• Title/Summary/Keyword: Sound Signal

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Simple Estimation of Sound Source Directivity in Diffused Acoustic Field: Numerical Simulation (확산음향장에서의 음원 지향성 간이추정: 수치시뮬레이션)

  • Kim, Kookhyun
    • Journal of Ocean Engineering and Technology
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    • v.33 no.5
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    • pp.421-426
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    • 2019
  • The directivity of an underwater sound source should be measured in an acoustically open field such as a calm sea or lake, or an anechoic water tank facility. However, technical difficulties arise when practically implementing this in open fields. Signal processing-based techniques such as a sound intensity method and near-field acoustic holography have been adopted to overcome the problem, but these are inefficient in terms of acquisition and maintenance costs. This study established a simple directivity estimation technique with data acquisition, filtering, and analysis tools. A numerical simulation based on an acoustic radiosity method showed that the technique is practicable for sound source directivity estimation in a diffused reverberant acoustic field like a reverberant water tank.

Condition Monitoring of Tool Wear and Breakage using Sound Pressure in Turning Processes (선삭공정에서 음압을 이용한 공구마멸 파손의 상태감시)

  • 이성일
    • Journal of the Korean Society of Manufacturing Technology Engineers
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    • v.6 no.3
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    • pp.36-43
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    • 1997
  • In order to make unmanned machining systems with satisfactory performances, it is necessary to incorporate appropriate condition monitoring systems in the machining workstations to provide the required intelligence of the expert. This paper deals with condition monitoring for tool wear and breakage during turning operation. Developing economic sensing and identification methods for turning processes, sound pressure measurement and digital signal processing technique are proposed. The validity of the proposed system is confirmed through the large number of cutting tests.

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Reinforcing Stethoscope Sound using Spectral Shift (스펙트럼 이동을 이용한 청진음 강화)

  • Jung, Dong Keun
    • Journal of Sensor Science and Technology
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    • v.30 no.1
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    • pp.47-50
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    • 2021
  • Human hearing sensitivity is frequency-dependent. The sensitivity is low at both ends of the audible frequency, and the sensitivity is the highest in the middle band at 3000 Hz. The heart sound of a healthy person is concentrated at a low frequency of 200 Hz or less, and despite using a stethoscope, the hearing sensitivity of the human body is low, and the stethoscope sound is low. Amplifying the sound of the stethoscope is not effective in distinguishing heart sounds in noisy environments because it maintains the same signal-to-noise ratio. In this study, a method of enhancing auditory stimulation was developed by applying a method of moving the spectrum of auscultation sounds into a high-frequency region where the human body is highly sensitive to hearing. The spectrum of the auscultation sound was moved up by 500 Hz in the frequency domain, and an inverse fast Fourier transform (FFT) was performed to reconstruct the auscultation sound. The heart sounds reconstructed by moving the spectra were divided into the first heart and second heart sound components, as in the original heart sound, and it was confirmed that the intensity was large in the cochleagram representing auditory stimulation. Therefore, this study suggested that spectral shift is a method to enhance auditory stimulation during auscultation without increasing the intensity of the auscultation sound.

Sound Synthesis of Gayageum by Impulse Responses of Body and Anjok (안족과 몸통의 임펄스 응답을 이용한 가야금 사운드 합성)

  • Cho Sang-Jin;Choi Gin-Kyu;Chong Ui-Pil
    • Journal of the Institute of Convergence Signal Processing
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    • v.7 no.3
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    • pp.102-107
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    • 2006
  • In this paper, we propose a method of a sound synthesis of Korean plucked string instrument, gayageum, by physical modeling which use impulse responses of body and Anjok. Gayageum consists of three kinds of systems: string, body, and Anjok. These are a serial combination of linear time invariant systems. String can be modeled by digital delay line. Body and Anjok can be estimated by their impulse responses. We found three resonance frequencies in the body impulse response, and implemented resonator as body. Anjok was implemented as high pass filter in fundamental frequency band of gayageum. RMSEs of synthesized sounds are distributed from 0.01 to 0.03. It was difficult to distinguish the resulting synthesized sounds from the originals sound by ear.

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Ultrasonic Bone Densitometer by Measuring the Speed of Sound (SOS를 이용한 초음파 골밀도 측정 장치)

  • Jung, Min-Sang;Hahn, Eun-Joo;Kim, Yeong-Sik
    • Progress in Medical Physics
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    • v.18 no.3
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    • pp.172-178
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    • 2007
  • An ultrasonic bone densitometer has been developed by measuring speed of sound signal transmitted and received on the skin, not through the horizontal axis but through the vertical one in tissue. The SOS(speed of sound) method measuring the time difference between the ultrasound signals reflected from the both sides of surface of bone could produce more precise result compared with the BUA(broadband ultrasound attenuation) method measuring the frequency difference. Middle finger is selected to be the best measurement position in order to increase the accuracy, after due consideration that the thickness of flesh at the down part of thumb shows too much variation although the ratio of the receiving signal is higher than the other fingers. The measured value by using SOS method shows almost the same result as compared with the conventional DEXA method.

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Localization of Rotating Sound Sources Using Beamforming Method (빔형성방법을 이용한 회전하는 음원의 위치 판별에 관한 연구)

  • Lee Jaehyung;Hong Suk-Ho;Choi Jong-Soo
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.14 no.12
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    • pp.1338-1346
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    • 2004
  • The positions of rotating sound sources have been localized by experiments with the Doppler effects removed. In order to de-Dopplerize the sound signals emitted from moving sources, two kinds of signal reconstruction methods were applied. One is the forward propagation method and the other is the backward propagation method. Forward propagation method analyze the source emission time based on the instantaneous distance between sensors and the assumed source position, then the signals are reconstructed with respect to the emission time. On the other hand, the backward method uses time delay to do-Dopplerize the acquired data for the received time of reference. In both techniques. the reconstructed signal data were processed using beamforming algorithm to produce power distributions at the frequencies of interest. Experiments have been carried out for varying frequencies, rotating speeds and the object distances. It is shown that the forward propagation method gives better performance in locating source position than the backward propagation method.

System Realization of Whale Sound Reconstruction (고래 사운드 재생 시스템 구현)

  • Chong, Ui-Pil;Jeon, Seo-Yun;Hong, Jeong-Pil
    • Journal of the Institute of Convergence Signal Processing
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    • v.20 no.3
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    • pp.145-150
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    • 2019
  • We develop the system realization of whale sound reconstruction by inverse MFCC algorithm with the weighted L2-norm minimization techniques. The output products from this research will contribute to the whale tourism and multimedia content industry by combining whale sound contents with the prototype of 3D printing. First of all, we develop the softwares for generating whale sounds and install them into Raspberry Pi hardware and fasten them inside a 3D printed whale. The languages used in the development of this system are the C++ for whale-sounding classification, MATLAB and Python for whale-sounding playback algorithm, and Rhino 6 for 3D printing.

Energy-Efficient Approximate Speech Signal Processing for Wearable Devices

  • Park, Taejoon;Shin, Kyoosik;Kim, Nam Sung
    • ETRI Journal
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    • v.39 no.2
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    • pp.145-150
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    • 2017
  • As wearable devices are powered by batteries, they need to consume as little energy as possible. To address this challenge, in this article, we propose a synergistic technique for energy-efficient approximate speech signal processing (ASSP) for wearable devices. More specifically, to enable the efficient trade-off between energy consumption and sound quality, we synergistically integrate an approximate multiplier and a successive approximate register analog-to-digital converter using our enhanced conversion algorithm. The proposed ASSP technique provides ~40% lower energy consumption with ~5% higher sound quality than a traditional one that optimizes only the bit width of SSP.

Source Identification of Non-Stationary Sound.Vibration Signals Using Multi-Dimensional Spectral Analysis Method (다차원 스펙트럼 해석법을 이용한 비정상 소음.진동 신호의 소음원 규명)

  • Sim, Hyoun-Jin;Lee, Hae-Jin;Lee, You-Yub;Lee, Jung-Youn;Oh, Jae-Eung
    • Transactions of the Korean Society of Mechanical Engineers A
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    • v.30 no.9 s.252
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    • pp.1154-1159
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    • 2006
  • In this paper, time-frequency analysis and multi-dimensional spectral analysis methods are applied to source identification and diagnostic of non-stationary sound vibration signals. By checking the coherences for concerned time, this simulation is very well coincident to expected results. The proposed method analyzes the signal instantaneously in both time and frequency domains. The MDSA (Multiple Dimensional Spectral Analysis) analyzes the signal in the plane of instantaneous time and instantaneous frequency at the same time. And it was verified by using the 1500cc passenger car which is accelerated from 70Hz to 95Hz in 4 seconds, the proposed method is effective in determining the vehicle diagnostic problems.

An Electronic Auscultation System Design using a Polymer Based Adherent Differential Output Sensor (Polymer based adherent differential output sensor를 이용한 전자 청진 시스템 설계)

  • 한철규;고성택;최민주
    • Journal of the Institute of Convergence Signal Processing
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    • v.2 no.1
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    • pp.108-112
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    • 2001
  • Heart sound contains rich information regarding the dynamics of the heart and the auscultation has been a first choice of routine procedures for diagnosis of the heart. However, heart sounds captured using a conventional stethoscope are not often loud or clear enough for doctors to precisely classify their characteristics, especially, under the noisy environments of the hospital. A simple auscultation device that removed shortcomings of the conventional stethoscope was constructed in the study. The device employed a polymer based adherent differential output sensor which was on contact with skin through a coupling medium and appropriated electronic circuits for signal amplification and conditioning An ordinary headphone is taken to hear the captured heart sounds and the volume can be adjusted to hear well. It is also possible that the device sends the captured heart sound signals to a PC where the signals are further processed and viualized.

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