• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.243-250
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• 2010

This paper proposes a spectral modeling of Korean traditional instrument, Haegeum, using cepstral analysis to naturally describe Haegeum sounds varying with time. To get a precise result of cepstral analysis, we set the frame size to 3 periods of input signal and more cepstral coefficients are used to extract formants. The performance is enhanced by flexibly controlling the cutoff frequency of bandpass filter depending on the resonances in the synthesis process of sinusoidal components and the deleting peaks remained in the residual signal. To detect the change of pitch, we divide the input frames into silence, attack, and sustain region and determine which region the current frame is involved in. Then, the proposed method readjusts the frame size according to the fundamental frequency in the case of the current frame is in attack region and corrects the extraction errors of the fundamental frequency for the frames in sustain region. With these processes, the synthesized sounds are much more similar to the originals. The evaluation result through the listening test by a Haegeum player says that the synthesized sounds are almost similar to originals (96~100 % similar to the original sounds).

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.268-272
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• 2009

In this paper, we propose a Gaussian Mixture Model (GMM) based incremental robust adaptation with a forgetting factor for the speaker verification. Speaker recognition system uses a speaker model adaptation method with small amounts of data in order to obtain a good performance. However, a conventional adaptation method has vulnerable to the outlier from the irregular utterance variations and the presence noise, which results in inaccurate speaker model. As time goes by, a rate in which new data are adapted to a model is reduced. The proposed algorithm uses an incremental robust adaptation in order to reduce effect of outlier and use forgetting factor in order to maintain adaptive rate of new data on GMM based speaker model. The incremental robust adaptation uses a method which registers small amount of data in a speaker recognition model and adapts a model to new data to be tested. Experimental results from the data set gathered over seven months show that the proposed algorithm is robust against outliers and maintains adaptive rate of new data.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.526-533
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• 2009

This paper proposes a formant synthesis method of Haegeum sounds using cepstral envelope for spectral modeling. Spectral modeling synthesis (SMS) is a technique that models time-varying spectra as a combination of sinusoids (the "deterministic" part), and a time-varying filtered noise component (the "stochastic" part). SMS is appropriate for synthesizing sounds of string and wind instruments whose harmonics are evenly distributed over whole frequency band. Formants extracted from cepstral envelope are parameterized for synthesis of sinusoids. A resonator by Impulse Invariant Transform (IIT) is applied to synthesize sinusoids and the results are bandpass filtered to adjust magnitude. The noise is calculated by first generating the sinusoids with formant synthesis, subtracting them from the original sound, and then removing some harmonics remained. Linear interpolation is used to model noise. The synthesized sounds are made by summing sinusoids, which are shown to be similar to the original Haegeum sounds.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.653-660
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• 2009

This paper proposes a robust audio watermarking algorithm for copyright protection and improvement of sound quality after embedding a watermark into an original sound. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and divides the spectrum into a subbands. Then, it is necessary to calculate the energy of each subband and sort n subbands in descending order corresponding to its power. After calculating the energy we choose k subbands in sorted order and find p peaks in each selected subbands, and then embed a length m watermark around the p peaks. When the listeners hear the watermarked sound, they do not recognize any distortions. Furthermore, the proposed method is robust as much as Cox's method to MP3 compression, cropping, FFT echo attacks. In addition to this, the experimental results show that the proposed method is generally 10 dB higher than Cox's method in SNR (signal-to-noise ratio) aspect.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.2 no.2
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• pp.49-56
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• 2001

The possibility of a noise reduction of piezoelectric single Panels is experimentally studied. Piezoelectric single panel is basically a plate structure on which piezoelectric patch with shunt circuit is mounted. The use of piezoelectric shunt damping can reduce the transmission at resonance frequencies of the panel structure. Piezo-damping is implemented by using a newly proposed tuning method. This method is based on electrical impedance model and maximizing the dissipated energy at the shunt circuit. By measuring the electrical impedance at the piezoelectric patch bonded on a structure, an equivalent electrical model is constructed near the system resonance frequency. Resonant shunt circuit for piezoelectric shunt damping is composed of register and inductor in series, and they are determined by maximizing the dissipated energy throughout the circuit. The transmitted noise reduction performance of single Panel is tested on an acoustic tunnel. The tunnel is a tube with a square cross section and a loud speaker is mounted at one side of the tube as a sound source. Panels are mounted in the middle of the tunnel and the transmitted sound pressure across Panels is measured. By enabling the piezoelectric shunt damping noise reduction is achieved at the resonance frequencies as well. Piezoelectric single panel with piezoelectric shunt damping is a promising technology for noise reduction in a broadband frequency.

• Journal of the Korean Wood Science and Technology
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• v.36 no.3
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• pp.70-84
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• 2008

A series of the studies on the applied physical properties of domestic species have been conducted last three years. Pinus densiflora was one of the three species examined for the first year. Because the same apparatus and experimental procedures were used for all species, their results can be easily comparable. The experiments for sorption property were conducted with 20- and 80-mesh wood powder and resulted in their EMC's and sorption isotherms at various heating conditions. The thermal conductivity and diffusivity, and electric resistance and volumetric electric resistivity were measured with a thermal-wire device and a high electric resistance meter. The differences of the thermal and electric properties between quarter- and flat-sawn specimens were observed, which was partially attributed to their anatomical differences. An acoustic measurement system was used to evaluate dynamic MOE and internal friction. This paper provides the useful fundamental data for designing a wood structure, correcting a portable resistance-type moisture meter, and nondestructive testing wood.

• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.8-15
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• 1997

In this paper, a real-time operating Motion Picture Experts Group-1 (MPEG-1) audio encoder system is implemented using a TMS320C31 Digital Signal Processor (DSP) chip. The basic operation of the MPEG-1 audio encoder algorithm based on audio layer-2 and psychoacoustic model-1 is first verified by C-language. It is then realized using the Texas Instruments (Tl) assembly in order to reduce the overall execution time. Finally, the actual BSP circuit board for the encoder system is designed and implemented. In the system, the side-modules such as the analog-to-digital converter (ADC) control, the input/output (I/O) control, the bit-stream transmission from the DSP board to the PC and so on, are utilized with a field programmable gate array (FPGA) using very high speed hardware description language (VHDL) codes. The complete encoder system is able to process the stereo audio signal in real-time at the sampling frequency 48 kHz, and produces the encoded bit-stream with the bit-rate 192 kbps. The real-time operation capability of the encoder system and the good quality of the decoded sound are also confirmed using various types of actual stereo audio signals.

• The Proceedings of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.9 no.1
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• pp.23-29
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• 1995

There are many articles about the HID lamps about it's characteristics and simulation methods for lamp design or ballast design. For the electronic ballasts of HID lamps, There are some problems caused by acoustic-resonance which destabilizes the arc of HID lamps. So, some studies suggest the methods to avoid it. For example, the methods suggested are high frequence driving over 100[kHz], and mixed frequence driving which alternates high frequence and low frequence, and squarewave driving etc. This study suggests the electronic ballast of HID lamps that solves the problems of acoustic-resonance, and can control the luminance by constant current boostes chopper of which frequence is 30[kHz] and by the squarewave driving of which frequence is 55.5[Hz.] As follows, we have the good electronic ballast for HID lamps which has the characteristics of a light weight, and a function of luminance control and a high quality luminosity and which saves electrical energy.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.24 no.7
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• pp.525-536
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• 2014

In this paper, GUI program for microspeaker system simulation program was developed and verified through closed box, vent box and 6th order bandpass enclosure system. By using the pseudo loudspeaker model concept, TS parameters and rear volume of microspeaker were identified. Their suitabilities were proved by comparing test results with simulations of electrical impedance and sound pressure response curves for the three box types; closed box, vent box and 6th order bandpass box. Also, MSSP was found to be effective regardless of the microspeaker's shape, either circular or rectangular shape. MSSP can be used for the microspeaker system simulation, and can give a general prediction of such as; sound pressure level curve, electrical impedance, diaphragm velocity and displacement curve according to multiple design parameters; diaphragm mass, compliance, force factor, front and rear volume, front and rear port's diameter and length.

• The Journal of the Acoustical Society of Korea
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• v.27 no.5
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• pp.221-228
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• 2008

To estimate array shape with reference sources in SONAR systems, nearfield signal modeling is required for the reference sources near a towed array. Array shape estimation method based on the nearfield signal modeling generally exploits the spatial covariance matrix of the received reference sources. Among those method, nearfield eigenvector method uses the eigenvector corresponding to the maximum eigenvalue as a steering vector of the reference source. In this paper, we propose a simplified subspace fitting method based on the nearfield signal modeling with spherical wave modeling. Furthermore, we analyze performance of the array shape estimation methods based on the nearfield signal modeling for various environments. The results of the numerical experiments indicate that the simplified subspace fitting method and the nearfield eigenvector method with single reference source shows almost similar performance. Furthermore, the simplified subspace fitting method with 2 reference sources consistently estimates the shape of the array regardless of the incident angle of the reference sources, whereas the nearfield eigenvector method cannot apply for the case of 2 reference sources.