• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2005.11a
• /
• pp.400-405
• /
• 2005

The diagnosis of mechanical load and of power transmission system failures is usually carried out through mechanical signals such as vibration signals, acoustic emissions, motor speed envelope. If the mechanical load comes from an electrical machine the mechanical failures could be detected previously. Mechanical rotor imbalances and rotor eccentricities are reflected in electric, electromagnetic and mechanical quantities. Therefore, many surveillance schemes apply to the Fourier spectrum of a line current in order to monitor the motor condition. Due to the interaction of the currents and voltages, both these current harmonics are also reflected by a single harmonic component in the frequency spectrum of the electric power. Motor Current Signature Analysis is the usuful technique to assess machine electrical condition.

• Journal of the Korean Society of Fisheries and Ocean Technology
• /
• v.16 no.1
• /
• pp.1-15
• /
• 1980

An underwater recording system was designed to measure the sound spectra of the underwater noises produced by fishing gears in operation. Recorded were noi~es from three types of fishing gears: an anchovy set net, three anchovy boat seine net and a stern trawlnet. Acoustic analysis were made using a heterodyne analyzer, a digital frequency analyzer and a level recorder. The no;'e produced by the anchovy set net was found in the high frequency region of the onset of ambient noise spectrum with a slope of - 6 dB/octave. Here the ambient noise spectrum is higher, though similar in shape, than Knudsen spectrum, and is attributed to the breaking action of the coastal wave. Measured noise spectra during the fishing operations of the anchovy boat seine nets are attributed to the background noise of the sea in the presence of the fishing vessels. The frequency distribution of the noise was 20~5, 000 Hz in the case of two steel anchovy boat seiners, and 20-3,000 Hz in the case of the wooden anchovy boat seiner. The predominant frequency range was 250~350 Hz and maximum sound pressure level was 122 dB (re $1\muPa$) in the case of the steel boat and ] 17 dB in the case of the wooden boat. The noises produced by the trawl fishing gears are remarkably higher than the background noi~e in the presence of the fishing vessel. The frequency distribution of the noi~e was 20-6,300 Hz. The predominant frequency range was 100~200 Hz and maximum sound pressure level was 137 dB ( re $1\muPa$) . The noise spectra were not so much different from that caused by vibrations of the towing cable and the structure of the ground rope of the trawl net towed in an experimental tank.

• Proceedings of the KIPE Conference
• /
• 2001.10a
• /
• pp.160-168
• /
• 2001

The RPWM (Random· Pulse Width Modulation) is a switching technique to spread the voltage and current harmonics over a wide frequency area. By using randomly changing switching frequency of the inverter, the power spectrum of the electromagnetic acoustic noise can be spread to the wide-band area. The wide­band noise is much more comfortable and less annoying than the narrow-band one. So, the RPWM has been attracting interest as an excellent method for the reduction of acoustic noise on the inverter drive system. In this paper a new RPPWM (Random Position Space Vector PWM) is proposed and implemented. Each of three phase pulses is located randomly in each switching interval. Along with the randomization of PWM pulses, the space vector modulation is also executed in the C167 micro-controller. The experimental results show that the voltage and current harmonics are spread to a wide band area and that the audible switching noise is reduced by the proposed RPPWM method.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
• /
• v.19 no.3
• /
• pp.17-26
• /
• 2005

The random pulse width modulation(RPWM) in inverter-fed induction motor drive is presented. This paper describes a RPWM method based on space vector PWM strategy for shaping the switching noise spectrum in such a way it can merge with the natural system noise. To verify the validity of the proposed RPWM scheme, the experiment based on the DSP56F803 microprocessor was executed finally, the simulation and experimental results will be given to demonstrate the effectiveness of the unposed RPWM scheme.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.6
• /
• pp.367-372
• /
• 2012

This paper introduces noise masking algorithm for 64 channel digital hearing aid. 125 Hz spectral resolution is maintained for 64 channels from 125 Hz to 8000 Hz. The same spectral masking processing effects as the cochlea are considered and applied for the present hearing aid noise reduction processing algorithm. Theoretical algorithm has been ported into assembler language program software and been embedded into a DSP IC chip for the digital hearing aid. Some parts of noise masking software program code are explained, and the results of the real-time noise reduction are verified by electro-acoustic measurements.

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.15 no.12 s.105
• /
• pp.1378-1388
• /
• 2005

Spatial control of sound is essential to deliver better sound to the listener's position in space. As it can be experienced in many listening environments. the quality of sound can not be manifested over every Position in a hall. This motivates us to control sound in a region we select. The primary focus of the developed method has to do with the brightness and contrast of acoustic image in space. In particular, the acoustic brightness control seeks a way to increase loudness of sound over a chosen area, and the contrast control aims to enhance loudness difference between two neighboring regions. This enables us to make two different kinds of zone - the zone of quiet and the zone of loud sound - at the same time. The other perspective of this study is on the direction of sound. It is shown that we can control the direction of perceived sound source by focusing acoustic energy in wavenumber domain. To begin with, the proposed approaches are formulated for pure-tone case. Then the control methods are extended to a more general case, where the excitation signal has broadband spectrum. In order to control the broadband signal in time domain, an inverse filter design problem is defined and solved in frequency domain. Numerical and experimental results obtained in various conditions certainly validate that the acoustic brightness, acoustic contrast, direction of wave front can be manipulated for some finite region in space and time.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2009.04a
• /
• pp.483-489
• /
• 2009

The Time Difference of Arrival (TDOA) algorithm is being used widely for identifying the location of a source emanating either electrical or acoustic signal. It's application areas will not be limited to identifying the source at a fixed location, for example the origin of an earthquake, but will also include the trajectory monitoring for a moving source equipped with a GPS sensor. Most of the TDOA algorithm uses time correlation technique to find the time delay between received signals, and therefore difficult to be used for identifying the location of multiple sources. In this paper a TDOA algorithm based on cross-spectrum is developed to find the trajectory of two sound sources with different frequencies. Although its application is limited to for the sources on a disk plane, but it can be applied for identifying the locations of more than two sources simultaneously.

• The Journal of the Acoustical Society of Korea
• /
• v.35 no.5
• /
• pp.368-374
• /
• 2016

This paper presents constructing transmitter and receiver by using a direct sequence spread spectrum techniques to DPSK (Differential Phase-Shift Keying) scheme in underwater acoustic communication. Since DPSK signal can be demodulated if the receiver knows only the phase difference between the adjacent bits, DPSK receiver structure has the advantage of being simplified. In the conventional receiver, two adjacent symbols of transmitted signal before despread are passed to the transition correlator that detects data by comparing maximum correlation outputs. At this time, the error for maximum value of the correlator output may increase because of low SNR (Signal to Noise Ratio) or high Doppler shift frequency according to the underwater channel. In this paper, we propose a method for accurate detection result using the width as well as the magnitude among outputs produced by the correlator. The performances of the proposed method was evaluated by simulation and lake trial data.

• Journal of Biomedical Engineering Research
• /
• v.13 no.3
• /
• pp.195-200
• /
• 1992

This paper deal with the acoustic cues of Pyoung-An Do dialect using linear prediction. Also, this paper descrbes a statistical comparison between standard tone speech data and Pyoung-An Do dia lects. The analysis done mainly focused on the distribution of formants and pitch periods accord to ac- cents variation. For the purpose of objective comparison, the experiments are performed by extracts for- mant LPC spectrum and pithch periods from average magnitude difference function waveforms. Summing up the results, In disyllable words (VCV pattern) , prepositioned vowels have longer phona lion time than postpositioned vowels and the intrin, iii phonation time is whore longer in the low vowels than in the high ones. The africative consonants show the mixed characteristics of the plosive and frlc ative consonants. The remarkable acoustic cues are the low frequency noise-like waves just before the 1st formants in the plosive consonants, the high frequency noise-like waves in the fricative consonants, and phonation time is not affected by the kinds of prepositioned or postpositioned vowels.

• The Journal of the Acoustical Society of Korea
• /
• v.31 no.3
• /
• pp.179-187
• /
• 2012

This paper presents a novel method for estimating reliable noise spectral magnitude for acoustic background noise suppression where only a single microphone recording is available. The proposed method finds noise estimates from spectral magnitudes measured at line spectral frequencies (LSFs), under the observation that adjacent LSFs are near the peak frequencies and isolated LSFs are close to the relatively flattened valleys of LPC spectra. The parameters used in the proposed method are LPC coefficients, their corresponding LSFs, and the gain of LPC residual signals, so it suits well to LPC-based speech coders.