• Korean Journal of Optics and Photonics
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• v.21 no.5
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• pp.181-184
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• 2010

A simple technique for a flexibly tunable and switchable dual wavelength-switchable erbium-doped fiber (EDF) laser using fiber Bragg gratings is proposed and demonstrated by using both linear cavity loss controlling and homogeneous gain broadening of the EDF. The oscillating wavelength switching is successfully realized by using the modulation of the acousto-optic modulator driven by the RF signal. The wavelength spacing and lasing wavelength of the proposed wavelength-switchable laser can also be controlled by using the bending-induced tension or compression strain. The proposed tunable dual wavelength-switchable fiber laser has a lot of advantages, such as a high extinction ratio larger than 40dB and a high tunability of ~7.2 nm/$m^{-1}$.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.55-62
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• 2001

In this paper, we propose a new adaptative pruning algorithm which effectively reduces the search space during the recognition process. As maximum probabilities between neighbor frames are highly interrelated, an efficient pruning threshold value can be obtained from the maximum probabilities of previous frames. The main idea is to update threshold at the present frame by a combination of previous maximum probability and hypotheses probabilities. As present threshold is obtained in on-going recognition process, the algorithm does not need any pre-experiments to find threshold values even when recognition tasks are changed. In addition, the adaptively selected threshold allows an improvement of recognition speed under different environments. The proposed algorithm has been applied to a Korean Address recognition system. Experimental results show that the proposed algorithm reduces the search space of average 14.4% and 9.14% respectively while preserving the recognition accuracy, compared to the previous method of using fixed pruning threshold values and variable pruning threshold values.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.33-40
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• 1996

In this paper, we resolved a trade-off problem of the blind equalizer based on the stop-and-go algorithm that is commonly used for QAM demodulation in CATV receiver. The stop-and-go algorithm has used the LMS(least mean square) algorithm in the updating operation of tap weights so that the structure of equalizer is simple, but there is a trade-off between convergence speed and steady state error as in the typical LMS algorithm. We used the variable step size algrithm to improve the convergence speed with the steady state error in the constant level. With respect to the same level of the steady state error, the variable step size stop-and-go algortihm improved convergence speed by about $36%{\sim}56%$ as compared with that of the constant step size algortihm.

• The Journal of the Acoustical Society of Korea
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• v.31 no.2
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• pp.79-86
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• 2012

In this paper, we propose an enhanced algorithm for affine projection algorithms which have been proposed to speed up the convergence of the conventional NLMS algorithm. Since affine projection (AP) or pseudo AP algorithms are based on the delayed input vector and error vector, they are complicated and not suitable for applying methods developed for the LMS-type algorithms which are based on the scalar error signal. We devised a variable step size algorithm for pseudo AP using the fact that pseudo AP algorithms are updated using the scalar error and that the error signal is getting orthogonal to the input signal. We carried out a performance comparison of the proposed algorithm with other pseudo AP algorithms using a system identification model. It is shown that the proposed algorithm presents good convergence characteristics under both stationary and non-stationary environments despites its low complexity.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.443-451
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• 2003

Since a codebook-based CELP coder models its excitation signal according to one of several bit rates pre-assigned to codebooks and synthesizes speech signal using codebooks, it can not support encoding of speech signal at an arbitrary bit rate in one encoder. The proposed variable bit rate speech coder encodes the excitation signal based on the bit rate assigned to a present frame of speech using one-dimensional SPIHT and wavelet transform. Also it does't need to model excitation signal (or codebook) to some types as CELP coder, and can encode excitation signal at various bit rates without exact pitch information according to user requirement. As a result, since the coder doesn't have a codebook structure, it has relatively low coder complexity and provides equal or better speech quality compared to G.729 and G.723.1 coder.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.33 no.3
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• pp.183-188
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• 1997

A method for designing a broadband transducer was investigated experimentally. Control of the resonance frequency of a Piezoelectric circular transducer with two pairs of electrodes was achieved by varying the inductance of external coil connected across the terminal of one pair of electrodes of transducer. The conductance curves of transducer in water were obtained as a function of the inductance value in mH of the coil used in the tuning. As the tuning inductance is increased in value, the resonance frequency is reduced toward the fundamental frequency of 50 kHz. This interesting result suggest that it is possible to produce a continuously tunable transducer covering a frequency range between 61.3 kHz and 121.7 kHz by varying the inductance value of external coil from 2.7 mH to 15.0 mH. One of other problems in the design and construction of such broadband transducer is the transducer efficiency, but this will be the subject of our future work.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.14 no.5
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• pp.43-49
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• 2000

In this paper, a new control method is presented to reduce acoustic resonance phenomena of metal halide lamps with electronic ballasts. A behavior of frequency controlled resonant inverter is analyzed and the transfer function is obtained. An integrator as a current controller for ballast is employed. By using both a disturbance of the lamp power and spread spectrum effects for reducing acoustic resonance, an amplitude modulation is done by using the current controller with 200[Hz] sinusoidal wave. Arc stabilities of MH 250[W] lamps are studied with changing the modulation index. Experiments with two MH 250[W] lamps, made by different manufacturers, showed the validity of the proposed method.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.51-56
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• 2000

Is this paper, we studied on the automatic speech control system in real-time windows environment using voice recognition. The applied reference pattern is the variable DMS model which is proposed to fasten execution speed and the one-stage DP algorithm using this model is used for recognition algorithm. The recognition vocabulary set is composed of control command words which are frequently used in windows environment. In this paper, an automatic speech period detection algorithm which is for on-line voice processing in windows environment is implemented. The variable DMS model which applies variable number of section in consideration of duration of the input signal is proposed. Sometimes, unnecessary recognition target word are generated. therefore model is reconstructed in on-line to handle this efficiently. The Perceptual Linear Predictive analysis method which generate feature vector from extracted feature of voice is applied. According to the experiment result, but recognition speech is fastened in the proposed model because of small loud of calculation. The multi-speaker-independent recognition rate and the multi-speaker-dependent recognition rate is 99.08% and 99.39% respectively. In the noisy environment the recognition rate is 96.25%.

• The Journal of the Acoustical Society of Korea
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• v.41 no.3
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• pp.319-325
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• 2022

The variation of utterance lengths is a representative factor that can degrade the performance of speaker verification systems. To handle this issue, previous studies had attempted to extract speaker features from various branches or to use convolution layers with different receptive fields. Combining the advantages of the previous two approaches for variable-length input, this paper proposes integrated receptive field diversification that extracts speaker features through more diverse receptive field. The proposed method processes the input features by convolutional layers with different receptive fields at multiple time-axis branches, and extracts speaker embedding by dynamically aggregating the processed features according to the lengths of input utterances. The deep neural networks in this study were trained on the VoxCeleb2 dataset and tested on the VoxCeleb1 evaluation dataset that divided into 1 s, 2 s, 5 s, and full-length. Experimental results demonstrated that the proposed method reduces the equal error rate by 19.7 % compared to the baseline.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.9
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• pp.147-152
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• 2015

Almost all existing inspection methods for catenary system have relied on dedicated track inspection cars. Two factors make it necessary to modify the conventional method: First, the climate has become increasingly similar to that of a subtropical area. In addition, high speed trains have been developed that can run as fast as 450Km/h. This paper presents a visual catenary inspection system structure that incorporates roadbed information for high-speed running environments. Conventional catenary inspection systems determined the degree of wearing only for the trolley wire. The facility inspection system presented in this paper on the other hand utilizes visual and sound data of inspection targets and adjusts inspection frequency and sensitivity depending on the geographic data.