• Title/Summary/Keyword: 음향상

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Noise Map Analysis for the Design of Noise Barrier at School Site (학교부지의 방음벽 설계를 위한 소음지도 해석)

  • Yun, Junho;Kim, Wonjin
    • Journal of Korean Society of Environmental Engineers
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    • v.34 no.4
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    • pp.232-238
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    • 2012
  • In this study, the noise mapping simulation is executed to design an effective barrier reducing noise levels of a school site. The geographical features of the ambient site and the school buildings are modelled in detail in order to consider sound propagation, deflection, and absorption phenomena etc. The main sound source, sound power level of expressway, is estimated on the basis of measured noise levels at several points of the site. The noise mapping simulation is performed by using ENPro, environmental noise prediction program based on ISO 9613 to analysis the effectiveness of noise barrier. Consequently, the noise barrier is designed to meet an environmental noise standard and satisfy low cost and safety conditions.

Design of a Variable half rate speech codec (가변율 half rate 음성 부호화기의 설계)

  • 성호상
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1998.06e
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    • pp.293-296
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    • 1998
  • 본 논문에서는 다양한 멀티미디어 서비스를 위해 가변율 half rate 음성 부호화기를 설계하였다. 유, 무성음과 묵음의 구분을 위해 본 논문에서는 프레임 에너지와 음성 파라메터들을 이용한 효과적인 voicing 결정 알고리즘을 사용하였다. 유성음을 위한 half rate 음성 부호화기는 저속에서 좋은 특성을 보이는 generalized AbS구조를 이용하였다. LPC 계수는 LSP 계수로 변환한 후 predictive 2-stage VQ를 통해서 양자화하며, 여기 신호는 음질저하를 최소화하며 복잡도를 감소시킨 shift 방식의 대수적 고정 코드북 구조를 사용하고, 적응코드북과 여기코드북의 이득은 VQ로 양자화 하였다. 무성음을 위한 부호화기는 대부분이 유성음을 위한 부호화기와 동일하지만, 무성음에서는 피치간 상관도가 매우 낮으므로 피치 보간 방법을 사용하지 않고 개루프로 피치 lag를 찾은 후 전체 프레임에 사용한다. 1 kb/s 부호화기는 묵음 구간과 주변소음 구간에 사용되며 이 구간의 신호를 피치 성분이 미약한 주변소음들로 제한하고 이에 최적인 부음성 부호화기를 설계하였다. 최종적으로 완성된 가변율 half rate 부호화기는 voice activity factor(VAF)가 0.47인 시험음성에서 약 2.6 kb/s의 평균 전송률을 보였다. 주관적 음질 평가의 일환으로 IS-96 표준 코덱인 가변율 8 kb/s QCELP와 A-B preference 시험을 실시하였다. 시험 결과 평균전송률이 약 2배인 가변율 8 kb/s QCELP 보다 우수한 음질 성능을 보였다.

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A Study on the Standard and Classification of Impact Sound Insulation Performance for Apartment House(II) (바닥 衝擊音 遮音性能基準 및 等級化에 관한 硏究(Ⅱ))

  • Kim, Sun-Woo;Son, Chul-Bong;Song, Yong-Sik;Jang, Gil-Soo;Lee, Tae-Kang;Kook, Chan;Kim, Jae-Soo;Han, Myung-Ho
    • The Journal of the Acoustical Society of Korea
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    • v.9 no.4
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    • pp.5-17
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    • 1990
  • This study aims to establish the appropriate class and standards of sound insulation performance, which are suitable for domestic conditions. For, these purpose, field surveys were conducted, correlationship between subjective response and impact sound insulation performance and then, the applicability of ISO and JIS evaluation method was reviewed. As a result, the JISs method is more advisable than the ISO to evaluate Impact sound insulation performance of domestic apartment house and it is desirable to use both light and heavy weight source (tapping machine, tire) as standard impact sources. And, whether including 63Hz bandwidth or not, the difference was recognized very small in calculating impact sound insulation index of JIS A 1419.

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A Study of QMSA Characteristics According to the Gap Variance between Ground Plane and Radiation Patch (접지판과 방사 패치 사이의 간격 변화에 따른 QMSA의 특성 연구)

  • Ryu, Hyun;Park, Sung-Kyo;Park, Chong-Baek
    • The Journal of the Acoustical Society of Korea
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    • v.18 no.4
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    • pp.11-16
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    • 1999
  • In this paper, we designed and fabricated QMSA(Quarter-Wavelength Microstrip Antenna) for 850MHz band on the CGP-500 PTFE substrate(by CHUKOH company) with ε/sub r/=2.6, H=1.6mm(±0.08), where width of the radiation patch is .identical with that of the ground plane. A well matched feed point was obtained by experiments and this QMSA was fed by using prove feed method. The resonant frequencies and the return losses were mcasured by cutting the gap L₃ between the radiation patch and the ground plane, with a 5mm cutting length, step by step. As a result, we found that the measured return losses were decreased and the resonant frequencies were increased when the gap L₃ was shorter, especially under 10mm, unlike we had expected.

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Study of Speech Recognition System Using the Java (자바를 이용한 음성인식 시스템에 관한 연구)

  • Choi, Kwang-Kook;Kim, Cheol;Choi, Seung-Ho;Kim, Jin-Young
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.6
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    • pp.41-46
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    • 2000
  • In this paper, we implement the speech recognition system based on the continuous distribution HMM and Browser-embedded model using the Java. That is developed for the speech analysis, processing and recognition on the Web. Client sends server through the socket to the speech informations that extracting of end-point detection, MFCC, energy and delta coefficients using the Java Applet. The sewer consists of the HMM recognizer and trained DB which recognizes the speech and display the recognized text back to the client. Because of speech recognition system using the java is high error rate, the platform is independent of system on the network. But the meaning of implemented system is merged into multi-media parts and shows new information and communication service possibility in the future.

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On Estimating the Incident Angles of Wide Band Signals in Low SNR Environment (신호 대 잡음비가 낮은 경우 광대역 신호의 입사각 추정)

  • Jo, Jeong-Gwon;Hwang, Yeong-Su;Cha, Il-Hwan;Yun, Dae-Hui
    • The Journal of the Acoustical Society of Korea
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    • v.8 no.4
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    • pp.44-52
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    • 1989
  • The UCERSS (Unit Circle Eigendecomposition Rational Signal Subspace) algorithm has extended MUSIC (MUltiple Signal Classification ) by using eigendecomposition on the unit circle in order to estimate incident angles of multiple wide band signals. The purpose of this thesis is to further extend the UCERSS to be able to estimate the direction of arrivals of multiple wide band signals in very low SNR . The wide band ESPRIT (Estimation of Signal Parameter via Rotational Invariance Technique) uses covariance difference matrices to reduce noise components. In this paper the wide band ESPRIT which combines the ideas of UCERSS and ESPRIT Is proposed. Computer simulation results Indicate that the performances of the proposed approaches are superior to those of the UCERSS in very low SNR.

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Simple Signal Reconstruction by Faster Adaptive MRP Algorithm (고속 적응 MRP 알고리즘에 의한 저주파 신호 복원)

  • Jeong, Won-Yong;Kim, Jong-Su;Choe, Tae-Won;Bae, Jin-Ho
    • The Journal of the Acoustical Society of Korea
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    • v.11 no.2
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    • pp.5-14
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    • 1992
  • In the fields of astronomy, communication, X-ray crystallography, and engineering, it is a very important and useful fact that the original signal can be reconstructed from a partial information, only spectral magnitude or phase, of the signal. In this paper, we proposed a modified iterative algorithm to solve the Magnitude Retrieval Problem (MRP) for 1-D, 2-D signals. In oder to accelerate the convergence rate, the unit constant initial function which is used in the references is replaced by the exponential initial function for the modified adaptive iterative method. As a result, MRP with 1-D signal and low-pass detail image is significantly enhanced from an iterative convergence rate and a computer storage memory points of view.

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Synthesizing the classical guitar sound using the Physical Model with string properties (현의 특성이 고려된 Physical 모델을 이용한 클래식 기타 음 합성)

  • Kang Myung-Soo;Kim Kyoo-Nyun
    • Proceedings of the Acoustical Society of Korea Conference
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    • spring
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    • pp.255-258
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    • 1999
  • 본 논문에서는 파동 방정식으로부터 클래식 기타의 Physical 모델을 유도해 낸 후 이를 구현하였다. 이러한 모델을 이용해 별도의 음원 데이터를 사용하지 않고 현재 전자 음악에서 일반적으로 사용되는 table look-Up 방식보다 효율적으로 악기 음을 함성 할 수 있도록 하였다. 파동 방정식은 현의 장력, 길이 및 질량 데이터를 이용해 현의 움직임을 표현한 것이며 이 식으로부터 Fourier Series를 유도하고 다시 Z 변환을 거쳐 현의 운동을 모델링하였다. 이 과정에서 현의 양끝에서 반사되는 신호의 크기를 모델링에 포함 시켰다. 이러한 현의 모델은 모든 종류의 현악기에 공통으로 적용될 수 있으며 현의 장력 길이, 질량 데이터를 변화해 다양한 현의 특성들을 모델링 할 수 있다. 또 음 합성을 위해 현의 초기 상태 및 연속되는 입력 데이터를 바꿔 클래식 기타의 다양한 음들을 합성 할 수 있다. 클래식 기타의 Physical 모델을 평가하기 위해, 실제 악기 음 및 table look-up 방식으로 합성된 음들을 녹음해 서로 비교하였다. 시간 및 주파수 도메인 상에서 비교가 이뤄 졌으며 table look-up 합성 방식에서 모든 주파수대가 동일하게 감소하고 비슷한 음역에서 음 높이에 적합한 배음 주파수 비율을 조절할 수 없는 등, 각 을의 특성들을 정확히 묘사할 수 없는 문제점을 극복할 수 있었다.

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Real-time Implementation of Multi-channel AMR Speech Coder (멀티채널 AMR 음성부호화기의 실시간 구현)

  • 지덕구;박만호;김형중;윤병식;최송인
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.8
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    • pp.19-23
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    • 2001
  • DSP-based implementation is pervasive in wireless communication parts for systems and handsets according to developing high-speed and low-power programmable Digital Signal Processor (DSP). In this paper, we present a real-time implementation of multi-channel Adaptive Multi-rate (AMR) speech coder. The real-time implementation of an AMR algorithm is achieved using 32-bit fixed-point TMS320C6202 DSP chip that operates at 250 MHz. We performed cross compile, linear assembly optimization and TMS320C62xx assembly optimization for real-time implementation. Furthermore, speech data input/output function and communication function with external CPU is included in an AMR speech coder. The AMR Speech coder developed using DSP EVM board was evaluated in ETRI IMT-2000 Test-bed system.

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A new Implementation of Perceptual LPC Cepstrum and its Application to Speech Recognition (인지 LPC cepstrum의 새로운 구현 및 음성인식에의 적용)

  • Kim, Jin-Young;Choi, Seong-Ho
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.5
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    • pp.61-64
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    • 1996
  • To improve the performance of a recognition system, namely the recognition rate, we propose a hew implementation of perceptual distance using LPC cepstrum(perceptual cepstrum, PLC). The PLC is caculated by convolution of a usual LPC cepstrum and a perceptual lifter(PL). To caculate PL, we define a new weighting function in the linear frequency domain considering the frequency scale(Bark-scale) characteristics. The PL is the inverse Fourier transform of the exponents of the weighting function. We verified our method through the speech recognition experiments. The performance of PLC was compared with that of the rasied sine liftering method.

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