• Speech Sciences
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• v.10 no.4
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• pp.189-200
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• 2003

The objective of this study was to examine two new parameters by which we could screen people with malignant vocal folds. The new parameters were the difference sums and Pearson correlation coefficients between adjacent pairs of intensity level matrices of narrow-band spectra. Audio files from the Korean Disordered Speech Database were analyzed by Praat, a speech analysis software, to obtain matrices of 400 intensity levels at 16 time points of each sustained vowel spectra. We limited our study to 12 normal subjects and 20 patients with malignant vocal folds who recorded at least three Korean vowels at a sound-proofed booth in Busan National University Hospital. Results indicated that the average coefficients of the abnormal subjects were much lower than those of the normal subjects while the average difference sums of the patients were much higher than those of the normal ones. Also, we found that the degree of the malignancy of the vocal folds was related to the coefficients and sums. However, some subjects at the initial stages of cancerous vocal folds yielded almost comparable coefficients and difference sums to those of the normal speakers. Further studies on larger databases will be desirable to set certain criteria or threshold levels for screening people with vocal fold diseases.

• Proceedings of the KIEE Conference
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• 2001.07d
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• pp.1964-1966
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• 2001

Active Noise control(ANC) is rapidly becoming the most effective way to reduce noises that can otherwise be very difficult and expensive to control. This research presents ANFIS (Adaptive Network Fuzzy Inference System) controller for adaptively noise cancelling in a duct. ANC system generates secondary control sound pressure with same amplitude and with opposite phase as noise to be eliminated. ANFIS controller is trained to optimize its parameters for adaptively cancelling noise. That is ANFIS train its parameters by gradient descent and LSE method so called hybrid method. This paper present ANFIS in active noise control which provides an improvement convergence speed and limitation of linearity condition. It can model nonlinear functions of arbitrary complexity and ANFIS can construct an input-ouput mapping based on both human knowledge in the form of Takagi and Sugeno's fuzzy if-then rules and stipulated input-output data pairs. This paper also shows that the proposed ANFIS active noise control system successfully cancelled noise.

• Tribology and Lubricants
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• v.32 no.6
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• pp.207-211
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• 2016

To interpret the perception that originates from tactile sensibility during people touch and recognize the object surfaces, this study focuses on the development of a friction model that can describe the interaction of a stylus pen sliding over the counter surfaces. In addition, the study includes several other experimental factors such as the pressure, temperature, and topology of surface, which can have an effect on the emotional user experience concerning various surfaces; this research aims to suggest a method to quantitatively evaluate the relation between these experimental parameters and emotional user experience. Accordingly, the objective of research comprises the friction characteristic technology for measurement of fine tribological behavior and a standard to quantify the emotional feedback. Existing panels or input devices that provide interaction feedback about user actions simply operate with a single frequency vibration or sound response. On the contrary, this research investigates various interaction characteristics including friction force, frequency, and surface topology synthetically. Using the developed model, which can explain the relation between the friction parameters and emotional user experience, developers can design their product in order to provide the user with expected emotional sensibility. Consequently, it can contribute to reduce the development cost about sensitivity model.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2001.11b
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• pp.1235-1241
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• 2001

Pantograph design process must be considered in terms of stability of aerodynamics and reduction of aeroacoustics. Furthermore Pantograph needs to be insensible to severe circumstance condition like typhoon, tunnel, a change of season. In this paper, robust design of panhead sections is conducted based on the Taguchi's design of experiment method. In the aeroacoustic noise analysis, an acoustic analogy using the Ffowcs Williams and Hawkings (FW-H) equation is used to calculate the flow induced sound pressure level. From the near-field CFD analysis data, the far-field noise is predicted at the positions of 25m away from panhead contact strips. Based on aerodynamic (CFD) and aeroacoustic (FW-H) analysis data, the optimal sizing and positioning ofpanhead elements are determined using robust design optimization method. Design parameters such as thickness, length and radius are controllable factors, while outdoor air temperature and atmospheric pressure are considered as uncontrollable factors in the context of Taguchi's approach. A number of CFD simulation and aeroacoustic analysis are performed based on orthogonal arrays. Using a parameter design procedure associated with signal-to-noise (SIN) ratio and sensitivity analysis, an optimal level of design parameters are extracted to minimize the disconnection ratio between contact strips and catenary system, and reduce the far-field aeroacoustic noise.

• Proceedings of the Korean Society of Propulsion Engineers Conference
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• 2008.11a
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• pp.11-14
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• 2008

Acoustic design parameters of a half-wave resonator are studied experimentally for acoustic stability in a model acoustic tube. According to standard acoustic-test procedures, acoustic-pressure signals are measured. Quantitative acoustic properties of sound absorption coefficient are evaluated and thereby, the acoustic damping capacity of the resonator is characterized. The diameter and the number of a half-wave resonator and the diameter of the tube are selected as design parameters for optimal tuning of the resonator. Optimum acoustic damping capacity is observed at smaller open area ratio as the resonator diameter increases.

• The Journal of the Korea institute of electronic communication sciences
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• v.18 no.2
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• pp.241-250
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• 2023

In this paper, when using the LMS algorithm for adaptive ultrasonic beamforming system, the effect of initial value setting on the margin of step parameters was studied. To this end, quasi-ideal beams, rotational beams with arbitrarily set beam widths were used as examples. In the numerical simulations, an arbitrary initial value(the number of sound sources fixed to any number) was set in the ultrasonic beamforming system, and the margin of the step parameter and convergence characteristics thereof were compared.

• Advances in nano research
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• v.14 no.2
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• pp.117-126
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• 2023

This experimental study investigated friction stir welding (FSW) of the acrylonitrile-butadiene-styrene (ABS) T-joint in the presence of various nano-silica levels. This study aim to handle the drawbacks of the friction stir welding (FSW) of an ABS T-joint with various quantity of nanoparticles and assess the performance of nanoparticles in the welded joint. Moreover, the relationship between the nanoparticle quantity and FSW was analyzed using response surface methodology (RSM) Box-Behnken design. The input parameters were the tool rotation speed (400, 600, 800 rpm), the transverse speed (20, 30, 40 mm/min), and the nano-silica level (0.8, 1.6, 2.4 g). The tensile strength of the prepared specimens was determined by the universal testing machine. Silica nanoparticles were used to improve the mechanical properties (the tensile strength) of ABS and investigate the effect of various FSW parameters on the ABS T-joint. The results of Box-Behnken RSM revealed that sound joints with desired characteristics and efficiency are fabricated at tool rotation speed 755 rpm, transverse speed 20 mm/min, and nano-silica level 2.4 g. The scanning electron microscope (SEM) images revealed the crucial role of silica nanoparticles in reinforcing the ABS T-joint. The SEM images also indicated a decrease in the nanoparticle size by the tool rotation, leading to the filling and improvement of seams formed during FSW of the ABS T-joint.

• MALSORI
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• no.43
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• pp.73-88
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• 2002

In this paper, for natural facial synthesis, lip-synch algorithm based on key-frame method using RBF(radial bases function) is presented. For lips synthesizing, we make viseme range parameters from phoneme and its duration information that come out from the text-to-speech(TTS) system. And we extract viseme information from Av DB that coincides in each phoneme. We apply dominance function to reflect coarticulation phenomenon, and apply bilinear interpolation to reduce calculation time. At the next time lip-synch is performed by playing the synthesized images obtained by interpolation between each phonemes and the speech sound of TTS.

• Phonetics and Speech Sciences
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• v.4 no.3
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• pp.119-125
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• 2012

This paper describes a new algorithm for encoding spectral shape and envelope in the generic mode of G.718 super-wide band (SWB). In the G.718 SWB coder, generic mode coding and sinusoidal enhancement are used for the quantization of modified discrete cosine transform (MDCT)-based parameters in the high frequency band. In the generic mode, the high frequency band is divided into sub-bands and for every sub-band the most similar match with the selected similarity criteria is searched from the coded and envelope normalized wideband content. In order to improve the quantization scheme in high frequency region of speech/audio signals, the modified generic mode by the improvement of the generic mode in G.718 SWB is proposed. In the proposed generic mode, perceptual vector quantization of spectral envelopes and the resolution increase for spectral copy are used. The performance of the proposed algorithm is evaluated in terms of objective quality. Experimental results show that the proposed algorithm increases the quality of sounds significantly.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4E
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• pp.146-155
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• 2002

We present a new technique for achieving source separation when given only a single charmel recording. The main idea is based on exploiting the inherent time structure of sound sources by learning a priori sets of time-domain basis functions that encode the sources in a statistically efficient manner. We derive a learning algorithm using a maximum likelihood approach given the observed single charmel data and sets of basis functions. For each time point we infer the source parameters and their contribution factors. This inference is possible due to the prior knowledge of the basis functions and the associated coefficient densities. A flexible model for density estimation allows accurate modeling of the observation, and our experimental results exhibit a high level of separation performance for simulated mixtures as well as real environment recordings employing mixtures of two different sources. We show separation results of two music signals as well as the separation of two voice signals.