• Title/Summary/Keyword: sound spectrum

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Sound Monitoring System of Machining using the Statistical Features of Frequency Domain and Artificial Neural Network (주파수 영역의 통계적 특징과 인공신경망을 이용한 기계가공의 사운드 모니터링 시스템)

  • Lee, Kyeong-Min;Vununu, Caleb;Lee, Suk-Hwan;Kwon, Ki-Ryong
    • Journal of Korea Multimedia Society
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    • v.21 no.8
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    • pp.837-848
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    • 2018
  • Monitoring technology of machining has a long history since unmanned machining was introduced. Despite the long history, many researchers have presented new approaches continuously in this area. Sound based machine fault diagnosis is the process consisting of detecting automatically the damages that affect the machines by analyzing the sounds they produce during their operating time. The collected sound is corrupted by the surrounding work environment. Therefore, the most important part of the diagnosis is to find hidden elements inside the data that can represent the error pattern. This paper presents a feature extraction methodology that combines various digital signal processing and pattern recognition methods for the analysis of the sounds produced by tools. The magnitude spectrum of the sound is extracted using the Fourier analysis and the band-pass filter is applied to further characterize the data. Statistical functions are also used as input to the nonlinear classifier for the final response. The results prove that the proposed feature extraction method accurately captures the hidden patterns of the sound generated by the tool, unlike the conventional features. Therefore, it is shown that the proposed method can be applied to a sound based automatic diagnosis system.

A Study on the Robust Sound Localization System Using Subband Filter Bank (서브밴드 필터 뱅크를 이용한 강인한 음원 추적시스템에 대한 연구)

  • 박규식;박재현;온승엽;오상헌
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.1
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    • pp.36-42
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    • 2001
  • This paper propose new sound localization algorithm that detects the sound source bearing in a closed office environment using two microphone array. The proposed Subband CPSP (Cross Power Spectrum Phase) algorithm is a development of previously Down CPSP method using subband approach. It first split the received microphone signals into subbands and then calculates subband CPSP which result in possible source bearings. This type of algorithm, Subband CPSP, can provide more robust and reliable sound localization system because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, a real time simulation was conducted and it was compared with previous CPSP method. From the simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than 5% average accuracy for sound source detection.

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Spatial Manipulation of Sound using Multiple Sources (다수의 음원을 사용한 공간의 소리 제어 방법론)

  • Choi, Joung-Woo;Kim, Yang-Hann;Park, Young-Jin
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2005.11a
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    • pp.620-628
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    • 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.

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Ring-shaped Sound Focusing using Wavenumber Domain Matching (파수영역매칭을 통한 링 형상의 음향집적공간 형성)

  • Park, Jin-Young;Kim, Yang-Hann
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2006.11a
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    • pp.506-509
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    • 2006
  • Shaped Sound Focusing is defined as the generation of acoustically bright shape in space using multiple sources. The acoustically bright shape is a spatially focused region with relatively high acoustic potential energy level. In view of the energy transfer, acoustical focusing is essential because acoustic energy is very small to use other type of energy. Practically, focused sound shape control not a point is meaningful because there are so many needs to enlarge the focal region especially in clinical uses and others. If focused sound shape can be controlled, it offers various kinds of solutions for clinical uses and others because a regional focusing is essentially needed to reduce a treatment time and enhance the performance of transducers. For making the shaped-sound field, control variables, such as a number of sources, excitation frequency, source positioning, etc., should be taken according to geometrical sound shape. To verify these relations between them, wavenumber domain matching method is suggested because wavenumber spectrum can provide the information of control variables of sources. In this paper, the procedures of shaped sound focusing using wavenumber domain matching and relations between control variables and geometrical sound shape are covered in case of an acoustical ring.

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Physical Analysis of the Prosthetic Valve Sound (기계판막치환후의 판막음에 대한 연구)

  • 조범구
    • Journal of Chest Surgery
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    • v.22 no.4
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    • pp.589-593
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    • 1989
  • The frequency spectrum of the metallic closing sound and its loudness were measured by non invasive techniques in 66 patients. They had examined a total of 7 Carbomedics valve, 10 Duromedics valve, 11 St. Jude heart valve in mitral position and 8 Carbomedics, 10 Duromedics, 20 St. Jude heart valve prostheses functioning normally in aortic position. Statistical comparison of the loudness from sound produced by the three valves in each position, the following; The Carbomedics valve has the lowest average loudness, followed by the St. Jude medical valve, and finally the Edward Duromedics valve. And we analysis the changing factor of the loudness of valve sound, only the velocity of the flow through the valve influenced to the valve sound.

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Development of Parameter Extraction Algorithm and Software Simulator For a Digital Music FM Synthesis (FM 방식의 디지털 악기음 합성을 위한 소프트웨어 시뮬레이터 및 파라미터 추출 알고리즘 개발)

  • Joon Yul Joo
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.31B no.3
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    • pp.24-38
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    • 1994
  • In this paper we develop the software simulator written in a C language for a frequency modulation synthesis and the approximate range of parameters, for a musically satisfactory timbre, obtained by using the software simulator will be applied to develop an algorithm for parameter extraction. For a frequency modulation synthesis, we also develop an algorithm for parameter extraction through waveform analysis in the time domain as well as spectrum analysis using a FFT in the frequency domain. To verify the validity of the developed algorithm as well as software simulator experimentally, we extract parameters for the several music instruments using the suggested algorithm and analyze the synthesized sound by applying the parameters to the software simulator. The evaluation of the synthesized sound is first done by listening the sound directly as a subjective testing. Secondly, to evaluate the synthesized sound objectively with an engineering sense, we compare the synthesized sound with an original one in a time domain and a frequency domain.

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Development of Sound Transmitting System using Power Line (전력선을 이용한 음성전달 시스템 개발)

  • Park, Yong-Jun;Kang, Min-Woo;Koo, Kyung-Wan;Lee, Myung-Sup;Han, Sang-Ok
    • Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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    • 2002.07b
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    • pp.1075-1079
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    • 2002
  • This paper presents a result of sound transmitting system with power line communication technique, Sound transmitting system is a transmitter which transmits modulated audio signal to power to power line and receiver which is capable of detesting it with earphone or speaker. It has been evaluated with the frequency characteristics and spectrum analysis. And, from the result of evaluation on the developed system, we confirm the superior sound transmitting characteristics, and the possibility of application on a language laboratory.

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Radiated Sound from Compliant and Viscoelastic Plates in a Turbulent Boundary Layer (난류 경계층에서 컴플라이언트 코팅과 점탄성 벽면의 방사 소음에 관한 실험적 연구)

  • Lee Seungbae;Lee Chang-Jun;Kwon O-Sup;Jeon Woo-Pyung
    • Proceedings of the KSME Conference
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    • 2002.08a
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    • pp.779-782
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    • 2002
  • We examine the problem in which porous/viscoelastic compliant thin plates are subject to pressure fluctuations under transitional or turbulent boundary layer. Measurements are presented of the frequency spectra of the near-field pressure and radiated sound by compliant surface. A porous plate consisting of 5mm thick, open-cell foam with fabric covering and a viscoelastic painted plate of 1mm thick over an acoustic board of 4m thick were placed over a rigid surface in an anechoic wind tunnel. Streamwise velocity and wall pressure measurements were shown to highly attenuate the convective wall pressure energy when the convective wavenumber ($k_{ch}$) was 3.0 or more. The sound source localization on the compliant walls is applied to the measurement of radiated sound by using an acoustic mirror system.

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Realization of Digital Music Synthesizer Using a Frequency Modulation (FM 방식을 이용한 디지탈 악기음 합성기의 구현)

  • 주세철;김진범;김기두
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.32B no.7
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    • pp.1025-1035
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    • 1995
  • In this paper, we realize a real time digital FM synthesizer based on genetic algorithm using a general purpose digital signal processor. Especially, we synthesize diverse music sounds nicely using a synthesis model consisting of a single modulator and multiple carriers. Also we present genetic algorithm-based technique which determines optimal parameters for reconstruction through FM synthesis of a sound after analyzing the spectrum of PCM data as a standard music sound using FFT. Using the suggested parameter extractiuon algorithm, we extract parameters of several instruments and then synthesize digital FM sounds. To verify the validity of the parameter extraction algorithm as well as realization of a real time digital music synthesizer, the evaluation is first done by listening the sound directly as subjective test. Secondly, to evaluate the synthesized sound objectively with an engineering sense, we compare the synthesized sound with an original one in a time domain and a frequency domain.

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Analysis of Sound Transmitting System using Power line Communication Technique (전력선을 이용한 음향전달 시스템의 구성 및 특성 분석)

  • Kim, Ho-Soo;Lee, Myung-Sub;Koo, Kyung-Wan;Han, Sang-Ok
    • Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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    • v.18 no.3
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    • pp.128-134
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    • 2004
  • This paper presents a result of sound transmitting system with power line communication technique. Sound transmitting system is a transmitter which transmits modulated audio signal to power to power line and receiver which is capable of detecting it with earphone or speaker. It has been evaluated with the frequency characteristics and spectrum analysis. And, from the result of evaluation on the developed system, we confirmed the superior sound transmitting characteristics, and the possibility of application on a language laboratory.