• Journal of the Institute of Convergence Signal Processing
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• v.20 no.2
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• pp.78-83
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• 2019

Sounds based machine fault diagnosis recovers all the studies that aim to detect automatically abnormal sound on machines using the acoustic emission by these machines. Conventional methods that use mathematical models have been found inaccurate because of the complexity of the industry machinery systems and the obvious existence of nonlinear factors such as noises. Therefore, any fault diagnosis issue can be treated as a pattern recognition problem. We propose here an automatic fault diagnosis method of hand drills using discrete wavelet transform(DWT) and pattern recognition techniques such as artificial neural networks(ANN). We first conduct a filtering analysis based on DWT. The power spectral density(PSD) is performed on the wavelet subband except for the highest and lowest low frequency subband. The PSD of the wavelet coefficients are extracted as our features for classifier based on ANN the pattern recognition part. The results show that the proposed method can be effectively used not only to detect defects but also to various automatic diagnosis system based on sound.

• International journal of advanced smart convergence
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• v.5 no.4
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• pp.15-20
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• 2016

When a virtual sound source for 3D auditory system is reproduced by a linear loudspeaker array, listeners can perceive not only the direction of the source, but also its distance. Control over perceived distance has often been implemented via the adjustment of various acoustic parameters, such as loudness, spectrum change, and the direct-to-reverberant energy ratio; however, there is a neglected yet powerful cue to the distance of a nearby virtual sound source that can be manipulated for sources that are positioned away from the listener's median plane. This paper address the problem of generating binaural signals for moving sources in closed or in open environments. The proposed perceptual enhancement algorithm composed of three main parts is developed: propagation, reverberation and the effect of the head, torso and pinna. For propagation the effect of attenuation due to distance and molecular air-absorption is considered. Related to the interaction of sounds with the environment, especially in closed environments is reverberation. The effects of the head, torso and pinna on signals that arrive at the listener are also objectives of the consideration. The set of HRTF that have been used to simulate the virtual sound source environment for 3D auditory system. Special attention has been given to the modelling and interpolation of HRTFs for the generation of new transfer functions and definition of trajectories, definition of closed environment, etc. also be considered for their inclusion in the program to achieve realistic binaural renderings. The evaluation is implemented in MATLAB.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.19 no.10
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• pp.1147-1158
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• 2008

In this paper, we designed a phase-looking circuit that locks the 16.8 GHz VTDRO to a 700 MHz SAW oscillator using SPD as a phase detector Direct phase locking with loop filter alone causes the problem of lock time, so VTDRO is phase leered by loop filter with the aid of time varying square wave current generator. The current generator is related to the loop filter and needs the systematic toning. In this paper, a systematic design of the current generator and loop filter is presented. The fabricated PLDRO shows a stabilized frequency of 16.8 GHz, a output power 6.3 dBm, and a phase noise of -101 dBc/Hz at the 100 kHz offset.

• Speech Sciences
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• v.8 no.4
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• pp.139-156
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• 2001

Speech intelligibility in patients with complete dentures is an important clinical problem depending on the material used. The objective of this study was to investigate the speech of two edentulous subjects fitted with a complete maxillary prosthesis made of two different palatal materials: chrome-cobalt alloy and acrylic resin. Three patients with complete dentures in the experiment group and ten people in the controls groups participated in the experiment. CSL, Visi-Pitch were used to measure speech characteristics. The test words consisted of a simple vowel /e/, meaningless three syllabic words containing fricative, affricated and stops sounds, and sustained fricative sounds /s/ and /$\int$/. The analysis speech parameters were vowel and lateral formants, VOT, sound durations, sound pressure level and fricative frequency. Data analysis was conducted by a series of paired T-test. The findings like the following: (1) Vowel formant one of patients with complete denture is higher than that of the control group (p<0.05), while lateral formant three of patients with complete denture is lower than that of the control group (p<0.0l). (2) Patients with complete denture produced lower speech intelligibility with low fricative frequency (/$\int$/) than control group (p<0.0). The speech intelligibility of patients with metal prosthesis was higher than that of those with resin prosthesis (p<0.05). (3) Fricative, lateral and stop sound durations of patients with complete denture were longer than those of the control group (p<0.01 and p<0.05), respectively. Total sound durations of patients with metal prosthesis were similar to that of the control group (p<0.05), while those with resin prosthesis had a shorter duration (p<0.01). This implied that those with metal prosthesis had higher speech intelligibility than those with resin prosthesis. (4) Patients with complete denture had higher sound pressure levels /t/ and /c/ than the control group (p<0.01). However, sound pressure levels for /c/ of patients with metal prosthesis or resin prosthesis was similar to the control group (p<0.05). (5) Patients with complete denture had higher fundamental frequency than the control group (p<0.01).

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.10
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• pp.1867-1872
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• 2016

In the area of digital signal processing, it is necessary to improve the quality of the speech signal after removing the background noise which exists in a various real environments. The important thing to consider when removing the background noise acoustically is that to solve the problem, depending on the information of the human auditory mechanism is mainly the amplitude spectrum of the speech signal. This paper introduces the characteristics of a frequency-weighted spectrum shaping filter for the extraction of the amplitude spectrum of the speech signal with the primary purpose. Therefore, this paper proposes an algorithm using the methods of a Wiener filter and the frequency-weighted spectrum shaping filter according to the acoustic model, after extracted the amplitude spectral information in the noisy speech signal. The spectral distortion (SD) output of the proposed algorithm is experimentally improved more than 5.28 dB compared to a conventional method.

• The Journal of Korean Academy of Prosthodontics
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• v.41 no.6
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• pp.762-771
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• 2003

Statement of problem. In dentistry, the minimally prepared inlay resin-bonded fixed partial denture (FPD) made of new ceromer / fiber-reinforced composite (FRC) was recently introduced. However, the appropriate dimensions for the long-term success and subsequent failure strength are still unknown. Purpose. The aim of this study was to investigate the most fracture-resistible thickness combination of the ceromer / FRC using a universal testing machine and an AE analyzer. Material and Methods. A metal jig considering the dimensions of premolars and molars was milled and 56-epoxy resin dies, which had a similar elastic modulus to that of dentin, were duplicated. According to manufacturer's instructions, the FRC beams with various thicknesses (2 to 4 mm) were constructed and veneered with the 1 or 2 mm-thick ceromers. The fabricated FPDs were luted with resin cement on the resin dies and stored at room temperature for 72 hours. AE (acoustic emission) sensors were attached to both ends, the specimens were subjected to a compressive load until fracture at a crosshead speed of 0.5 mm/min. The AE and failure loads were recorded and analyzed statistically. Results. The results showed that the failure strength of the ceromer/FRC inlay FPDs was affected by the total thickness of the connectors rather than the ceromer to FRC ratio or the depth of the pulpal wall. Fracture was initiated from the interface and propagated into the ceromer layer regardless of the change in the ceromer / FRC ratio. Conclusion. Within the limitations of this study, the failure loads showed significant differences only in the case of different connector thicknesses, and no significant differences were found between the same connector thickness groups. The application of AE analysis method in a fiber-reinforced inlay FPD can be used to evaluate the fracture behavior and to analyze the precise fracture point.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.8
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• pp.1767-1775
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• 2003

The SAW device is extensively used as a electro$.$mechanical band­pass filter in which a two­pairs of interdigital transducers are provided over the surface of the piezoelectric substrate. For the design requirement, the central frequency and the bandwidth of the passband, and the attenuation level of the stopband region are specified. The configuration is made so as to satisfy the specification given. The central frequency is mainly determined by the distance between the pair of the finger electrodes. The design is considered as an optimization problem with which the error norm, the distance between the desired characteristics and the calculated for a given model is to be minimized. The delta function model and the electrical equivalent circuit model are utilized to represent the SAW filter characteristics. Genetic algorithm is used for optimization in which apodization of the transducer fingers is chosen as a design variable.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.7-14
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• 2003

MPEG-1 audio standard is the algorithm for the compression of high-qualify digital audio signals. The standard dictates the functions of encoder and decoder pair, and includes three different layers as the complexity and the performance of the encoder and decoder. In this paper, we implemented the real-time system of MPEG-1 audio layer III decoder(MP3) with the TMS320C541 fixed point DSP chip. MP3 algorithm uses psycho-acoustic characteristic of human hearing system, and it reduces the amount of data with eliminating the signals hard to be heard to the hearing system of human being. It is difficult to implement MP3 decoder with fixed Point DSP because of it's broad dynamic range. We implemented realtime system with fixed DSP chip by using weighted look-up tables to reduce the amount of calculation and solve the problem of broad dynamic range.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.61 no.1
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• pp.179-185
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• 2012

Since the eco-era is getting closer, the importance of noise reducing in the passenger cars of high-speed train is very important. The active noise control is best choice to reduce low frequency noise because the passive one is too heavy for high speed trains where weight is so critical. Also ANC is able to reduce the ambient noise when the environmental-factor changes. To reduce a three-dimensional closed-space sound field like a car of a high-speed rail is hard to do using single channel ANC control system. We used multi-channel FXLMS algorithm which calculation speed is fast and the secondary path estimation is possible in order to take into account the physical delay in electro acoustic hardware control loudspeaker and power amplifier. Firstly, we have measured interior noise of KTX and estimated noise path in KTX test-bed. However there was some problem related to algorithm divergence and increasing the filter order. We have made a simulation of interior environment of KTX car by using three frequency bands of 120Hz, 280Hz, 360Hz as the most important for KTX ANC system. During this research the interior noise reduction of KTX car was made by using the multi-channel FXLMS algorithm. Reduction performance was evaluated and compared each other for open space section and tunnel section. in-situ experiment for the KTX noise reduction by proposed ANC was performed based on data obtained in simulation and they were compared for open space section and tunnel section as well.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.22 no.10
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• pp.985-993
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• 2012

In this paper, a new method is proposed to estimate the sound pressure generated from gasoline direct injection (GDI) engine. There are many noise sources as much as components in GDI engine. Among these components, fuel pump, fuel injector, fuel rail, pressure pump and intake/exhaust manifolds are major components generated from top of the engine. In order to estimate the contribution of these components to engine noise, the total sound pressure at the front of the engine is estimated by using airborne source quantification (ASQ) method. Airborne source quantification method requires the acoustic source volume velocity of each component. The volume velocity has been calculated by using the inverse method. The inverse method requires many tests and has ill-condition problem. This paper suggested a method to obtain volume velocity directly based on the direct measurement of sound intensity and particle velocity. The method is validated by using two known monopole sources installed at the anechoic chamber. Finally the proposed method is applied to the identification and contribution of noise sources caused by the GDI components of the test engine.