• 제목/요약/키워드: Sound Speaker

검색결과 266건 처리시간 0.025초

소음계 교정 자동화 시스템 개발 및 성능평가 (Development and Performance of Automated Calibration System of Sound Level Meters)

  • 김용태;조문재;이용봉;서재갑
    • 소음진동
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    • 제8권5호
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    • pp.879-886
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    • 1998
  • An automated calibration system of sound level meters was developed and tested. As a standard sound source, the speaker unit(Forstex FE208) cabineted by 440$\times$390$\times$490 $\textrm{mm}^3$(LHW) volume wood box was adopted. Including this source, the driving part was found out to have a good linearity of sound pressure output vs AC voltage input. The Hybrid-Bisect/Newton-Raphson method modified by the linearity was adopted as a searching algorithm. Uisng GPIB interface, the console PC make the control, measurements, and calculations and finally make the accumulation of useful data and results automatically by the instructon in the program coded by C languate. Several trials of automatic calibration using this developed system give the reliable results.

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사운드스케이프 적용을 위한 옥외 P.A. 시스템의 적정 인지레벨에 관한 실험적 연구 (An Experimental Study on the Optimistic Recognition Level of Public Address System as a Soundscape Application Facility)

  • 송민정;장길수
    • 한국소음진동공학회논문집
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    • 제17권11호
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    • pp.1050-1055
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    • 2007
  • P.A.(public address) system is considered as an useful active soundscape appliance which can gives a place identity and vitality by introducing conventional musics, environmental musics, bird singing sounds etc. In this study, the main aim is to know the optimistic distance from the speaker and sound pressure level range of introducing sound. So, the sound pressure level of P.A. system due to distances were measured and subjects' responses with level variations were checked. The main results are as follows. Level range from 64 dB to 71 dB is comfortable for subjects. And the optimal level of introducing sound is related with sound source characteristics. The results of this study could be used for street furniture location design and P.A. system output level.

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

  • 김호수;이명섭;구경완;한상옥
    • 조명전기설비학회논문지
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    • 제18권3호
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    • pp.128-134
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    • 2004
  • 본 연구에서는 전력선통신 기술을 이용하여 전송주파수에 음성신호를 변조한 후 전력선으로 전송하고 전력선으로부터 수신 및 복조하여 변조된 음성신호를 검출하는 음향전달 시스템을 제작하였다. 제작된 시스템으로부터 변복조 및 주파수 특성을 분석하여, 어학실습 교육을 위한 용도로서의 활용 가능성을 평가하였다. 그 결과 전달된 음향 신호가 최적상태는 아니지만 어학실습 교육용으로 청취하기에 충분한 수준으로 판단되었다.

VR 환경을 위한 실시간 음장 재현 시스템 구현 (Real-Time 3D Sound Rendering System Implementation for Virtual Reality Environment)

  • 채수복;방승범;황신;고희동;김순협
    • 한국감성과학회:학술대회논문집
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    • 한국감성과학회 2000년도 추계학술대회 논문집
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    • pp.222-227
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    • 2000
  • 본 논문은 VR시스템에서 실시간으로 3D Sound를 Rendering하기 위한 음장재현 시스템구현에 관한 것이다. 2개의 Speaker 또는 헤드폰을 사용하여 음장을 제어할 수 있다. 음장 제어는 레이 트레이싱(Ray Tracing)기법을 이용하여 음장을 시뮬레이션하고 가상 공간의 음장 파라미터를 추출하여 음원에 적용하면서 실시간으로 음장효과를 렌더링한다. 이 시스템은 펜티엄-II 333MHz, 128M RAM, SoundBlaster SoundCard를 사용한 시스템에서 구현하였다. 최종적으로 청취자는 2개의 스피커 또는 헤드폰을 이용하여 3D 음장을 경험하게 된다.

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Voice Command-based Prediction and Follow of Human Path of Mobile Robots in AI Space

  • Tae-Seok Jin
    • 한국산업융합학회 논문집
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    • 제26권2_1호
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    • pp.225-230
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    • 2023
  • This research addresses sound command based human tracking problems for autonomous cleaning mobile robot in a networked AI space. To solve the problem, the difference among the traveling times of the sound command to each of three microphones has been used to calculate the distance and orientation of the sound from the cleaning mobile robot, which carries the microphone array. The cross-correlation between two signals has been applied for detecting the time difference between two signals, which provides reliable and precise value of the time difference compared to the conventional methods. To generate the tracking direction to the sound command, fuzzy rules are applied and the results are used to control the cleaning mobile robot in a real-time. Finally the experiment results show that the proposed algorithm works well, even though the mobile robot knows little about the environment.

Japanese Vowel Sound Classification Using Fuzzy Inference System

  • Phitakwinai, Suwannee;Sawada, Hideyuki;Auephanwiriyakul, Sansanee;Theera-Umpon, Nipon
    • 한국융합학회논문지
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    • 제5권1호
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    • pp.35-41
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    • 2014
  • An automatic speech recognition system is one of the popular research problems. There are many research groups working in this field for different language including Japanese. Japanese vowel recognition is one of important parts in the Japanese speech recognition system. The vowel classification system with the Mamdani fuzzy inference system was developed in this research. We tested our system on the blind test data set collected from one male native Japanese speaker and four male non-native Japanese speakers. All subjects in the blind test data set were not the same subjects in the training data set. We found out that the classification rate from the training data set is 95.0 %. In the speaker-independent experiments, the classification rate from the native speaker is around 70.0 %, whereas that from the non-native speakers is around 80.5 %.

Acoustic Performance Enhancement in PVDF Speakers by Using Buckled Nanospring Carbon Nanotubes

  • Ham, Sora;Lee, Yun Jae;Kim, Jung-Hyuk;Kim, Sung-Ryong;Choi, Won Kook
    • 센서학회지
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    • 제28권6호
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    • pp.360-365
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    • 2019
  • A polyvinylidene fluoride (PVDF)-based film speaker is successfully fabricated with enhanced bass sound by incorporating buckled nanospring carbon nanotubes (NS-CNTs) as fillers. Various concentrations up to 1-7 wt% of uniformly dispersed buckled NS-CNTs are loaded to increase the beta (β)-phase fraction, crystallinity, and dielectric constant of the speaker, and this results in the bass part enhancement of about 19 dB full scale (dBFS) at 7 wt% filler loading of the piezoelectric film speaker.

한국인의 영어피치악센트 발음에 관한 연구 (An Acoustic Study of the Pronunciation of English Pitch. Accents Uttered by Korean Speakers)

  • 구희산
    • 음성과학
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    • 제10권2호
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    • pp.223-236
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    • 2003
  • The purpose of this experimental study is to investigate characteristics of English pitch accents uttered by Korean speakers. Six English sentences were uttered five times by fifteen male undergraduate and graduate students from three groups, Seoul, Yongnam and Honam dialect speakers. We compared the subjects' data with the data of a native speaker of English as model pronunciation of English pitch accents. Acoustic features(Fo, duration, amplitude) were measured from sound spectrograms made by the PC Works. Results showed that (1) acoustic features of English pitch accents are Fo and duration for the native speaker and Korean speakers altogether, (2) Seoul dialect speakers uttered English pitch accents more similarly to the English native speaker than the other dialect speakers and (3) Korean speakers generally have difficulties in pronouncing L* accents. It appears that Korean speakers have more problems in pronouncing L* accents than H* accents.

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Perceptual Experiment on Number Production for Speaker Identification

  • Yang, Byung-Gon
    • 음성과학
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    • 제8권1호
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    • pp.7-19
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    • 2001
  • The acoustic parameters of nine Korean numbers were analyzed by Praat, a speech analysis software, and synthesized by SenSynPPC, a Klatt formant synthesizer. The overall intensity, pitch and formant values of the numbers were modified dynamically by a step of 1 dB, 1 Hz and 2.5% respectively. The study explored the sensitivity of listeners to changes in the three acoustic parameters. Twelve subjects (male and female) listened to 390 pairs of synthesized numbers and judged whether the given pair sounded the same or different. Results showed that subjects perceived the same sound quality within the range of 6.6 dB of intensity variation, 10.5 Hz of pitch variation and 5.9% of the first three formant variations. The male and female groups showed almost the same perceptual ranges. Also, an asymmetrical structure of high and low boundary was observed. The ranges may be applicable to the development of a speaker identification system while the method of synthesis modification may apply to its evaluation data.

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포르만트 주파수를 이용한 한국어 음성의 자동인식에 관한 연구

  • 김순협;박규태
    • 한국통신학회:학술대회논문집
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    • 한국통신학회 1983년도 춘계학술발표회논문집
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    • pp.16-17
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    • 1983
  • In Speech signal processing, ARMA spectral estimation method is used. It has been demonstrated that the ARMA model provides better spectral estimation then the more specialized AR model and MA model. Dynamic program is used to achieve time algnment. Speech sound similarity is defined to be proportional to the distance seperating to sound in a vector space defined by ARMA model. AS a result, the recognition rate of 97.3% for three speaker is obtained.

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