• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.05a
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• pp.327-327
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• 2010

자동차에서 실내음질은 구매결정 요소들 중의 하나로 그 중요성이 점차 증가하고 있다. 따라서 다양한 운전조건에서 운전자가 기대하는 실내음질의 기대 수준을 충족시켜야 한다. 소비자는 운전경험과 습관에 따라 기대하는 음질에 차이가 있고 소리에 대한 표현방식도 모호하기 때문에 이러한 주관적 특성을 하나의 통일된 표현으로 정의 하기가 어렵다. 그러나 지난 이십여 년 동안의 음질개발과 차량 실내소음 주관평가의 통계처리로 통일된 표현을 할 수 있었다. 나아가 심리음향학 및 신호처리기술의 발달과 꾸준한 음질연구결과로 소리특성을 객관적으로 나타내는 소리의 시각화가 가능하였으며, 운전자가 인식하는 주관평가와의 상관관계를 높여 차량의 대표적인 음질인자로 정량화하여 음질목표를 설정할 수 있었다. 실내소음의 구성은 엔진 투과음, 흡배기 소음, 바람 소음, 도로 기인 소음 등으로 다양하므로 소음원에 따라 음의 균형을 맞추어 조화로운 음질개발을 하는 것이 중요했다. 또한 차량 판매되는 지역에 따라 선호음이 상이하여 지역별 실내음질의 차별화가 필요했다. 궁극적으로는 운전자의 감성품질을 만족할 수 있도록 음을 제어하여 브랜드 사운드를 개발하고 있다. 이러한 실내음질을 달성하기 위한 방법으로 소음원과 전달경로에 대해 기여도를 분석하고, 경로를 구성하는 시스템 별로 세분화하여 시스템 목표를 설정하였다. 시스템 개발에 중요한 인자로 차량의 동강성 및 흡차음 성능을 들 수 있다. 특히 디젤차량의 비중이 큰 유럽업체의 차량의 동강성 및 흡차음 개발 능력은 높게 평가되고 있다. 이에 유럽의 부품전문회사가 가지고 있는 해석과 시험적인 개발 방법을 통하여 전달계 특성을 만족하기 위한 시스템의 동강성 및 흡차음 특성을 개발하고 있다. 차량음질 튜닝의 중요한 기법 중 하나로 흡배기 개발을 추진하고 있다. 친환경자동차인 하이브리드차량, 전기차량 및 연료전지차량의 경우 전기구동부품에서 발생하는 각종 이음 발생을 최소화 했다. 보행자를 보호하고 운전의 즐거움을 향상하기 위한 가상사운드 개발을 진행하고 있다. 회사 수익성 향상을 위한 원가절감 및 구조 경량화에 따른 음질악화와 연비 향상 및 배기가스 규제 강화로 고성능 고출력 엔진탑재에 따른 음질악화 요인을 극복해야 했다. 운전자의 청감은 차량의 운전성에 따라서도 크게 영향을 받게 되므로 엔진제어와 변속기제어를 통해 음질과 운전성이 조화를 이룰 수 있도록 개발하고 있다. 향후, 소음원에 따른 시스템 최적화 개발, 운전성과 음질 연계 개발과 친환경차량의 가상사운드 개발 등이 자동차 음질 개발의 중요한 이슈로 생각한다.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.405-414
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• 2002

In this paper, we propose a new objective speech quality measure using noise masking threshold for speech quality assessment of mobile telephone network environments, and verify the effectiveness of the proposed method through the experiments. For such a purpose, well known objective speech quality measures such as BSD and PSQM are first evaluated for digital mobile telephone network environments. However, these conventional methods does not have good performance under mobile networks environments compared to literary results. To be mote effective objective speech quality measure under mobile telephone environments, the proposed method employs human psychoacoustic masking effect. The DMOS, instead of MOS, is used as a subjective speech quality measure for performance evaluation. The performance comparison are carried out with speech data collected from digital mobile telephone environments. As results, the proposed measure have and average 4% higher performance, in terms of correlation, than existing objective speech quality measures such as BSD and PSQM.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.21-24
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• 2001

본 논문은 신경 회로망을 이용하여, 전화망 대역의 음성, 즉, narrowband 음성에서 wideband 음성을 복원하고자 했다. BP 알고리즘을 사용하는 기존의 신경 회로망의 경우에는 음성과 같이 복잡하고 크기가 큰 훈련데이터에 대해서는 훈련이 제대로 되지 않는 단점이 있다. 그러므로 븐 논문에서는 이를 해결하기 위해 입력으로 들어온 LPC 켑스트럼 벡터를 k-means 알고리즘을 이용하여 미리 정한 개수의 cluster로 나눈 다음, 각각의 cluster에 대해 독립적인 신경 회로망을 적용했다 이로 인해 각각의 신경 회로망은 제한되고 서로 상관관계가 많은 음성들만 훈련하면 되므로, 기존의 신경 회로망에서 생기는 훈련의 정체를 개선할 수 있었다. 또 clustering 과정에서 생기는 오류를 보완하기 위해 후보신경 로망들의 출력에 fuzzy 개념을 적용해서 최종 출력을 내도록 했다 실험 결과에서, 제안한 알고리즘은 기존의 codebook mapping 알고리즘보다 스펙트럼 거리척도에 의한 비교 및 주관적인 음질 평가 양쪽에서 개선된 성능을 보였다.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.37-46
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• 2001

This paper proposes a new noise suppression method using the Wavelet transform analysis. The noise suppressor using the Wavelet transform shows the more effective advantages in a babble noise than one using the short-time Fourier transform. We designed a new channel structure based on spectral subtraction of Wavelet transform coefficients and used the Wavelet mask pattern with more higher time resolution in high frequency. It showed a good adaptation capability for babble noise with a non-stationary property. To evaluate the performance of proposed noise canceller, the informal subjective listening tests (Mos tests) were performed in background noise environments (car noise, street noise, babble noise) of mobile communication. The proposed noise suppression algorithm showed about MOS 0.2 performance improvements than the suppression algorithm of EVRC in informal listening tests. The noise reduction by the proposed method was shown in spectrogram of speech signal.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.248-254
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• 2005

In this Paper, the structure of a split bandwidth wideband speech coder and its highband coder for tone qualify elevation are Proposed. The lowband and highband by the split bandwidth method are encoded independently applying the G.729E and MLT (Modulated Lapped Transform) residual model. In the highband structure which is encoded by low bit rate of 4kbps, the MLT residual signals are distinguished to voice and unvoice signal . The voice signals are applied to MLT peak picking method by lowband pitch period. Because transformed MLT residual signals are represented by periodic signal that have periodic peak. The unvoice signals are applied to MLT which linear prediction spectral response is added and do vector quantization. Performance for proposed 15.8kbps wideband speech coder was verified through subjective listening test.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.3 no.3
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• pp.541-546
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• 1999

The head-related transfer function (HRTF), which expresses the acoustic process from the sound source to the human ears in the free field, contains critical informations which the location of the source can be traced. It also makes it possible to realize multi-dimensional acoustic system that can approximately generate non-existing sound source. The use of non-individual, common HRTF brings performance degradation in localization ability such as front-back judgment error, elevation judgment error. In this paper, we have reduced the error on vertical plane by increasing the spectral notch level. The performance of the proposed method was Proved through subjective test that it is Possible to improve the ability to locate stationary/moving source.

• The Journal of the Acoustical Society of Korea
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• v.13 no.1
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• pp.71-83
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• 1994

This paper deals with the improvement of the synthesized speech quality and naturality in the Korean TTS(Text-to-Speech) system. We had extracted the parameters(table2) such as its amplitude, duration and pitch period in a syllable through the analysis of speech waveforms(table1) in the time domain and synthesized syllables using them. To the frequencies of the Korean pronunciation large vocabulary dictionary we had synthesized speeches selected 229 syllables such as V types are 19, CV types are 80. VC types are 30 and CVC types are 100. According to the 4 Korean syllable types from the data format dictionary(table3) we had tested each 15 syllables with the objective MOS(Mean Opinion Score) evaluation method about the 4 items i.e., intelligibility, clearness, loudness, and naturality after selecting random group without the knowledge of them. As the results of experiments the qualities of them are very clear and we can control the prosodic elements such as durations, accents and pitch periods (fig9, 10, 11, 12).

• Proceeding of Spring/Autumn Annual Conference of KHA
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• 2003.11a
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• pp.59-65
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• 2003

The present article has the purpose of evaluating subjective response of Won Buddhism sanctums by psychoacoustics experiment. Furthermore, basing on this experiment, the present article makes a suggestion of a physical sound-evaluating index. To achieve the purpose, I conducted an psychoacoustics experiment at 6 sanctuaries out of the representing 40 sanctuaries of Won Buddhism sanctums, which size is bigger than 1,000 m$^3$. I made a reanalysis with 3 Buddhist sanctums out of the subjects of psychoacoustics experiment which are generally rated superior. From the result, I can make each evaluation type into two types, sound clarity and space sense of sound. Also, I make various suggestions of physical parameter, with which I can evaluate the proper sound condition for Won Buddhism sanctums.

• Proceeding of Spring/Autumn Annual Conference of KHA
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• 2004.11a
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• pp.355-360
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• 2004

This study is a research that evaluates and analyzes on the subjective reaction of Won buddhism small sanctum through psychoacoustics experiment. For doing it, after choosing of 5 Won buddhism small sanctums of which its building volume at $400m^{3}{\sim}650m^{3}$, a psychoacoustics experiment has been conducted. Based on those A, B, C, E temple-halls which appeared as higher than 'average' in the result of subjective reaction evaluation of the Won buddhism small sanctum through psychoacoustics experiment, it could be classified as the conclusion of the factor analysis: the first factor is 'space sense of sound', the second factor is 'intimacy-feeling with sound' and the third factor is 'clearness of sound', respectively.