• Journal of KSNVE
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• v.11 no.3
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• pp.401-409
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• 2001

Passive type mufflers installed on every car haute inherent problem of lowering engine power and fuel efficiency caused by backpressure which is byproduct of complex internal structure. Recent improvements like installing a calve to change exhaust gas path depending on power requirement and rpm have only marginally improved performance. Tremendous amount of recent research works on active exhaust noise control have failed to commercialize because of numerous physical and economical reasons. In this paper, a unique seal-active muffler using difference of transmission paths is presented. In this system exhaust pipe is divided into two and joined again downstream. Exhaust noise is reduced by destructive interference when two-divided noise join again with transmission paths'difference which is half of the wavelength of a main noise frequency. One divided path has a sliding mechanism to change length thereby transmission path length difference is adjusted to entwine rpm change. The proposed system has minimal backpressure and does not need a secondary sound source like a speaker so it can overcome many problems of failed active noise control methods. We have verified proposed system's superior performance by simulation and comparison experiment with passive mufflers.

• Journal of Korean Ophthalmic Optics Society
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• v.6 no.2
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• pp.77-80
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• 2001

The purpose of this work is that by using speckle phenomena of low-level He-Ne laser, it analyzes the movement condition of surface elements on a substance vibrating. The lights from laser reflect against a copper plate and will be shown as speckles on a vibration board. If we vibrate these speckles by sound waves of a speaker installed on the back, the speckles change with various shapes. When a vibration board has a maximum vibration, the frequency becomes a dynamical resonance frequency and we can mark the changes of speckles at that time on a vibration board.

• ETRI Journal
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• v.27 no.2
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• pp.153-165
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• 2005

The main purpose of a spatial audio system is to give a listener the same impression as if he/she were present in a recorded environment. A dummy head microphone is generally used for such purposes. Because of its human-like shape, we can obtain good spatial sound images. However, its shape is a restriction on its public use and it is difficult to convert a 2-channel recording into multi-channel signals for an efficient rendering over a multi-speaker arrangement. In order to solve the problems mentioned above, a spatial audio system is proposed that uses multiple microphones on a rigid sphere. The system has five microphones placed on special points of the rigid sphere, and it generates audio signals for headphone, stereo, stereo dipole, 4-channel, and 5-channel reproduction environments. Subjective localization experiments show that front/back confusion, which is a common limitation of spatial audio systems using the dummy head microphone, can be reduced dramatically in 4-channel and 5-channel reproduction environments and can be reduced slightly in a headphone reproduction.

• The Journal of the Acoustical Society of Korea
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• v.24 no.2E
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• pp.71-76
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• 2005

Compared to normal speech, distant-talking speech is characterized by the acoustic effect due to interfering sound and echoes as well as articulatory changes resulting from the speaker's effort to be more intelligible. In this paper, the acoustic features for distant-talking speech due to the articulatory changes will be analyzed and compared with those of the Lombard effect. In order to examine the effect of different distances and articulatory changes, speech recognition experiments were conducted for normal speech as well as distant-talking speech at different distances using HTK. The speech data used in this study consist of 4500 distant-talking utterances and 4500 normal utterances of 90 speakers (56 males and 34 females). Acoustic features selected for the analysis were duration, formants (F1 and F2), fundamental frequency, total energy and energy distribution. The results show that the acoustic-phonetic features for distant-talking speech correspond mostly to those of Lombard speech, in that the main resulting acoustic changes between normal and distant-talking speech are the increase in vowel duration, the shift in first and second formant, the increase in fundamental frequency, the increase in total energy and the shift in energy from low frequency band to middle or high bands.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.6 no.1
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• pp.27-32
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• 2006

Verbal communication is the most commonly used mean of communication. A spoken word carries a lot of informations about speakers and their emotional states. In this paper we designed a model to recognize emotional states in a speech, a first phase of two phases in developing a toy machine that recognizes emotional states in a speech. We conducted an experiment to extract and analyse the emotional state of a speaker in relation with speech. To analyse the signal output we referred to three characteristics of sound as vector inputs and they are the followings: frequency, intensity, and period of tones. Also we made use of eight basic emotional parameters: surprise, anger, sadness, expectancy, acceptance, joy, hate, and fear which were portrayed by five selected students. In order to facilitate the differentiation of each spectrum features, we used the wavelet transform analysis. We applied ANFIS (Adaptive Neuro Fuzzy Inference System) in designing an emotion recognition model from a speech. In our findings, inference error was about 10%. The result of our experiment reveals that about 85% of the model applied is effective and reliable.

• Journal of the Korean Society of Safety
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• v.22 no.3 s.81
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• pp.7-12
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• 2007

This paper presents the development of an electrical fire detection system using digital temperature and current detectors in order to sound for electrical fire in advance. As the demand for electricity is increasing and industrial facilities are getting more complex and larger in size, the losses of human life and property are on the increase by electrical fires. In order to prevent electrical fires, it is required to find out fire signatures, or electric signal of the overcurrent and overheating. Therefore, in this paper, developed is an electrical fire detection system based on the detection of signal for overcurrent and overheating to prevent electrical accidents in advance that happen in electrical wires. The developed system gives an alarm by computer monitor, speaker system and mobile phone before electrical fires occur and give severe damages to human beings and properties, and the system can be implemented and supplied for business and residental buildings at a low price. The usefulness and validity of the system, also, verified in this paper by case study and experiments.

• Transactions of the Korean Society of Automotive Engineers
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• v.2 no.6
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• pp.29-37
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• 1994

Active noise control of random noise which propatate in the vehicle cabin as a form of spherical wave is the target of this study. In the previous study, the adaptive algorithm for adaptive controller is presented for the application in active noise control system. And for the preliminary study of adaptive active noise control in vehicle cabin as a real system, a computer simulation is performed on the effectiveness of the adaptive algorithm in the amplitude of the pressure fluctuation. This work studies the implementation of multi-channel feedforward adaptive algorithm for the reduction of the noise inside a vehicle cabin using a number of secondary sources derived by adaptive filtering of reference noise source. Multi-channel adaptive feedforward algorithm are verified in numerical simulation and semi-experimental justification of developed system is made on a domestic passenger car. In the results of semi-experimental study, the noise of specific region in the interior of automobile are reduced for the appreciabe sound pressure level in the operating engine rpm and finally this study suggests the capabilities of the real time active noise control in 3 dimensional acoustic fields.

• Phonetics and Speech Sciences
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• v.11 no.1
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• pp.1-8
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• 2019

Formant trajectories reflect the continuous variation of speakers' articulatory movements over time. This study examined formant trajectories of English vowels produced by ninety-three American men and women; the values were normalized using the scale function in R and compared using generalized additive mixed models (GAMMs). Praat was used to read the sound data of Hillenbrand et al. (1995). A formant analysis script was prepared, and six formant values at the corresponding time points within each vowel segment were collected. The results indicate that women yielded proportionately higher formant values than men. The standard deviations of each group showed similar patterns at the first formant (F1) and the second formant (F2) axes and at the measurement points. R was used to scale the first two formant data sets of men and women separately. GAMMs of all the scaled formant data produced various patterns of deviation along the measurement points. Generally, more group difference exists in F1 than in F2. Also, women's trajectories appear more dynamic along the vertical and horizontal axes than those of men. The trajectories are related acoustically to F1 and F2 and anatomically to jaw opening and tongue position. We conclude that scaling and nonlinear testing are useful tools for pinpointing differences between speaker group's formant trajectories. This research could be useful as a foundation for future studies comparing curvilinear data sets.

• Journal of Digital Contents Society
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• v.17 no.2
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• pp.71-79
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• 2016

This study is intended to compare the performances of sound source localization methods used for stable automobile control by improving voice recognition rate in automobile environment and suggest how to improve their performances. Generally, sound source location estimation methods employ the TDOA algorithm, and there are two ways for it; one is to use a cross correlation function in the time domain, and the other is GCC-PHAT calculated in the frequency domain. Among these ways, GCC-PHAT is known to have stronger characteristics against echo and noise than the cross correlation function. This study compared the performances of the two methods above in automobile environment full of echo and vibration noise and suggested the use of a median filter additionally. We found that median filter helps both estimation methods have good performances and variance values to be decreased. According to the experimental results, there is almost no difference in the two methods' performances in the experiment using voice; however, using the signal of a song, GCC-PHAT is 10% more excellent than the cross correlation function in terms of the recognition rate. Also, when the median filter was added, the cross correlation function's recognition rate could be improved up to 11%. And in regarding to variance values, both methods showed stable performances.

• Journal of Sasang Constitutional Medicine
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• v.16 no.1
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• pp.61-73
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• 2004

This study was prepared to investigate the correlation between Sasang constitutional groups and voice characteristics using voice analysis system(in this study, CSL). I focused on the voice characteristics in terms of harmonics, Formant frequency and Formant Bandwidth. The subjects were 71 males. I classified them into three groups, that is Soeumin group, Soyangin group and Taeumin group. The classification method of Constitution used two ways, QSCCII(Questionnarie for the Sasang Constitution Classification II) and Interview with a specialist in Sasang Constitution. So 71 people were categorized into 31 Soeumin(people), 18 Soyangin(people) and 22 Taeumin(people). Pitch is approximately similar to the fundamental frequency(F0) in voices. Shimmer in dB gives an evaluation of the period-to-period variability of the peak-to-peak amplitude within the analyzed voice sample. FFT(Fast Fourier Transform) method in CSL can display sampled voices into harmonics. H1 is the first peak and h2 is the second peak in the harmonics. The amplitude difference of h1 and h2(h1-h2) can be explained as the speaker's phonation type, And Formant frequency and bandwidth can be explained as the speaker's vocal tract. So I checked the harmonics and Formant frequency and Bandwidth as the voice parameters. First I have captured /e/ voices from all subjects using microphone. And then I analyzed /e/ voices with CSL. Power Spectrum and Formant History is the menu in the CSL which can display harmonics and Formant frequency and bandwidth. The results about the correlation between Sasang Constitutional Groups and voice parameters are as follows; 1. There is no significant amplitude difference of harmonics(h1-h2) among three groups. 2. There is the significant difference between Soeumin Group and Soyangin Group in Formant Frequency 1 and Formant Bandwidth 1(p<0.05). Any other parameters have no significance. I assume that Soyangin Group has clearer and brighter voice than Soeumin Group according to the Formant Bandwidth difference. And I think its result has coincidence with the context of "Dongyi Suse Bowon" and "Sasangimhejinam".