• Title/Summary/Keyword: Source speaker

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Design of the broadband and compact phase-calibrator for array microphones (어레이 마이크로폰용 광대역 소형 위상교정기의 설계)

  • Ju, Hyeong-Sick;Kim, Yang-Hann
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2004.11a
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    • pp.1032-1035
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    • 2004
  • Pressure distribution is measured by way microphones to identify noise sources in the space. For example, beam-forming method or acoustic holography use phase information to identify the source. Therefore, the phase is significant information to correctly identify the source position. However, due to the microphone characteristics and measuring systems, measured signals always have errors, which make the identification difficult. Therefore, phase calibration of microphones is needed. Duct and speaker systems are generally used as calibrators. Acoustic characteristics of the calibrator are, of course, functions of many Parameters of the system: i.e. duct size, frequency, and microphone spacing. In this paper, design parameters which effect on the performance and size of the calibrators are considered. Then the parameters would be applied to design and real product of the phase-calibrator.

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Automatic Directional-gain Control for Binaural Hearing Aids using Geomagnetic Sensors (지자기 센서를 이용한 양이 보청기의 방향성 이득 조절 연구)

  • Yang, Hyejin;An, Seonyoung;Jeong, Jaehyeon;Choi, Inyong;Woo, Jihwan
    • Journal of Biomedical Engineering Research
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    • v.37 no.6
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    • pp.209-214
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    • 2016
  • Binaural hearing aids with a voice transmitter have been widely used to enhance sound quality in noisy environment. However, this system has a limitation on sound-source localization. In this study, we investigated automatic directional-gain control method using geomagnetic sensors to provide directional information to binaural hearing aid user. The loudness gains of two hearing aids were differently controlled based on the directional information between a speaker position and a viewing direction of hearing aids user. This relative directional information was measured by two geomagnetic sensors on hearing aids user and a speaker. The results showed that the loudness gains were accurately controlled and could provide directional information based on the cue of interaural level differences.

A Study of Enemy Aptitude of Pistol Sound Source for Space Estimation (공간평가를 위한 피스톨음원의 적정성에 관한 연구)

  • Shon, Jang-Ryul;Kim, Jung-Joong
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.15 no.3 s.96
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    • pp.320-328
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    • 2005
  • Last target of architectural acoustics is that people wish to convey voice effectively from the space adaptively in use purpose in building. But, how exactly through space sound (sound source) that wish to deliver from indoor can be passed method to do quantification and evaluate quantity of sound by method to serve indoor architectural acoustics estimation summer period and methods to estimate definition propose. This Study searches special quality of sound source about MLS signal that is occurred short-answer sound source (pistol sound source) and nondirectional speaker among indoor sound estimation method, and measure and analyzed reverberation time (RT60), definition (C80, D50) by regulation of each ISO 3382 in age place (classroom, hall, gymnasium). Analysis result and sound factor among could know that d of two sound sources converges in measurement error extent about reverberation time (RT60) of analysis incidental and sound factors and value shows change irregularly about sound factor of D50, C80, pistol sound source judged there is problem. Also, could know that problem is happened in deflection except reverberation time is in deflection analysis with wave that measure each in fixed distance in branch. Finally, when differ size of sound source and measure about change of sound pressure level in case measure sound pressure level giving difference about 10 dB, sound factor could know that there is no different effect.

The Analysis and Implementation of Realistic Sound using Doppler Effect (도플러 효과를 이용한 실감 음향 분석 및 구현)

  • Yim, Yong-Min;Lim, Heung-Jun;Heo, Jun-Seok;Park, Jun-Young;Do, Yun-Hyung;Lee, Kangwhan
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2017.05a
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    • pp.523-526
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    • 2017
  • In modern recently technology, 3D-Audio is used to enhance immersion in Virtual Reality. This includes interest of people about VR and AR, which related to the field of computer graphics. In fact, a lot of research has been carried out in recent years into a 3D sound field. However, the existing 3D generator device used for virtual reality does not contain Doppler effect occurred by the sound comes to or leave from a listener, while an angle from the listener and the altitude of the source sound are applied. Therefore, this paper present 3D real sound utilizing Doppler effect with spatial-rotation-speaker. We map the source sound in 3D-space into a real space where a user stays and present 3D real sound by manipulating with rotation angle, phase difference, sound output volume of the sound in 3D-space, according to the location of a virtual source sound. Utilizing both natural Doppler effect of rotating sound that is occurring by spatial-rotation-speaker and artificial Doppler effect generated by frequency-modulation of sound quality could improving the virtual reality for sound condition for perspective listening.

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Real-Time DSP Implementation of Adaptive Multi-Rate with TMS320C542 board (TMS320C542보드를 이용한 Adaptive Multi-Rate 음성부호화기의 실시간 구현)

  • 박세익;전라온;이인성
    • Proceedings of the IEEK Conference
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    • 2000.09a
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    • pp.827-830
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    • 2000
  • 3GPP and ETSI adopted AMR(Adaptive Multi-Rate) as a standard for next generation IMT-2000 service. In this paper, we analyzed algorithm about AMR and optimized ANSI C source on the C complier and assembly language of Texas Instrument . The implemented AMR speech codec requires 28.2MIPS of complexity for encoder and 5.5MIPS for decoder. we performed real-time implementation of AMR speech codec using 82% of TMS320C5402 with 40 MIPS specification. We give proof that the output speech of the implemented speech codec on DSP board is identical with result of C source program simulation. Also the reconstructed speech is verified in the real-time environment consisted of microphone and speaker.

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A study on the noise reduction of practical duct system with the air cavity (공기층을 갖는 실제덕트 구조물에서의 소음저감에 관한 연구)

  • Kim, Chan-Mook;Lee, Doo-Ho;Bahng, Keuk-Ho
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2000.06a
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    • pp.1687-1692
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    • 2000
  • In this paper, experimental methods to find acoustic characteristics of acoustically treated air-conditioning duct system are proposed. Existing methods to analyze acoustic properties of duct with absorbent material have a dilemma which has to assume the wave in duct to be a plane wave. Under this assumption, applicable frequency limitation makes accurate analysis of practical air-conditioning system impossible. In order to analyze the properties of in-lined treated absorbent with high degree of accuracy, in this experiments the range of exciting frequency of sound source is broadband, which means that source speaker excites higher mode of in-duct sound field. Also, to define the relations of air cavity to the acoustic characteristics, acoustic experiments on ducts with air cavity of different depth are operated. In conclusion, air-cavity makes the absorbing ability of duct improved in low frequency range. Due to the interactions between the air cavity depth and the depth of absorbents, according to depth of cavity, the magnitude of absorption coefficients vs frequencies in specific range is changed. In lower frequency range, the absorption of sound energy by air cavity is more dominant than by absorbent itself, in higher range, the inversion is true.

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Storing and Broadcast System of Smart Multi Encoding Image (Smart 멀티 인코딩 영상 저장 및 방송 시스템)

  • Kim, Chang-Su;Kim, Jung-Woo;Jung, Hoe-Kyung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.17 no.7
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    • pp.1633-1638
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    • 2013
  • The mobile phone has now evolved into an effective multimedia devices to watch video content with your PC in addition to the calling features. Thus, the effectiveness of the video content streaming services smartphone will be available. And content should be able to deliver effectively. Be provided with textbook images and video of the speaker means that the effective content delivery. In this paper, we propose a integrated video management system that can be real-time VOD services on the Internet as input Multi-Source of audio-video, video content encoding system to meet the requirements of the above two.

String and Broadcast System of Smart Multi Encoding Umage (Smart 멀티 인코딩 영상 저장 및 방송 시스템)

  • Kim, Jeong-Woo;Kim, A-Yong;Ban, Tae-Hak;Jung, Hoe-Kyung
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2013.05a
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    • pp.830-832
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    • 2013
  • The mobile phone has now evolved into an effective multimedia devices to watch video content with your PC in addition to the calling features. Thus, the effectiveness of the video content streaming services smartphone will be available. And content should be able to deliver effectively. Be provided with textbook images and video of the speaker means that the effective content delivery. In this paper, we propose a integrated video management system that can be real-time VOD services on the Internet as input Multi-Source of audio-video, video content encoding system to meet the requirements of the above two.

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A Study on the Absorption Characteristics of Absorbents in Duct System with the Air Cavity (공기층을 갖는 공조덕트 구조물에서 흡음재의 흡음특성에 관한 연구)

  • 김찬묵;김도연;방극호
    • Journal of KSNVE
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    • v.10 no.5
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    • pp.892-897
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    • 2000
  • In this paper, experimental methods to find acoustic characteristics of acoustically treated air-conditioning duct system are proposed. Existing methods to analyze acoustic properties of duct with absorbent material have dilemma which has to assume the wave in duct to be a plane wave. Under this assumption. applicable frequency limitation makes accurate analysis of practical air-conditioning system impossible. In order to analyze the properties of in-lined treated absorbent with high degree of accuracy, in this experiments the range of exciting frequency of sound source is broadband, which means that source speaker excited higher mode of in-duct sound field. Also, to define the relations of air cavity to the acoustic characteristics, acoustic experiments on ducts with air cavity of different depth are operated. In conclusion, air-cavity makes the absorbing ability of duct improved in low frequency range. Due to the interactions between the air cavity depth and the depth of absorbents, according to depth of cavity, the magnitude of absorption coefficients vs frequencies in specific range is changed. In lower frequency range, the absorption of sound energy by air cavity is more dominant than by absorbent itself, in higher range, the inversion is true.

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Absolute sound level algorithm for contents platform (콘텐츠 플랫폼 적용을 위한 절대음량 알고리즘)

  • Gyeon, Du-Heon
    • The Journal of the Acoustical Society of Korea
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    • v.39 no.5
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    • pp.424-434
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    • 2020
  • This paper describes an algorithm that calculates Absolute Sound Level (ASL) for contents platform. ASL is a single volume representing individual sound sources and is a concept designed to integrate and utilize the sound level units in digital sound source and physical domain from a speaker in practical areas. For this concept to be used in content platforms and others, it is necessary to automatically derive the ASL without having to go through a hearing of mastering engineers. The key parameters of which a person recognizes the representative sound level of an individual single sound source are the areas of "frequency, maximum energy, energy variation coefficient, and perceived energy distribution," and the ASL was calculated through the normalizing of the weights.