• 한국소음진동공학회논문집
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• 제18권2호
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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.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 추계학술대회논문집
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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.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2002년도 춘계학술대회논문집
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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 %.

• 전자공학회논문지
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• 제50권12호
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• pp.217-223
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• 2013

본 논문에서는 오디오 핑거프린팅 방식과 스파셜 오디오 처리 방식을 결합한 오디오 핑거프린팅 기반 입체음향 재현 시스템을 제안한다. 제안된 시스템에서는 변조스펙트럼 기반의 명확한 오디오 정점 핑거프린트를 이용하여 잡음환경에서 오디오 핑거프린팅 시스템의 검색정확도를 향상시켰으며, 메타데이터로 제공되는 스파셜 오디오 정보는 청취자에게 소리가 실제로 녹음된 공간에서 소리를 듣는 것 같은 느낌을 준다.

• 한국항해학회지
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• 제16권4호
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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.

• IEIE Transactions on Smart Processing and Computing
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• 제5권3호
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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%.

• The Journal of the Acoustical Society of Korea
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• 제16권3E호
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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.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 추계학술대회논문집
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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.

• 한국소음진동공학회논문집
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• 제18권2호
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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.

• 한국항해항만학회:학술대회논문집
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• 한국항해항만학회 1993년도 춘계학술발표회
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• pp.3-10
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• 1993