• 제목/요약/키워드: Sound Processing

검색결과 614건 처리시간 0.024초

Hardware Implementation for Real-Time Speech Processing with Multiple Microphones

  • Seok, Cheong-Gyu;Choi, Jong-Suk;Kim, Mun-Sang;Park, Gwi-Tea
    • 제어로봇시스템학회:학술대회논문집
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    • 제어로봇시스템학회 2005년도 ICCAS
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    • pp.215-220
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    • 2005
  • Nowadays, various speech processing systems are being introduced in the fields of robotics. However, real-time processing and high performances are required to properly implement speech processing system for the autonomous robots. Achieving these goals requires advanced hardware techniques including intelligent software algorithms. For example, we need nonlinear amplifier boards which are able to adjust the compression radio (CR) via computer programming. And the necessity for noise reduction, double-buffering on EPLD (Erasable programmable logic device), simultaneous multi-channel AD conversion, distant sound localization will be explained in this paper. These ideas can be used to improve distant and omni-directional speech recognition. This speech processing system, based on embedded Linux system, is supposed to be mounted on the new home service robot, which is being developed at KIST (Korea Institute of Science and Technology)

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프린터 소음에 대한 감성소음 평가 시스템 개발 (Development of Sound Quality Evaluation System for a Printer Noise Based on Human Sensibility)

  • 박상원;이현호;나은우;이상권;박영재;김종우
    • 한국소음진동공학회논문집
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    • 제20권5호
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    • pp.427-436
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    • 2010
  • The printer sound has many aspects which define its quality because the printer has lots of components and its operation is very complicated. These sound qualities are related to the international competition in printer markets. Recordings inside anechoic chamber were analyzed and a large number of sounds were stimulated using digital signal processing technique. First subjective tests of the printer sound were conducted using semantic different method. By applying factor analysis to the subjective response, two important factors of sound quality were extracted. Second subjective tests were conducted to evaluate the quietness and the impulsiveness of the printer sounds. On the other hand, sound metrics are calculated applying psychoacoustic theories. In this paper, the nonlinear relation between subjective evaluation and sound metrics was identified using artificial neural network and the printer sound quality index was developed. Later, subjective sound quality evaluation will be estimated and evaluated using this index.

위상 처리 방식과 선착 효과를 이용한 텔레비전에서의 스테레오 음상 확대 (Stereo Sound Image Expansion Using Phase Shifting and Precedence Effect in Television)

  • 오제화;이종철
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 1998년도 추계종합학술대회 논문집
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    • pp.1239-1242
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    • 1998
  • In television stereo system, to produce a realistic sound effect is very difficult because the distance between stereo speakers is very narrow. Many signal processing methods of widening the sound image for spatial impression have been studied. One of the methods of widening the sound image is using the Precedence Effect by reflected sound. However, this method does not work effectively in lower frequencies because of directivity of a speaker. In this paper, we propose an effective method of expanding stereo image using Precedence Effect and Phase Shifting method to produce a whole band frequency sound expansion. In experiments, we confirm the usefulness of the proposed stereo sound image expansion system.

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음상 확장 기능을 갖는 텔레비전 수상기에서 센터 스피커에 관한 연구 (A Study on Center Speaker in Television Receiver with Sound Image Expansion)

  • 이상훈;김동수
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 1998년도 추계종합학술대회 논문집
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    • pp.1231-1234
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    • 1998
  • Many signal processing methods of widening the sound image for spatial impression have been studied. Most typical methods of widening the sound image are related to the phase shifting and precedence effect. However, these methods are not effective in center sound image. As listener's position moves from center to outside, the center sound image is shifted to the speaker. That is to say, the directional localization of center sound image is unstable. In this paper, we propose a television audio system including center speaker, and analyze the role of center speaker using theory of Makida and precedence effect. In experiments, we confirm the usefulness of the center speaker for the stability of center sound image.

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선착효과 및 반사음을 이용한 스테레오 음상확대 (Stereo Sound Irmge Extension Using Preredence Effect and Reflected Sounds)

  • 한찬호;이법기;정원식;고일석;최영수
    • 한국콘텐츠학회논문지
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    • 제1권1호
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    • pp.24-31
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    • 2001
  • AV시스템에서, 스테레오 스피커간의 거리가 좁기 때문에 실감나는 사운드효과를 내기가 매우 어렵다. 지금까지 공간적으로 음상을 확장시키는 신호처리방법이 많이 연구되었다. 음상을 화장하는 전형적인 방법은 대부분 위상이동과 관련된다. 그러나 이 방법은 반사성이 놓은 콘크리트 벽 구조에서는 효과적이지 못하다. 본 논문에서는 선착효과와 반사음을 이용하여 스테레오 음상을 효과적으로 확장하는 방법을 제안하였다. 실험을 통하여 기존 AV시스템이 스테레오 음상을 확장하는 방법이 실내가 넓을수록 유용함을 확인하였다.

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비음수 제약을 통한 일반 소리 분류 (Classification of General Sound with Non-negativity Constraints)

  • 조용춘;최승진;방승양
    • 한국정보과학회논문지:소프트웨어및응용
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    • 제31권10호
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    • pp.1412-1417
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    • 2004
  • 전체관적인 표현방법인 희소 코딩 또는 독릴 성분 분해(ICA)는 이전의 청각의 처리와 소리 분류의 작업을 해명하는데 성공적으로 적용되었다. 반대로 부분 기반 표현법은 뇌에서 물체를 인식하는 방법을 이해하는 또 다른 방법이다. 이 논문에서, 우리는 소리 분류의 작업에 부분기반 표현법을 학습시키는 비음수화 행렬 분해(NMF)(1) 방법을 적용하였다. 잡음이 존재할 때와 존재하지 않을 때 두 가지 상황에서, NMF를 이용하여 주파수-시간영역의 소리로부터 특징을 추출하는 방법을 설명한다. 실험결과에서는 NMF에 기반을 둔 특징이 ICA에 기반을 두어 추출한 특징보다 소리 분류의 성능을 향상시킴을 보여준다.

위너필터 후처리를 통한 비음수행렬분해 기법의 배경음 저감 성능 향상 (Improvement of Background Sound Reduction Performance by Non-negative matrix Factorization Method by Wiener Filter Post-processing)

  • 이상협;김현태
    • 한국전자통신학회논문지
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    • 제14권4호
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    • pp.729-736
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    • 2019
  • 본 논문에서는 비음수 행렬 분해 필터 뒷단에 위너필터를 추가하여 배경음 분리 성능을 향상하는 방법을 제안한다. 배경음이 혼재된 음성 신호의 경우 비음수 행렬 분해 기법으로 1차 분리된 신호에는 아직 완전히 분리되지 못한 부분이 잔류할 수 있다. 이러한 경우 위너필터에 의해 잔류하는 신호의 크기에 비례하여 줄여줄 수 있어 배경음 분리 또는 저감 효과를 기대할 수 있다. 실험을 통해 위너필터를 추가한 경우가 비음수행렬 분해 기법만 적용한 경우에 비해 저감 효과가 높은 것을 확인할 수 있었다.

Investigating the Effects of Hearing Loss and Hearing Aid Digital Delay on Sound-Induced Flash Illusion

  • Moradi, Vahid;Kheirkhah, Kiana;Farahani, Saeid;Kavianpour, Iman
    • Journal of Audiology & Otology
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    • 제24권4호
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    • pp.174-179
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    • 2020
  • Background and Objectives: The integration of auditory-visual speech information improves speech perception; however, if the auditory system input is disrupted due to hearing loss, auditory and visual inputs cannot be fully integrated. Additionally, temporal coincidence of auditory and visual input is a significantly important factor in integrating the input of these two senses. Time delayed acoustic pathway caused by the signal passing through digital signal processing. Therefore, this study aimed to investigate the effects of hearing loss and hearing aid digital delay circuit on sound-induced flash illusion. Subjects and Methods: A total of 13 adults with normal hearing, 13 with mild to moderate hearing loss, and 13 with moderate to severe hearing loss were enrolled in this study. Subsequently, the sound-induced flash illusion test was conducted, and the results were analyzed. Results: The results showed that hearing aid digital delay and hearing loss had no detrimental effect on sound-induced flash illusion. Conclusions: Transmission velocity and neural transduction rate of the auditory inputs decreased in patients with hearing loss. Hence, the integrating auditory and visual sensory cannot be combined completely. Although the transmission rate of the auditory sense input was approximately normal when the hearing aid was prescribed. Thus, it can be concluded that the processing delay in the hearing aid circuit is insufficient to disrupt the integration of auditory and visual information.

능동 신호 처리 이용한 기어의 이상 진단 (Fault Diagnosis in Gear Using Adaptive Signal Processing)

  • 이상권
    • 한국소음진동공학회:학술대회논문집
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    • 한국소음진동공학회 2000년도 춘계학술대회논문집
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    • pp.1114-1118
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    • 2000
  • Impulsive sound and vibration signals in gear are often associated with their faults. Thus these impulsive sound and vibration signals can be used as indicators in the diagnosis of gear fault. The early detection of impulsive signal due to gear fault prevents from complete failure in gear. However it is often difficult to make objective measurement of impulsive signals because of background noise signals. In order to ease the detection of impulsive signals embedded in background noise, we enhance the impulsive signals using adaptive signal processing.

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감성로봇을 위한 음원의 위치측정 및 분리 (Sound Source Localization and Separation for Emotional Robot)

  • 김경환;김연훈;곽윤근
    • 한국정밀공학회지
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    • 제20권5호
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    • pp.116-123
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    • 2003
  • These days, the researches related with the emotional robots are actively investigated and in progress. And human language, expression, action etc. are merged in the emotional robot to understand the human emotion. However, there are so many sound sources and background noise around the robot, that the robots should be able to separate the mixture of these sound sources into the original sound sources, moreover to understand the meaning of voice of a specific person. Also they should be able to turn or move to the direction of a specific person to observe his expression or action effectively. Until now, the researches on the localization and separation of sound sources have been so theoretical and computative that real-time processing is hardly possible. In this reason for the practical emotional robot, fast computation should be realized by using simple principle. In this paper the methods for detecting the direction of sound sources by using the phase difference between peaks on spectrums, and the separating the sound sources by using fundamental frequency and its overtones of human voice, are proposed. Also by using these methods, it is shown that the effective and real-time localization and separation of sound sources in living room are possible.