• Journal of the Korea Society of Computer and Information
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• v.19 no.6
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• pp.93-100
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• 2014

The objective of this study is to propose the measurement system implementation method for the evaluation and measurement of the indoor-impulsive over 170 dB noise source. For the purpose of measuring impulse noise, design and implementation constructed followed subsystems of the testing center, microphone, ear simulator, head and torso simulator and so on. Measurement systems for the accuracy and reliability of impulse noise are implemented when measuring 3 ways of measurements method by the simultaneous measurement system design. For the accuracy and reliability of three mutually indoor-impulse noise measurements were compared, three kinds of measuring methods in accordance with the peak sound pressure level and octave band. Comparing the results of data, the indoor-impulse noise by analyzing a frequency characteristic was validated in difference for the statistical significance. Result are determined by the influence of the reflected wave. Therefore, the flexible size of the interior test site while interior impulse noise measurement system was constructed. Throughout this system can be affected by parameters that are the impulse noise source and the corresponding frequency-characteristic analysis to determine the spectrum of the reflected wave. And, in the near future, indoor impulse noise measurement systems for acquisition and analysis are utilized in useful data.

• Journal of Sasang Constitutional Medicine
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• v.17 no.1
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• pp.84-102
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• 2005

Objectives The purpose of this study was to objectify the diagnosis of Sasang constitution. It was analyzed the adult female voices into 44 factors with PSSC-2004. Methods The study was conducted on the subjects inputted /a/ 2.5-3 sec of 217 adult women voices to PSSC-2004. The statistical analyses are applied to three groups: total group, under 55 year-old group, over 55 year-old group. The group of total 216 was composed with 8 Taeyangins, 60 Soyangins, 66 Taeumins and 83 Soeumins. The under 55 year-old group was composed with 6 Taeyangins, 53 Soyangins, 55 Taeumins and 76 Soeumins. The over 55 year-old group was composed with 2 Taeyangins, 7 Soyangins, 11 Taeumins and 7 Soeumins. Results 1. In total group and under 55 year-old one, Soeumin's center pitch were significantly high compared with the others(P=0.005)(P=0.019). It was significant for distinction between Taeumin and Soeumin at the result of post mortem. 2. In over 55 year-old group, Soeumin's center freq.(total) and center freq.(1) were significantly high compared with the others(P=0.024)(P=0.016). In under 55 year-old group, Soeumin's center freq.(4) was significantly high(P=0.025). While Taeumin's center freq.(3) in under 55 year-old group and center freq.(4) in over 55 year-old group were significantly high(P=0.049)(P=0.043). 3. In over 55 year-old group, Soeumin's APQ(center), APQ(2/5), APQ(3/5), APQ(4/5) and APQ(2/3) were significantly high compared with the others. It was significant for distinction between Taeumin and Soeumin at the result of post mortem(P=0.004) (P=0.003) (P=0.004) (P=0.013) (P=0.004). 4. In over 55 year-old group, Soeumin's APQ(1/3), APQ(1/2), APQ(2/2) and zeroover APQ were significantly high compared with the others (P=0.013) (P=0.003) (P=0.008) (P=0.012). 5. In over 55 year-old group, Taeyangin's octave 3 shimmer was significantly high compared with the others(P=0.000). 6. In under 55 year-old group, Taeyangin's total energy was significantly high compared with the others(P=0.040). 7. In under 55 year-old group, Taeyangin's zeroover energy sum was significantly high compared with the others(P=0.038). Conclusions From above result, there is the possibility of efficiency standard guide for constitutional diagnosis by analyzation of the voices.

• Journal of the Korean Institute of Intelligent Systems
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• v.20 no.2
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• pp.251-256
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• 2010

This paper aims for developing the intelligent robot emulating human synesthetic skills which associate a color image with sound, so that we are able to build an application system based on the principle of mutual conversion between color image and sound. As the first step, in this study, we have tried to realize a basic system using the color image to sound conversion. This study describes a new conversion method to convert color image into sound, based on the likelihood in the physical frequency information between light and sound. In addition, we present the method of converting color image into sound using color model conversion as well as histograms in the converted color model. In the basis of the method proposed in this study, we built a basic system using Microsoft Visual C++(ver. 6.0). The simulation results revealed that the hue, saturation and intensity elements of a input color image were converted into F0, harmonic and octave elements of a sound, respectively. The converted sound elements were synthesized to generate a sound source with WAV file format using Csound toolkit.

• 월간산업보건
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• s.5
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• pp.4-15
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• 1988

In order to proffer the fundamental data for the better working environment and the effective establishment of hearing conservation program on workers exposed to industrial noise, author assessed noise levels on the 42 noisy processes among 84 manufactures of 9 industries and measured noise gearing loss by the type of industries on 3,104 workers at these noisy processes from March, 1986 to Februry, 1987. The results were summarized as follows: 1. The averge of A-weight sound level of 23 processes(54.8%) and the avergae of sound level at each octave band of 14 processes(33.3%) exceeded the permissible exposure limits in 8 hours per day. 2. The noise level was the highest in process of cocking of ship building(109.1dBA), and followed by plating of steel rolling(104.3dBA), rivet of manufacture of motor vehicles(102.5dBA), shot of ship building(98.5dBA), aciding(95.7BA) and steel tubing(95.0dBA) of steel rolling, weaving of textiles(95.0dBA). 3. The permissible exposure time for the average of sound level at each octave band was only 30 minutes in the process of cocking of ship building, plating of steel rolling and rivet of manufacture of motor vehicles. 4. As a result of audiometric examination in 3,104 workers, the rate of hearing loss over 50dB at 4,000Hz was 7.3%(227 workers) and the rate of hearing loss over 41dB at 60average method was 2.9%(89 workers). 5. The prevalence of occupational hearing loss in ship building and manufacture of motor vehicle was 5.2% and it was the highest among prevalence of these 9 industries. 6. As a result of this suvery, the noise control and gearing conservation program were required especially in the industry of ship building and manufacture of motor vehicle.

• Journal of the Institute of Convergence Signal Processing
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• v.16 no.4
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• pp.175-181
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• 2015

In this paper, we propose a design method for a digital filter using software in order to analyze the radio astronomy observation data. Recently the analysis method for radio astronomy observing system is transferring from hardware to software by developing of state-of-the-art of computer system. The existing hardware system is not able to easily change the specification because it is implemented to meet special requirements and it takes a high cost and time. In case of software, however, it has an advantage to implement with small cost if open software is used, and flexibly changes to satisfy the desired specification. But, in order to analyze the massive data like radio astronomy with software, the good performance system is needed for computer. Therefore, this paper proposes a digital filter design method using software with the same performance as that of digital filter implemented with hardware in observation system which is operated by the KVN(Korean VLBI Network). To design a digital filter, the proposed method is performed with standard C language and the simulation is conducted with GNU(GNU's Not Unix) Octave and investigated to show its effectiveness. In addition, for the high speed operation of the designed digital filter, the SSE(Streaming SIMD Extensions) library is adopted for available parallel operation. By the proposed digital filter, the digital filtering is performed for the wide band observation data in the KVN observation mode, the filtering result of narrow band observation has no ripple inside of stop band, and confirmed the effectiveness of the proposed method.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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• pp.31-37
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• 1998

This paper proposes a feature extraction method using wavelet transform for speech recognition. Speech recognition system generally carries out the recognition task based on speech features which are usually obtained via time-frequency representations such as Short-Time Fourier Transform (STFT) and Linear Predictive Coding(LPC). In some respects these methods may not be suitable for representing highly complex speech characteristics. They map the speech features with same may not frequency resolutions at all frequencies. Wavelet transform overcomes some of these limitations. Wavelet transform captures signal with fine time resolutions at high frequencies and fine frequency resolutions at low frequencies, which may present a significant advantage when analyzing highly localized speech events. Based on this motivation, this paper investigates the effectiveness of wavelet transform for feature extraction of wavelet transform for feature extraction focused on enhancing speech recognition. The proposed method is implemented using Sampled Continuous Wavelet Transform (SCWT) and its performance is tested on a speaker-independent isolated word recognizer that discerns 50 Korean words. In particular, the effect of mother wavelet employed and number of voices per octave on the performance of proposed method is investigated. Also the influence on the size of mother wavelet on the performance of proposed method is discussed. Throughout the experiments, the performance of proposed method is discussed. Throughout the experiments, the performance of proposed method is compared with the most prevalent conventional method, MFCC (Mel0frequency Cepstral Coefficient). The experiments show that the recognition performance of the proposed method is better than that of MFCC. But the improvement is marginal while, due to the dimensionality increase, the computational loads of proposed method is substantially greater than that of MFCC.

• The Journal of the Acoustical Society of Korea
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• v.36 no.2
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• pp.144-150
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• 2017

Musical genre is indispensible in constructing music information retrieval system, such as music search and classification. In general, the spectral characteristics of a music signal are obtained based on a subband decomposition to represent the relative distribution of the harmonic and the non-harmonic components. In this paper, we investigate the subband decomposition parameters in extracting features, which improves musical genre classification accuracy. In addition, the linear projection methods are studied to reduce the resulting feature dimension. Experiments on the widely used music datasets confirmed that the subband decomposition finer than the widely-adopted octave scale is conducive in improving genre-classification accuracy and showed that the feature-dimension reduction is effective reducing a classifier's computational complexity.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.11 s.116
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• pp.1124-1131
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• 2006

This paper presents a novel method to determine sound power level(PWL) emitted by a travelling vehicle for road traffic noise simulation. The PWL is evaluated by the equivalent sound pressure level (SPL) measured by close proximity method and the sound power correction factor derived from the maximum SPL measured by pass-by method and the propagation attenuation of vehicle noise during the pass-by measurement. Using the method, we derive the empirical formula for PWL estimation in 1/1-octave and overall frequency bands for 8 vehicles (automobile, SUV, small truck, large bus, trailer, 3 dump trucks) tested at two road surfaces (dense graded asphalt, 30mm transverse tinning concrete) of Korean highway test road. The suggested approach, if securing sufficient data to represent the acoustic characteristics of all vehicle types, has il strong merit to be able to evaluate sound power levels for any combination of vehicle categories and traffic volumes.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.05a
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• pp.962-965
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• 2005

A high intensity progressive wave tube is installed at Korea Aerospace Research Institute (KARI) for acoustic environmental tests. The test facility has 700 mm x 800 mm cross-sectional area, and provides acoustic environment of 165 dB over the frequency range of $25Hz{\sim}10,000Hz$. The facility consists of a 6 m long acoustic wave tube, acoustic power generation systems, gases nitrogen supply systems, and acoustic control systems. This paper describes how the basic parameters of the facility and power generation systems are controlled to meet the requirement of the test. The shape and length of the tube has been designed by using the size of test objects and the wave propagation characteristics of the tube. The capacity of acoustic power generation systems is determined by the energy conversion of acoustic wave and the efficiency of acoustic modulators. Moreover, the paper introduces test run results of the tube. Overall of 163dB has been generated by using the test facility.

• Journal of KSNVE
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• v.4 no.3
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• pp.377-384
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• 1994

This paper is an investigation of the performance of absorption diffusers to suppress the vent noise emitted when high pressure gas is throttled. First, experiment for the static performance is carried out. When there is no through-flow, the insertion loss has been obtained in terms of 1/3 octave band spectrum and the effect of the number of diffusers and the thickness of the absorption material on the static performance has been obtained. And the similarity in the spectrum of the static insertion loss is confirmed by comparing two similar models with different size. Second, the dynamic performance has been obtained by experiment using blow-down of compressed air from a storage tank through an orifice of diameter 10 mm. The back pressure by the diffuser is measured and compared with that of a single diffuser. It is found that the insertion loss of asingle diffuser is very low around 3 dB at high frequencies with negative value at low frequencies. By absorption material between the diffuser tubes, however, the performance is increased considerably. Without flow the static insertion loss increases by 3 - 4 dB by doubling the thickness or the density of the absorptionmaterial. With flow, however, the dynamic insertion loss increases. While, the back pressure by the diffuser is small enough to be neglected.