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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
The Journal of the Acoustical Society of Korea
Journal Basic Information
Journal DOI :
The Acoustical Society of Korea
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Volume & Issues
Volume 28, Issue 8 - Nov 2009
Volume 28, Issue 7 - Oct 2009
Volume 28, Issue 6 - Aug 2009
Volume 28, Issue 5 - Jul 2009
Volume 28, Issue 4 - May 2009
Volume 28, Issue 3 - Apr 2009
Volume 28, Issue 2 - Feb 2009
Volume 28, Issue 1 - Jan 2009
Selecting the target year
High Quality Multi-Channel Audio System for Karaoke Using DSP
Kim, Tae-Hoon ; Park, Yang-Su ; Shin, Kyung-Chul ; Park, Jong-In ; Moon, Tae-Jung ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 1~9
This paper deals with the realization of multi-channel live karaoke. In this study, 6-channel MP3 decoding and tempo/key scaling was operated in real time by using the TMS320C6713 DSP, which is 32 bit floating-point DSP made by TI Co. The 6 channel consists of front L/R instrument, rear L/R instrument, melody, and woofer. In case of the 4 channel, rear L/R instrument can be replaced with drum L/R channel. And the final output data is generated as adjusted to a 5.1 channel speaker. The SOLA algorithm was applied for tempo scaling, and key scaling was done with interpolation and decimation in the time domain. Drum channel was excluded in key scaling by separating instruments into drums and non-drums, and in processing SOLA, high-quality tempo scaling was made possible by differentiating SOLA frame size, which was optimized for real-time process. The use of 6 channels allows the composition of various channels, and the multi-channel audio system of this study can be effectively applied at any place where live music is needed.
Modeling of Scattered Signal from Ship Wake and Experimental Verification
Ji, Yoon-Hee ; Lee, Jae-Hoon ; Kim, Jea-Soo ; Kim, Jung-Hae ; Kim, Woo-Shik ; Choi, Sang-Moon ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 10~18
A moving surface vessel generates a ship wake which contains a cloud of micro-bubbles with radii ranging between
. Such micro-bubbles can be detected by active sonar system for more than ten minutes depending on the size and speed of the surface vessel. In this paper, a reverberation model for the ship wake is presented. The developed model consists of the acoustic scattering model due to the distribution of the micro-bubbles and the kinematic model for the moving active sonar. The acoustic scattering model is based on the volume integration, where the volume scattering strengths are obtained from the spatial distribution of micro-bubbles. Since the directivity and look-direction of active sonar are important factors for moving active sonar, the kinematic model utilizes the Euler transformation to obtain the relative motion between the global and local coordinates. In order to verify the developed model, a series of sea experiment was executed in September 2007 to obtain the spatial-temporal distribution of a bubble cloud, and analyzed to be compared with the simulation results.
A Study on the Stack Temperature Profile of a Standing Wave Thermoacoustic Cooler
Paek, In-Su ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 19~24
Investigations of the relation between the stack temperature profile of a standing wave thermoacoustic cooler and the cooling efficiency were performed. Based on the mathematical derivations using the Rott Equation, it was found that the temperature profile along the stack becomes nonlinear as the enthalpy flux passing through the stack increases. It was also found that such nonlinear temperature profiles lower the cooling efficiency. Simulations using a thermoacoustic simulation program called DELTAE showed that the nonlinear temperature profile occurs with a long stack and large cooling load.
Application of Spectral Element Method for the Vibration Analysis of Passive Constrained Layer Damping Beams
Song, Jee-Hun ; Hong, Suk-Yoon ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 25~31
This paper introduces a spectrally formulated element method (SEM) for the beams treated with passive constrained layer damping (PCLD). The viscoelastic core of the beams has a complex modulus that varies with frequency. The SEM is formulated in the frequency domain using dynamic shape functions based on the exact displacement solutions from progressive wave methods, which implicitly account for the frequency-dependent complex modulus of the viscoelastic core. The frequency response function and dynamic responses obtained by the SEM and the conventional finite element method (CFEM) are compared to evaluate the validity and accuracy of the present spectral PCLD beam element model. The spectral PCLD beam element model is found to provide very reliable results when compared with the conventional finite element model.
Performance Improvement of Cardiac Disorder Classification Based on Automatic Segmentation and Extreme Learning Machine
Kwak, Chul ; Kwon, Oh-Wook ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 32~43
In this paper, we improve the performance of cardiac disorder classification by continuous heart sound signals using automatic segmentation and extreme learning machine (ELM). The accuracy of the conventional cardiac disorder classification systems degrades because murmurs and click sounds contained in the abnormal heart sound signals cause incorrect or missing starting points of the first (S1) and the second heart pulses (S2) in the automatic segmentation stage, In order to reduce the performance degradation due to segmentation errors, we find the positions of the S1 and S2 pulses, modify them using the time difference of S1 or S2, and extract a single period of heart sound signals. We then obtain a feature vector consisting of the mel-scaled filter bank energy coefficients and the envelope of uniform-sized sub-segments from the single-period heart sound signals. To classify the heart disorders, we use ELM with a single hidden layer. In cardiac disorder classification experiments with 9 cardiac disorder categories, the proposed method shows the classification accuracy of 81.6% and achieves the highest classification accuracy among ELM, multi-layer perceptron (MLP), support vector machine (SVM), and hidden Markov model (HMM).
American Acoustician Alfred M. Mayer's Acoustical Research
Ku, Ja-Hyon ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 44~50
While American acoustics had been lagged behind European one in the nineteenth century, Alfred Mayer greatly contributed to enhance American experimental acoustics. He inherited experimental skills in collaboration with Koenig in Paris, and got chances to obtain research ability from leading researchers such as Rayleigh during his visit to England. His accomplishments, which brought him fame in Europe, included the creation of the acoustic pyrometer which measured the high temperature by means of the thermal change of the sound velocity, the discovery of mosquito's hearing by selective resonance, the formalization of the duration of the residual sensation of sound, the invention of the topophone which searched for the direction of the sound source, the construction of the apparatus for visualizing the frequency of sound and so on. He not only added new research results to Europe's acoustics but applied acoustics to physical education to help produce the next generation of American acousticians.
Effective Room Equalization Using Warped Common Acoustical Pole and Zero
Lee, Jun-Ho ; Park, Young-Cheol ; Youn, Dae-Hee ; Lee, Seok-Pil ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 51~60
This paper presents a new method of designing room equalization filters using a warped common acoustical pole and zero (WCAPZ) modeling. The proposed method is capable of significantly reducing the order of the equalization filters without sacrificing the filter performance, especially, at low frequencies. Thus, the associated input-output delay is much smaller than the conventional block transform method while its computational complexity is comparable to it. The computational complexity also is still comparable to the conventional room equalization method, since the filter is implemented in the linear frequency domain after the pole-zero dewarping. Simulation results confirm that the use of the proposed algorithm significantly improves the room equalization over a range of low frequencies.
A Method for Measuring Inter-Utterance Similarity Considering Various Linguistic Features
Lee, Yeon-Su ; Shin, Joong-Hwi ; Hong, Gum-Won ; Song, Young-In ; Lee, Do-Gil ; Rim, Hae-Chang ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 61~69
This paper presents an improved method measuring inter-utterance similarity in an example-based dialogue system, which searches the most similar utterance in a dialogue database to generate a response to a given user utterance. Unlike general inter-sentence similarity measures, the inter-utterance similarity measure for example-based dialogue system should consider not only word distribution but also various linguistic features, such as affirmation/negation, tense, modality, sentence type, which affects the natural conversation. However, previous approaches do not sufficiently reflect these features. This paper proposes a new utterance similarity measure by analyzing and reflecting various linguistic features to improve performance in accuracy. Also, by considering substitutability of the features, the proposed method can utilize limited number of examples. Experimental results show that the proposed method achieves 10%p improvement in accuracy compared to the previous method.
Improvement of Speech Intelligibility in Noisy Environments
Yoon, Jae-Yul ; Kim, Jung-Hoe ; Oh, Eun-Mi ; Park, Ho-Chong ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 70~76
In speech communications in noisy environments, speech intelligibility is seriously degraded due to the masking effect of ambient noise. In this paper, a new method to improve speech intelligibility in noisy environments is proposed. Based on the perception theory that the temporal envelope plays a major role in determining intelligibility, the proposed method uses a novel operation that enhances the fluctuation of band-wise temporal envelope and also contains pitch enhancement for improving speech naturalness. In addition, a new subjective evaluation scheme employing binaural listening is proposed in order to measure more reliable performance. The subjective performance measured with the proposed scheme shows that the proposed method improves both intelligibility and naturalness in various environments, whereas a function parameter can control the performance trade-off between intelligibility and naturalness.
Noise-Biased Compensation of Minimum Statistics Method using a Nonlinear Function and A Priori Speech Absence Probability for Speech Enhancement
Lee, Soo-Jeong ; Lee, Gang-Seong ; Kim, Sun-Hyob ;
The Journal of the Acoustical Society of Korea, volume 28, issue 1, 2009, Pages 77~83
This paper proposes a new noise-biased compensation of minimum statistics(MS) method using a nonlinear function and a priori speech absence probability(SAP) for speech enhancement in non-stationary noisy environments. The minimum statistics(MS) method is well known technique for noise power estimation in non-stationary noisy environments. It tends to bias the noise estimate below that of true noise level. The proposed method is combined with an adaptive parameter based on a sigmoid function and a priori speech absence probability (SAP) for biased compensation. Specifically. we apply the adaptive parameter according to the a posteriori SNR. In addition, when the a priori SAP equals unity, the adaptive biased compensation factor separately increases
each frequency bin, and vice versa. We evaluate the estimation of noise power capability in highly non-stationary and various noise environments, the improvement in the segmental signal-to-noise ratio (SNR), and the Itakura-Saito Distortion Measure (ISDM) integrated into a spectral subtraction (SS). The results shows that our proposed method is superior to the conventional MS approach.