm in order to decrease chattering by changing control input according as effect of disturbance if a control response become within allowance error range. In this experiments, we used DSP(Digital Signal Processor) controller to suppress chattering by time delay of calculation and to carry out real time control.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.4
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• pp.355-362
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• 2009

Sound source localization method based on the time delay of arrival(TDOA) is applied to many research fields such as a robot auditory system, teleconferencing and so on. When multi-microphones are utilized to localize the source in 3 dimensional space, the conventional localization methods based on TDOA decide the actual source position using the TDOAs from all microphone arrays and the detection measure, which represents the errors between the actual source position and the estimated ones. Performance of these methods usually depends on the number of microphones because it determines the resolution of an estimated position. In this paper, we proposed the localization method using spatially mapped GCC functions. The proposed method does not use just TDOA for localization such as previous ones but it uses spatially mapped GCC functions which is the cross correlation function mapped by an appropriate mapping function over the spatial coordinate. A number of the spatially mapped GCC functions are summed to a single function over the global coordinate and then the actual source position is determined based on the summed GCC function. Performance of the proposed method for the noise effect and estimation resolution is verified with the real environmental experiment. The mean value of estimation error of the proposed method is much smaller than the one based on the conventional ones and the percentage of correct estimation is improved by 30% when the error bound is ${\pm}20^{\circ}$.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.3
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• pp.125-131
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• 2002

In this paper, a new concatenation scheme of serial concatenated convolutional codes is proposed and the performance analyzed. In the proposed scheme, each of information and parity bits of outer code is entered into inner code through interleaver and deinterleaver. Therefore, the interleaver size is same as the length of input frame. Since the interleaver size of proposed type is reduced to half of the conventional Benedetto type, the interleaver delay time required for iterative decoding is reduced. In addition the multiplexer and demultiplexer are not used in the decoder of the proposed type, the complexity of decoder can be also reduced. As results of simulation, the performance of proposed type shows the better error performance as compared to that of the conventional Benedetto type in case of the same interleaver size. And it can be observed that the difference of BER performance is increased with the increase of Eb/No. In case of the same length of input frame, the proposed type shows almost same performance with Benedetto type despite that the interleaver size is reduced by half.

• The Journal of the Acoustical Society of Korea
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• v.39 no.5
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• pp.483-489
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• 2020

This paper proposes acoustic model training using self-attention for low-resource speech recognition. In low-resource speech recognition, it is difficult for acoustic model to distinguish certain phones. For example, plosive /d/ and /t/, plosive /g/ and /k/ and affricate /z/ and /ch/. In acoustic model training, the self-attention generates attention weights from the deep neural network model. In this study, these weights handle the similar pronunciation error for low-resource speech recognition. When the proposed method was applied to Time Delay Neural Network-Output gate Projected Gated Recurrent Unit (TNDD-OPGRU)-based acoustic model, the proposed model showed a 5.98 % word error rate. It shows absolute improvement of 0.74 % compared with TDNN-OPGRU model.