• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.1
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• pp.112-118
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• 2012

In daily life, there are many information broadcasting by using voice information systems. If surrounding noises are mixed with the information signals, the clarity of the signal become down graded too much to understand. Surrounding noises are not uniformed, but very irregular signals always changing. Therefore, it is very hard to control the output signals along with the irregular signals. This paper suggests a method to change the level of the voice information adapting to the surround noise in the white noise environment. The surround noise level is measured by subtracting the stored output voice signal from the voice signal degraded by the noise. The noise is used to estimation of SNR. And, the method to change the output level of voice signal adapting to the noise level. The suggested adaptive voice information system has the advantage to improve listeners' speech perception and to use amplifier's energy effectively.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.4 s.304
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• pp.159-166
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• 2005

This thesis deals with a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. First, the cross-talk cancellation method that eliminates cross-talks on the path from left speaker to right ear and from right speaker to left ear is explained. Then the application and replaying method using the cross-talk cancellation explained here is introduced. The acoustical model for cross-talk cancellation is the free field model This model minimizes distortion of sound. Many experts also make studies on this model. I used the bark scale sound quality compensation based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain.

• The KIPS Transactions:PartB
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• v.11B no.3
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• pp.345-352
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• 2004

This paper deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. 1 used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound qualify assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the tort chamber which satisfies the ITU-R specifications. 1 uses the IACC(Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38dB separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4∼0.5 point Increased then before.

• Journal of Korea Game Society
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• v.12 no.1
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• pp.103-111
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• 2012

Games in 1970s have used the savings of analog wavelength as in music and the like on hardwares of small cassette, gramophone, and the like. Due to these configuration elements, durability has to be declined so much. In the event of using the music on the video game, more affordable method is to use the computer chip to convert the analog sound into the computer code to convert into the electric wavelength to send to the speaker. The sound effect of the game is generated in this method. The technical limit has been gradually overcome to grant more freedom to the composers and the sound track pre-recorded on the optic disc and the like has emerged. The game developers of today have made several attempts on the technology to produce the game music. This study has contemplated the process of advancement in the change of game music production with the influence on technology and business.

• Journal of Digital Contents Society
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• v.9 no.2
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• pp.203-211
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• 2008

Due to the advance of H/W and S/W techniques to play 3D sound, the virtual space contented by 3D graphics and sounds can provide users more improved realities and vividness. However for the small 3D contents developers and companies, it is hard to implement 3D sound techniques because the implementation requires expensive sound engines, 3D sound technical understanding and 3D sound programming skills. Therefore 3D-sound-playing-interface is necessary to easy and cost-effective 3D sound implementation. Using this interface, graphics experts can easily add 3D sound techniques to their applications. In this paper, the followings are designed and implemented as a preliminary stage in the way of developing the 3D sound playing interface. First, we develop 3D sound S/W modules converting mono to 3D sound in PC based systems. Second, we develop the interconnection modules to map 3D graphic objects and sound sources. The developed modules in this paper can allow the user to percept sound source position and surround effect at the moving positions in the virtual world. In the coming works, we are going to develop the more completed 3D sound playing interface consisted of the synchronization technique for sound and moving objects, and HRTF.

• The Journal of the Korea Contents Association
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• v.5 no.6
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• pp.74-86
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• 2005

In this paper, we proposed an algorithm for the approximation of high order FIR filters by low order IIR filters to efficient implementing two channel 3-D surround sound systems using Head-related transfer functions(HRTFs). The algorithm is based on a concept of the balanced model reduction. The binaural sounds using the approximated HRTFs are reproduced by headphone, and serves as a cue of sound image localization. HRTFs of dummy-head are approximated from 512-order FIR filters by 32-order IIR filters and compare with each other. .Experiment of sound image are carried out for 10 participants. We perform the experiment based on computer simulation and hardware experiment with TMS320C32. The results of the experiments show that the localization using the approximated HRTFs is the same accuracy as the case of FIR filters that simulate the HRTFs.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.541-544
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• 2004

본 논문에서는 서라운드 패닝알고리즘에서의 음색보정필터 설계기법에 대한 것으로 기존 패닝알고리즘에 음색보정필터를 추가하여 가상음원의 음색왜곡을 보정하는 알고리즘을 제안한 다. 가상음상과 실제 라우드 스피커의 머리전달함수 분석을 통해 기존 일정파워패닝알고리즘의 음색왜곡 문제점을 지적하고 이를 완화시키기 위한 새로운 패닝알고리즘을 제안한다.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.693-704
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• 2011

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music contents. MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved Study on DIS at the 96th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC, ACELP, and TCX) for low frequency regions, SBR for high frequency regions, the MPEG Surround for stereo information, and window transition technology for smoothing transition between various core coder. USAC can provide consistent sound quality for both speech and music contents and can be applied to various applications such as multi-media download to mobile devices, digital radio, mobile TV and audio books.

• The Journal of the Acoustical Society of Korea
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• v.28 no.7
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• pp.589-598
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• 2009

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music content MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved WD3 at the 88th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC ACELP and TCX) for low frequency regions, SBR for high frequency regions and the MPEG Surround tool for stereo information. USAC can provide consistent sound quality for both speech and music content and can be applied to various applications such as multi-media download to mobile device Digital radio Mobile TV and audio books.

• Journal of Broadcast Engineering
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• v.20 no.5
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• pp.667-675
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• 2015

As the interest on the high-definition and high-quality broadcasting is increased, the request on the high quality sound signal is enlarged as well as on the video signal's quality. One factor contributing to the high-quality of audio signal is an expansion of reproduction channels like 10.2channel and 22.2channel, but there is a problem of speaker installation issue of these many channels. One solution to solve this problem, we can use frontal loudspeaker array reproduction technique making virtual surround sound. So in this paper, we introduce theocratical analysis on the Wave Field Synthesis used for speaker array based sound reproduction and also present the result about the subjective listening test of reproduction performance based on this technique to check the perfoemance of this system. As a result, we showed WFS based frontal loudspeaker array reproduction method could provide sufficient performance compared to conventional discrete 5.1 channel reproduction method.