• KIEAE Journal
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• v.7 no.3
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• pp.63-70
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• 2007

It is known that early reflections, their energy and delay times after the arrival of direct sound are important factors for speech intelligibility. In this basis, acoustical parameters like D50 and C80 had been proposed and are widely used for assessing the listening condition of rooms. These parameters are focused on the fraction of the early energy to the total, regardless of the spatial characteristics of the early reflections. This means that all the early reflections, arrived in certain time boundary. from front, behind, down and upside have the same impact on speech intelligibility. From the questionable simplicity, the influence of the direction of early reflections on speech intelligibility is examined in this study. A computer simulation speech intelligibility test, conducted for 22 university students, found that the reflection of vertical direction with method of the Paired comparison also the preference of 0.746 degree was visible an increase.

• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• Phonetics and Speech Sciences
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• v.2 no.2
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• pp.43-50
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• 2010

English vowels can be divided into monophthongs and diphthongs depending on the number of vocal tract shapes. Diphthongs are usually produced with more than one shape. This study attempts to collect acoustical data of English diphthongs published by Hillenbrand et al.(1995) online and to examine acoustic features of the diphthongs for phoneticians and English teachers. Sixty three American males and females were chosen after excluding those subjects with different target vowels or ambiguous formant tracks. The author used Praat to obtain the acoustical data systematically at eleven equidistant timepoints over the diphthongal segment. Obvious errors were corrected based on the spectrographic display of each diphthong. Results show that the formant trajectories of the diphthongs produced by the American males and females appeared quite similar. When the female formant values were uniformly normalized to those of the males, almost a perfect collapse occurred. Secondly, the diphthongal movements on the vowel space appeared not linear due to the coarticulatory gesture for the following consonant. Thirdly, the average duration of the diphthongs produced by the females was 1.156 times longer than that of the males while the pitch ratio between the two groups turned out to be 1.746 with a similar contour over measurement points. The author concludes that English diphthongs produced by various groups can be compared systematically when the acoustical values are obtained at proportional timepoints. Further studies will be desirable on the comparison of English diphthongs produced by native and nonnative speakers.

• Transactions of the Korean Society of Automotive Engineers
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• v.14 no.5
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• pp.93-99
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• 2006

Flexible couplers are widely used for exhaust transmitted vibration reduction in vehicles. This paper describes an investigation into the acoustical characteristics of exhaust flexible coupler by the simulation and testing. Computational acoustic simulation is carrying out to investigate resonance frequency and transmission loss of decoupler using the boundary element method and transfer matrix approach. To confirm the acoustical simulation results of exhaust decoupler, we compare with measured experimental results by the test of transmission loss measurement system. In the comparison with simulation results and tests results, there is correctly fit the resonance frequency and the trend of transmission loss. Also, we show that the acoustical structure of decoupler is analogous to the expended tube or side branch resonator. The characteristics of exhaust decoupler have a marked increase in the acoustic attenuation at the specified frequency bend. Therefore the decoupler is applied to develop the exhaust system not only for the vibration isolator but also for the noise attenuator.

• The Journal of the Acoustical Society of Korea
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• v.24 no.4
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• pp.194-201
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• 2005

Since the necessity for the accurate measurement in the field of acoustics, vibration, and ultrasound was internationally increased, CCAUV (Consultative Committee for Acoustics , Ultrasound and Vibration) was created by the CIPM (International Committee for Weights and Measures) in 1998. One of the first activities of CCAUV is the key comparison on 1' standard microphones. The key comparison was done from April. 1999 to April. 2001 and 12 NMIs(National Metrology Institutes) including KRISS (Korea Research Institute of Standards and Science) were participated. The pilot laboratory was the National Physical Laboratory(NPL). The traveling standards were two LSIP(Laboratory Standard 1' Pressure Type) microphones specified in IEC 61094-1 and the calibration frequency ranges covers from 63Hz to 8kHz. In this paper, the analyzed results in the final report were summarized.

• The Journal of the Acoustical Society of Korea
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• v.18 no.3
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• pp.35-42
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• 1999

In this paper, a modified clustering algorithm is proposed to improve the discrimination of discrete HMM(Hidden Markov Model), so that it has increased recognition rate of 2.16% in comparison with the original HMM using the K-means or LBG algorithm. And, for preventing the decrease of recognition rate because of insufficient training data at the training scheme of HMM, a modified probabilistic smoothing method is proposed, which has increased recognition rate of 3.07% for the speaker-independent case. In the experiment applied the two proposed algorithms, the average rate of recognition has increased 4.66% for the speaker-independent case in comparison with that of original VQ/HMM.

• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.46-52
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• 2000

Internal waves existing in the stratified ocean significantly affect acoustic propagation. In order to understand the effects of internal waves on acoustic propagation, the sound speed fluctuations due to internal waves are generated based on the Garrett-Munk spectrum which is derived from measured data in the East Sea. The acoustic propagation, where internal waves are present, is simulated numerically using a Galerkin higher order parabolic equation method(SNUPE). These results show favorable comparison to in-situ acoustic propagation data from the East Sea. To investigate the effects of acoustic propagation in random media, scintillation index is adopted and comparison between the measured and numerically simulated data is made.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.457-467
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• 2018

This paper presents performance comparison of several high-resolution frequency estimation algorithms for pure real tone signal. Algorithms are DFT (Discrete Fourier Transform - for reference purpose), Jacobsen, Candan, reassignment and Cedron. Each algorithm is evaluated under various experimental conditions, e.g., different SNR (Signal to Noise Ratio), window function and window length and performance is compared in the perspective of bias, MSE (Mean Square-Error) and variance. Experimental results indicate that Cedron algorithms works well in the most cases. For actual usage in the engineering problem, each algorithm needs additional analysis and modification.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.21 no.9
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• pp.842-849
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• 2011

The human ear canal amplifies the sound pressure level at specific frequency bands. The characteristics of the ear canal are very similar to those of curved cylindrical tube. In this study, the characteristics of sound transfer in human ear canal were measured and the acoustical space of ear canal was reproduced from the canal cavity geometry. For the measurement of sound transfer function in ear canal, a probe microphone and a reference microphone were used. The sound transfer functions were measured for 5 human subjects. To reproduce the acoustical space of the ear canal, two kinds of ear simulator were designed. The first one is a straight cylindrical tube type and the other is a real-shape ear of which geometry was taken from a micro-CT scanning of a human ear. The characteristics of the reproduced apparatus were compared with those of the human and a commercial ear simulator, RA0045 of G.R.A.S. Inc. The comparison results show that the developed apparatus well represent the ear canal characteristics in the low frequency, but have limited coincidence in level over high frequency range.

• The Journal of the Acoustical Society of Korea
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• v.24 no.6
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• pp.334-342
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• 2005

In this Paper. a time-domain geoacoustic inversion was performed using the bulb signals measured during MがU. 04 experiment conducted in the East Sea of Korea in 2004. An obiective function was defined as a direct cross-correlation between the measured and the simulated signals in time domain. The ray theory was used to model the wave propagation in time domain and optimizations were Performed using VFSA (very fast simulated annealing) algorithm. Comparison of inversion results with those from transmission loss matching (an accompanying paper in this issue of the Journal of the Acoustical Society of Korea) shows that Parameters are consistently inverted. Direct time series comparisons between the measured signals and the simulated signals are Presented based on inversion results.