• Title/Summary/Keyword: acoustical comparison

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Computation of Zwicker's loudness and design optimization with Pad$\acute{e}$ approximation (Pad$\acute{e}$ 근사법을 이용한 Zwicker 라우드니스의 계산과 최적화)

  • Kook, Jung-Hwan;Jensen, Jakob S.;Wang, Se-Myung
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2011.10a
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    • pp.279-284
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    • 2011
  • The calculation of Zwicker's loudness which is needed for multiple frequency response with a fine frequency resolution using the finite element (FE) procedure usually requires significant computation time since a numerical solution must be obtained for each considered frequency. Furthermore, if the analysis is the basis for an iterative optimization procedure this approach imposes high computational cost. In this work, we present an efficient approach for obtaining Zwicker's loudness via the Pad$\acute{e}$ approximants and applying in an acoustical topology optimization procedure. The paper is focused on an efficient and accurate calculation of Zwicker's loudness, design sensitivity analysis, and the acoustical topology optimization method by using Pad$\acute{e}$ approximants. The paper compares the efficient algorithm to results obtained by a standard FEM. Comparison are made both in terms of accuracy and in terms of CPU-times needed for the calculation.

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A simple formula for insertion loss prediction of large acoustical enclosures using statistical energy analysis method

  • Kim, Hyun-Sil;Kim, Jae-Seung;Lee, Seong-Hyun;Seo, Yun-Ho
    • International Journal of Naval Architecture and Ocean Engineering
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    • v.6 no.4
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    • pp.894-903
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    • 2014
  • Insertion loss prediction of large acoustical enclosures using Statistical Energy Analysis (SEA) method is presented. The SEA model consists of three elements: sound field inside the enclosure, vibration energy of the enclosure panel, and sound field outside the enclosure. It is assumed that the space surrounding the enclosure is sufficiently large so that there is no energy flow from the outside to the wall panel or to air cavity inside the enclosure. The comparison of the predicted insertion loss to the measured data for typical large acoustical enclosures shows good agreements. It is found that if the critical frequency of the wall panel falls above the frequency region of interest, insertion loss is dominated by the sound transmission loss of the wall panel and averaged sound absorption coefficient inside the enclosure. However, if the critical frequency of the wall panel falls into the frequency region of interest, acoustic power from the sound radiation by the wall panel must be added to the acoustic power from transmission through the panel.

Prediction of acoustic radiation efficiency for trapezoidal corrugated plates (사다리꼴 주름평판의 음향방사효율 해석)

  • Jungsoo Ryue;Seungho Jang
    • The Journal of the Acoustical Society of Korea
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    • v.42 no.2
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    • pp.83-93
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    • 2023
  • Trapezoidal corrugated plates are often treated as orthotropic plates to simplify the modelling of the corrugation. However, this simplification is not valid at high frequencies in which the localized vibration within the parts of corrugation takes place. In this study, the vibrational and acoustical characteristics of corrugated plates are investigated up to high frequencies by means of the wavenumber domain numerical approach. Based on the findings from this numerical analysis, an approximate method to predict vibro-acoustic characteristics of corrugated plates is proposed. This approximate model consists of four equivalent plates which can represent global and local behaviours of corrugated plates. The radiation efficiency of corrugated plates is predicted from the approximate model and validated through the comparison with those of the numerical method.

Fast LBG Algorithm to Reduce the Computational Complexity

  • Kim Dong-Hyun;Kang Chul-Ho
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.4E
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    • pp.123-127
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    • 2005
  • In this paper, we propose a new method for reducing the number of distance calculations in the LBG (Linde, Buzo, Gray) algorithm, which is widely used method to construct a codebook in vector quantization of speech recognition system. The proposed algorithm can reduce the distance calculation between input vector and codeword by utilizing the observation that codewords are quickly stabilized as the number of iteration increases. From the simulation results, it is shown that we can reduce the running times over $43.77\%$ on average in comparison with current LBG algorithm without sacrificing the performance of codebook.

Music Similarity Search Based on Music Emotion Classification

  • Kim, Hyoung-Gook;Kim, Jang-Heon
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.3E
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    • pp.69-73
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    • 2007
  • This paper presents an efficient algorithm to retrieve similar music files from a large archive of digital music database. Users are able to navigate and discover new music files which sound similar to a given query music file by searching for the archive. Since most of the methods for finding similar music files from a large database requires on computing the distance between a given query music file and every music file in the database, they are very time-consuming procedures. By measuring the acoustic distance between the pre-classified music files with the same type of emotion, the proposed method significantly speeds up the search process and increases the precision in comparison with the brute-force method.

Speech Recognition in Car Noise Environments Using Multiple Models Based on a Hybrid Method of Spectral Subtraction and Residual Noise Masking

  • Song, Myung-Gyu;Jung, Hoi-In;Shim, Kab-Jong;Kim, Hyung-Soon
    • The Journal of the Acoustical Society of Korea
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    • v.18 no.3E
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    • pp.3-8
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    • 1999
  • In speech recognition for real-world applications, the performance degradation due to the mismatch introduced between training and testing environments should be overcome. In this paper, to reduce this mismatch, we provide a hybrid method of spectral subtraction and residual noise masking. We also employ multiple model approach to obtain improved robustness over various noise environments. In this approach, multiple model sets are made according to several noise masking levels and then a model set appropriate for the estimated noise level is selected automatically in recognition phase. According to speaker independent isolated word recognition experiments in car noise environments, the proposed method using model sets with only two masking levels reduced average word error rate by 60% in comparison with spectral subtraction method.

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On a Performance Comparison of Pitch Search Algorithms by using a Correlation Properties for the CELP Vocoder (CELP 보코더의 피치 검색시간 단축법의 비교)

  • 배명진
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1993.06a
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    • pp.280-287
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    • 1993
  • Code Excited Linear Prediction(CELP) speech coders exhibit good performance at data rates as low as 4800bps. The major drawback to CELP type paper, a comparative performance study of three pitch searching algorithms for the CELP vocoder was conducted. For each of the algorithms, a standard pitch searching algorithm was used by the sequential pitch searching algorithm that was implimented in the QCELP vocoder. The algorithms used in this study were 1) using the skip table(TABLE), 2) using the symmetrical property of the autocorrelation(SYMMT), and 3) using the preprocessing autocorrelation(PREPC). Performance scores are presented for each of the three pitch searching algorithms based on computation speed and on pitch prediction error.

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A Comparison of Discrete and Continuous Hidden Markov Models for Korean Digit Recognition (한국어 숫자음 인식을 위한 이산분포 HMM과 연속분포 HMM의 성능 비교 연구)

  • 홍형진
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1994.06c
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    • pp.157-160
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    • 1994
  • 본 논문에서는 한국어 숫자음 인식에 대한 이산분포 HMM과 연속분포 HMM의 인식 성능을 비교하였다. 일반적으로 연속분포 HMM은 많은 계산량이 필요하고, 학습시 초기값이 매우 민감하다는 단점이 있지만, 이산분포 HMM의 VQ로 인한 왜곡을 제거함으로써 인식률을 향상시킬 수 있다. 여기서는 성능비교를 위해서 mel-cepstrum의 분석차수, 이산분포 HMM의 codebook 크기, 연속분포 HMM의 miture 개수등에 따른 인식성능을 비교하였다. 실험 결과 이산분포 HMM에서는 mel-cepstrum 벡터가 14차이고, codebook 크기가 64일 때 가장 좋은 성능을 나타냈으며, 연속부포 HMM에서는 mel-cepstrum 벡터가 16차이고 miture가 3개일 때 가장 좋은 결과를 얻을 수 있었다. 특히 학습 데이터의 양이 적은 경우에는 연속분포 HMM이 이산분포 HMM보다 더 좋은 인식률을 나타내었다.

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On a Performance Comparison of Pitch Search Algorithms with the Correlation Properties for the CELP Vocoder (상관관계 특성을 이용한 CELP 보코더의 피치검색시간 단축법의 비교)

  • 김대식
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1994.06c
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    • pp.188-194
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    • 1994
  • Code excited linear prediction speech coders exhibit good performance at data rates as low as 4800bps. But the major drawback to CELP type coders is their large computational requirements. Therefore, in this paper a comparative performance study of three pitch searching algorithms for the CELP vocoder was conducted. For each of the algorithms, a standard pitch searching algorithm was used by the full pitch searching algorithm that was implimented in the QCELP vocoder. The algorithms used in this study is to reduce the pitch searching time 1) using the skip table, 2) using the symmetrical property of the autocorrelation , and 3) using the preprocessing autocorrelation, 4) using the positive autocorrelation, 5) using the preliminary pitch. Performance scores are presented for each of the five pitch searching algorithms based on computation speed and on pitch prediction error.

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On Improving the Effects of Varying the Window Length on Speech Energy Computation (음성 에너지계산에서 창함수-길이 변화영향의 개선에 관한 연구)

  • Bae, Myung-Jin;Ann, Sou-Guil
    • The Journal of the Acoustical Society of Korea
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    • v.9 no.2
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    • pp.34-41
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    • 1990
  • The energy parameter is widely used in pre-processing of speech signals, because it represent the phoneme characteristics of well But, the energy parameter is affected by the window length during the extracting. Thus, in this paper, the window length effects are studied in detail, and we proposed a new energy extraction algorithm that reduces the length effects. The energy contours with this algorithm are well representing for the characteristics of speech phonemes. And the computations to implement the algorithm are only required one subtraction, one addition, and two comparison aperation per speech sample.

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