• Title/Summary/Keyword: Sound Reproduction

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HRTF-field Reproduction for Robust Virtual Source Imaging (머리 전달 함수장 재현을 통한 광대역 입체 음향 구현)

  • Choi, Joung-Woo
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.18 no.2
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    • pp.199-207
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    • 2008
  • A hybrid technique that combines the advantages of binaural reproduction and sound field reproduction technique is proposed. The concept of HRTF-field, which is defined as the set of HRTFs corresponding to the various head dislocations, enables us to realize virtual source imaging over a wide area. Conventional binaural($2{\times}2$) reproduction system is redefined as a MIMO system composed of multiple control sources and multiple head locations, and HRTF variations corresponding to various head movement are quantified. Through the direct control of HRTF-field, reproduction error induced by head dislocation can be minimized in least-square-error sense, and consequential disturbances on the virtual source image can be reduced within a selected area. Simple lateralization examples are investigated, and the reproduction error of the proposed technique is compared to that of higher-order Ambisonics.

HRTF-field reproduction for robust virtual source imaging (머리 전달 함수장 재현을 통한 광대역 입체 음향 구현)

  • Choi, Joung-Woo
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2007.11a
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    • pp.997-1004
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    • 2007
  • A hybrid technique that combines the advantages of binaural reproduction and sound field reproduction technique is proposed. The concept of HRTF-field, which is defined as the set of HRTFs corresponding to the various head dislocations, enables us to realize virtual source imaging over a wide area. Conventional $2{\times}2$ definition is redefined as a MIMO system composed of multiple control sources and multiple head locations, and HRTF variations corresponding to various head movement are quantified. Through the direct control of HRTF-field, reproduction error induced by head dislocation can be minimized in least-square-error sense, and consequential disturbances on the virtual source image can be reduced within a selected area. Simple lateralization examples are investigated, and the reproduction error of the proposed technique is compared to that of Higher-order Ambisonics.

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A DIRECT INVERSE-BASED CROSS-TALK CANCELLATION METHOD FOR STEREO AUDIO SYSTEMS (직접 역필터 설계법을 이용한 스테레오 재생시스템의 Cross-talk 제거)

  • Kim, Sang-Myeong;Dogeun Han;Semyung Wang
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2002.05a
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    • pp.559-564
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    • 2002
  • Cross-talk cancellation, inverse filter design or deconvolution in a generic term, is a vital process for a virtual sound realization in the stereo sound reproduction system. Most, if not all, of the design algorithms available for the inverse filter are based on a linearized model of the real physical plant. The result of such a plant-based design method, which may be referred to here as the indirect method, is biased due to both modelling and inversion errors. This paper presents a novel direct cross-talk cancellation method that may be free from the inversion error. The direct method can directly models the inverse filter by a suitable rearrangement of the input and output ports of the original plant so that no inversion is required here. Advantages are discussed with various experiments in an anechoic chamber using a PC soundcard. Binaural reproduction tests conducted showed that the conventional indirect method yields about 8 % reproduction performance error on both ear positions, whereas the direct method offers about 3 %.

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Audio Fingerprinting Based Spatial Audio Reproduction System (오디오 핑거프린팅기반 입체음향 재현 시스템)

  • Ryu, Sang Hyeon;Kim, Hyoung-Gook
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.12
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    • pp.217-223
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    • 2013
  • This paper proposes a spatial audio reproduction system based on audio fingerprinting that combines the audio fingerprinting and the spatial audio processing. In the proposed system, a salient audio peak pair fingerprint based on modulation spectrum improves the accuracy of the audio fingerprinting system in real noisy environments and spatial audio information as metadata gives a listener a sensation of being listening to the sound in the space, where the sound is actually recorded.

Computer Simulation on the Modelling of OSS Equalizer for the Reproduction of Original Sound Field (원음장 재생을 위한 OSS 등화기의 모델링에 관한 컴퓨터시뮬레이션)

  • 임정빈;김천덕
    • Journal of the Korean Institute of Navigation
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    • v.16 no.4
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    • pp.55-63
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    • 1992
  • This computer simulation is the basic research for realize a real-time hardware of the reproduction system in original sound field with two loudspeakers based on the OSS(Ortho Stereophonic System) method which was proposed by Hamada of Japan in 1983. Through the computer simulation, presumed the system function of OSS equalizer using HRTF(Head Related Transfer Function), constructed the model of OSS equalizer and , evaluated the modelling OSS equalizer by evaluation formula. The obtained results are summarized as follows : 1) By the modelling OSS equalize operate as inverse filter of HRTF, an input signal reproduced effectively. 2) Known that the real-time hardware of OSS equalizer can be made by the fast convolution between the impulse response of OSS equalizer and input speech signal. 3) Since the system function of OSS equalizer presumed from HRTF, the study on the measuring of HRTF have to proceed.

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Automatic Vowel Sequence Reproduction for a Talking Robot Based on PARCOR Coefficient Template Matching

  • Vo, Nhu Thanh;Sawada, Hideyuki
    • IEIE Transactions on Smart Processing and Computing
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    • v.5 no.3
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    • pp.215-221
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    • 2016
  • This paper describes an automatic vowel sequence reproduction system for a talking robot built to reproduce the human voice based on the working behavior of the human articulatory system. A sound analysis system is developed to record a sentence spoken by a human (mainly vowel sequences in the Japanese language) and to then analyze that sentence to give the correct command packet so the talking robot can repeat it. An algorithm based on a short-time energy method is developed to separate and count sound phonemes. A matching template using partial correlation coefficients (PARCOR) is applied to detect a voice in the talking robot's database similar to the spoken voice. Combining the sound separation and counting the result with the detection of vowels in human speech, the talking robot can reproduce a vowel sequence similar to the one spoken by the human. Two tests to verify the working behavior of the robot are performed. The results of the tests indicate that the robot can repeat a sequence of vowels spoken by a human with an average success rate of more than 60%.

HRTF Measurement and Its Application for 3-D Soung Localization

  • Kang, Kyeong Ok;Kang, Dong-Gyu;Hahn, Minsoo;Jho, Moon Jae;Jeong, Dae-Gwon
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.3E
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    • pp.50-55
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    • 1997
  • Based on the anthropometric data of Korea male adults, a head and torso simulator(HATS) is constructed to measure its head related transfer functions (HRTFs) which can be used for three dimensional (3-D) sound localization. The HRTFs binaural impulse responses, are measured in an anechoic chamber using a burst maximum length sequence (MLS) signal of 65,535 samples and 32,768 samples acquisition at the sampling rate of 75.47kHz. Also measured are the impulse responses of a driving loudspeaker and some headphones for sound reproduction to get the exact HRTF of the HATS-alone. Through a post-processing procedure, the impulse-version HRTFs at the sampling frequency of 44.1 kHz, which have filter lengths of 512 points, are finally obtained. As an application of the measured HRTFs, a 3-D sound processor for headphone reproduction has been developed. The signal intervals to be processed can be selected and each interval is manipulated to have its diretionality and distance information by using corresponding HRTF and energy control.

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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.