• Title/Summary/Keyword: sound impulse response

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Implementation of ray tracing simulator for extracting sound field parameters (음장파라미터 추출을 위한 음선추적 시뮬레이터의 구현)

  • Lee, Deok-Su;Seong, Goeng-Mo
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.3
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    • pp.82-89
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    • 1995
  • A sound field simulator is constructed to obtain the sound field paramaters such as the magnitudes and directions of early reflections with moderate efforts. The proposed simulator is based on the hybrid ray tracing method that traces rays reached the listener position and convert them to image sound sources. By this approach, we can obtain the directional impulse response relatively easily with minimum casts. Simulation experiment results of several performace places are reported to how the versatility of the proposed simulator system.

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Korea Industrial Standardization Draft for ISO 354 (ISO 354 (잔향실법 흡음률 측정방법)의 KS 규격화)

  • Jung, Sung-Soo;Kook, Chan;Kim, Sun-Woo
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2003.11a
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    • pp.857-860
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    • 2003
  • We introduced the basic direction of revision for KS F 2805 (method for measurement of sound absorption coefficient in a reverberation room). It is fundamentally equal to ISO 354 (measurement of sound absorption in a reverberation room) which was state of ISO/FDIS 354. Two main points were strongly reviewed. First, the sound absorption coefficient values for Type J test specimen mounting was investigated. Next, the reverberation time difference between the conventional method and new additional one, impulse response method were compared.

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Effect of the Measuring Method of Reverberation Time Using Impulse Response Method on the Normalized Impact Sound Pressure Level (임펄스응답적분법을 이용한 잔향시간의 측정방법이 규준화 바닥충격음레벨에 미치는 영향)

  • Lee, J.W.;Kwon, Y.P.
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.16 no.1 s.106
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    • pp.34-39
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    • 2006
  • For the evaluation of the normalized impact sound pressure level, the reverberation time of the receiving room should be measured. This paper deals with the effect of the time constant of FFT analyzer and the measuring points on reverberation time. It is found that the time constant should be in the range between 10 ms and 35 ms. While the effect of measuring points on the reverberation time is significant when the bandwidth is narrow it is negligible in the evaluation of the normalized impact sound pressure level.

3-D Sound-Field Creation Implementing the Virtual Reality Ship Handling Simulator(I): HRTF Modeling (가상 현실 선박 조종 시뮬레이터 구현을 위한 3차원 음장생성(I) : 머리전달함수 모델링)

  • 임정빈
    • Journal of the Korean Institute of Navigation
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    • v.22 no.3
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    • pp.17-25
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    • 1998
  • This paper describes elemental technologies for the creation of three-dimensional(3-D) sound-field to implement the next-generation Ship Handling Simulator with human -computer interaction, known as Virtual Reality. In the virtual reality system, Head-Related Transfer Functions(HRTF's) are used to generate 3-D sound environmental context. Where, the HRTF's are impulse response characterizing the acoustical transformation in a space. This work is divided into two parts, the part Ⅰis mainly for the model constructions of the HRTF's, the part Ⅱis for the control of 3-D sound-field by using the HRTF's . In this paper, as first part, we search for the theory to formulate models of the HRTF's which reduce the dimensionalityof the formulation without loss of any directional information . Using model HRTF's we report results from psychophysical tests used to asses the validity of the proposed modleing method.

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Headphone-based multi-channel 3D sound generation using HRTF (HRTF를 이용한 헤드폰 기반의 다채널 입체음향 생성)

  • Kim Siho;Kim Kyunghoon;Bae Keunsung;Choi Songin;Park Manho
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.42 no.1
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    • pp.71-77
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    • 2005
  • In this paper we implement a headphone-based 5.1 channel 3-dimensional (3D) sound generation system using HRTF (Head Related Transfer Function). Each mono sound source in the 5.1 channel signal is localized on its virtual location by binaural filtering with corresponding HRTFs, and reverberation effect is added for spatialization. To reduce the computational burden, we reduce the number of taps in the HRTF impulse response and model the early reverberation effect with several tens of impulses extracted from the whole impulse sequences. We modified the spectrum of HRTF by weighing the difference of front-back spec01m to reduce the front-back confusion caused by non-individualized HRTF DB. In informal listening test we can confirm that the implemented 3D sound system generates live and rich 3D sound compared with simple stereo or 2 channel down mixing.

Simulation Software for Instrument Placement on Stage Based on the Acoustic Properties of Concert Halls (연주홀 특성을 적용한 악기 무대 배치 시뮬레이션 소프트웨어 제작)

  • Kim, Wan-Jung;Yoo, Won-Dae;Kim, Keun-Hyung;Lee, Ki-Beom;Yeo, Woon-Seung
    • Journal of Korea Multimedia Society
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    • v.13 no.7
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    • pp.960-972
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    • 2010
  • In this paper, we present a software for placing instruments on stage based on the acoustic properties of the concert hall. In order to simulate the changes in sound depending on the positions of the instruments, we incorporated the idea of location-based reverberation effect which can be realized through the convolution of instrument sounds with the impulse responses from the respective instrument positions. And we developed a software with a real-time convolution engine which enables the user to conveniently simulate the resulting sound of various instrument placements. The software was tested with the impulse response data measured at two concert halls of the National Center for Korean Traditional Performing Arts and Korean traditional instrument sounds. Results of these experiments show that simulated reverberation effects properly represent the spatial placement of instruments on stage.

Numerical Analysis for Modeling of Sound Absorbing Medium using Transmission Line Matrix Modeling (전달선로행렬법을 이용한 흡음재 모델링에 대한 수치해석)

  • Park, Kyu-Chil;Yoon, Jong-Rak
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.16 no.8
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    • pp.1599-1605
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    • 2012
  • We introduced an approach of modeling of a sound absorbing medium that had different absorbing coefficient according to frequency. To obtain the time domain result of the frequency characteristics of a sound absorbing medium, transmission line matrix modeling was used. To input sound absorbing effect in TLM modeling, we added a FIR filter at a node instead of absorbing component using resistance component. There were simulated the characteristics of time-shift, low pass filter, high pass filter using the FIR filter with 7-tap coefficients, then compared with theoretical results. From various simulation results, we could find that added FIR filter coefficient in TLM modeling was an useful way to model a sound absorbing medium.

A Stduy on Acoustics Estimation of PANSORI hall by Scale Model (축척모형을 이용한 판소리 홀의 음향평가에 관한 연구)

  • Shin, Young-Moo;Chung, Sa-Hee
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.6
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    • pp.66-72
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    • 1996
  • In order to the sound effects and acoustics estimation of PANSORI hall, we are researched into the impulse response measuring and convolution integral of dry music(PANSORI) by using 1/10 scale model. Results are as follwo. First, impulse responses are measured by spark sound of electrodes and it is absolutely necessary many times of synchronous calculating for the obtain to enough S/N ratio. Second, a simulation technique of scale model is confirmed one of an effectual method of indoor acoustics estimation. Further, using the these new techniques and hearing test, its are recognized that reverberation time of PANSORI hall is about $1.0{\sim}$12.$ second suitable.

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Outdoor Noise Propagation: Geometry Based Algorithm (옥외 소음의 전파: 음 추적 알고리즘)

  • 박지헌;김정태
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.4
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    • pp.339-438
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    • 2002
  • This paper presents a method to simulate noise propagation by a computer for outdoor environment. Sound propagated in 3 dimensional space generates reflected waves whenever it hits boundary surfaces. If a receiver is away from a sound source, it receives multiple sound waves which are reflected from various boundary surfaces in space. The algorithm being developed in this paper is based on a ray sound theory. If we get 3 dimensional geometry input as well as sound sources, we can compute sound effects all over the boundary surfaces. In this paper, we present two approaches to compute sound: the first approach, called forward tracing, traces sounds forwards from sound sources. while the second approach, called geometry based computation, computes possible propagation routes between sources and receivers. We compare two approaches and suggest the geometry based sound computation for outdoor simulation. Also this approach is very efficient in the sense we can save computational time compared to the forward sound tracing. Sound due to impulse-response is governed by physical environments. When a sound source waveform and numerically computed impulse in time is convoluted, the result generates a synthetic sound. This technique can be easily generalized to synthesize realistic stereo sounds for virtual reality, while the simulation result is visualized using VRML.

An attenuation effect of noise according to the direction of secondary sound source in duct ANC system (Duct ANC 시스템에서 2차음원 방향별 소음감소효과)

  • Lee, Hyung-Seok;Lee, Eung-Suk
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2008.11a
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    • pp.497-502
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    • 2008
  • In this paper, we studied on an attenuation effect of automobile exhaust noise according to the direction of secondary sound source in duct ANC system. Automobile exhaust noise was recorded at 800rpm. 3500rpm and 5000rpm of a diesel engine. Directions of loudspeaker(second sound source) can be exchanged to $30^{\circ}$, $90^{\circ}$ and $150^{\circ}$ against the primary noise flow by acrylic ducts to be made for experimentation. DSP board with TMS320C6416 chip of Texas Instrument Co used to control adaptive ANC system. This ANC system is based on the single-channel FxLMS algorithm. In experiment result, when the loud speaker direction was $150^{\circ}$, the attenuation effect showed largely. In case of $90^{\circ}$ duct, the noise was a little increased. In case of $30^{\circ}$ duct, the noise was a little increased or decreased according to the frequency range and the sound pressure(dB) of exhaust noise to comply with engine rpm.

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