• Title/Summary/Keyword: Sound Reverberator

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Digital Filter Model for Analog Helical Coil Spring Reverberator (헬리컬 코일 스프링 잔향기의 디지털 필터 모델)

  • Park Joon;Chon Sang-Bae;Lee Jong-Hoon;Sung Koeng-Mo;Song Sang-Seob
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.6
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    • pp.291-297
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    • 2006
  • This paper proposes a new Digital Reverberator that models Analog Helical Coil Spring Reverberator for guitar amplifiers. While the conventional digital reverberators are proposed to provide better sound field mainly based on room acoustics, no algorithm or analysis of digital reverberators those model Helical Coil Spring Reverberator was proposed. Considering the fact that approximately $70{\sim}80$ percent of guitar amplifiers are still with Helical Coil Spring Reverberator, research was performed based not on Room Acoustics but on Helical Coil Spring Reverberator itself as an effector. After performing simulations with proposed algorithm, it was confirmed that the Digital Reverberator by proposed algorithm provides perceptually equivalent response to the conventional Analog Helical Coil Spring Reverberators.

Fast Convolution Method using Psycho-acoustic Filters in Sound Reverberator (잔향 생성기에서 심리 음향 필터를 이용한 고속 컨벌루션 방법)

  • Shin, Min-Cheol;Wang, Se-Myung
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2007.11a
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    • pp.1037-1041
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    • 2007
  • With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral psycho-acoustic filters considering masking effects are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for realtime implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

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Fast Convolution Method Using Real-time Masking Effects in Sound Reverberator (잔향 생성기에서 실시간 마스킹 효과를 이용한 고속 컨벌루션 방법)

  • Shin, Min-Cheol;Wang, Se-Myung
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.18 no.2
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    • pp.231-237
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    • 2008
  • With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral real-time masking blocks are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for real-time implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

A Very Low-Bit-Rate Analysis-by-Synthesis Speech Coder Using Zinc Function Excitation (Zinc 함수 여기신호를 이용한 분석-합성 구조의 초 저속 음성 부호화기)

  • Seo Sang-Won;Kim Jong-Hak;Lee Chang-Hwan;Jeong Gyu-Hyeok;Lee In-Sung
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.6
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    • pp.282-290
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    • 2006
  • This paper proposes a new Digital Reverberator that models Analog Helical Coil Spring Reverberator for guitar amplifiers. While the conventional digital reverberators are proposed to provide better sound field mainly based on room acoustics, no algorithm or analysis of digital reverberators those model Helical Coil Spring Reverberator was proposed. Considering the fact that approximately $70{\sim}80$ percent of guitar amplifiers are still with Helical Coil Spring Reverberator, research was performed based not on Room Acoustics but on Helical Coil Spring Reverberator itself as an effector. After performing simulations with proposed algorithm, it was confirmed that the Digital Reverberator by proposed algorithm provides perceptually equivalent response to the conventional Analog Helical Coil Spring Reverberators.

Analysis and Design Algorithm of Time Varying Reverberator for Low Memory Applications (저전력 환경에 적합한 시간변화 잔향기의 분석 및 설계 알고리듬)

  • Choi Tack-Sung;Park Young-Cheol;Youn Dae-Hee
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.43 no.5 s.311
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    • pp.62-71
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    • 2006
  • Development of an artificial reverberation algorithm with low memory requirements has been an issue of importance in applications such as mobile multimedia devices. One possible solution to this problem is to embed a time-varying all-pass filter to the feedback loop of the comb filter. In this paper, theoretical and perceptual analyses of reverberators embedding time-varying all-pass filters are presented. The analyses are to iud a perceptually acceptable degree of phase variation by the all-pass filter. Based on the analyses, we propose a new methodology of designing reverberators embedding time-varying all-pass filters. Through the subjective tests, we showed that, even with smaller memory, the proposed method is capable of providing perceptually comparable sound quality to the conventional methods involving time-invariant parameters.