• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.142-147
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• 1998

Recently, 8 kb/s CS-ACELP coder of G.729 is atandardized by ITU-T SG15 and it has been reported that the speech quality of G729 is better than or equal to that of 32kb/s ADPCM. However G.729 is the fixed rate speech coder, and it does not consider the property of voice activity in mutual conversation. If we use the voice activity, we can reduce the average bit rate in half without any degradations of the speech quality. In this paper, we propose an efficient variable rate algorithm for G.729. The variable rate algorithm consists of two main subjects, the rate determination algorithm and algorithm, we combine the energy-thresholding method, the phonetic segmentation method by integration of various feature parameters obtained through the analysis procedure, and the variable hangover period method. Through the analysis of noise features, the 1 kb/s sub rate coder is designed for coding the background noise signal. So, we design the 4 kb/s sub rate coder for the unvoiced parts. The performance of the variable rate algorithm is evaluated by the comparison of speed quality and average bit rate with G.729. Subjective quality test is also done by MOS test. Conclusively, it is verified that the proposed variable rate CS-ACELP coder produced the same speech quality as G.729, at the average bit rate of 4.4 kb/s.

• The Journal of the Acoustical Society of Korea
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• v.24 no.6
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• pp.303-308
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• 2005

In this paper, the acoustic properties of the light bulb are presented based on a new light bulb source system of continuously transmitting implosive signal . We describe the results of analysis of bulb signals and comparison with Previous works. The results show that Peak-source-level and Primary resonant frequency are increasing with increasing source depth. This bulb source can be used for the purpose of geoacoustic parameter inversion and source tracking in sha]low water via matched field processing.

• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.108-115
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• 1999

In this study we suggested a digital filter method to enhance car audio sound quality against the sound distortion due to cabin's acoustic characteristics and car driving noises. The digital filters designed were based on the characteristics on car driving noises and cabin acoustic characteristics. Car driving noises were analyzed by two ways; one is an objective method, octave band frequency analysis method. The other is a subjective method; sensory evaluation method, NCB method. On these results, seven sets of modified coefficients of eleven band digital filters were obtained. To find optimum audio sound quality among nine sound samples filtered by designing seven types of digital filters, which were mixed car driving noises at 100km/h, subjective evaluation method was used, paired comparison method; Scheffe' seven point method.

• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.48-53
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• 1999

This paper presents an analytic derivation of the erroneous effect when the sliding-DFT is implemented in a recursive way with the finite-bit approximation of the twiddle factors. The analysis result is obtained in a closed form equation of the noise-to-signal power ratio(NSR) employing the zero-mean white Gaussian signal as the target input of the DFT. The parameters of the wordlength used in representing the twiddle factors and the blocklength of the DFT appear in the NSR explicitly as its function variables. The derivation is based on the error dynamic equation which is derived from the recursive SDFT, and on the analytic exploration of the statistical characteristics of the approximation coefficients treating them as random variables of having spatial distributions. The analytically derived results are verified through the comparison with the data actually measured from the computer simulation experiment.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.315-322
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• 2003

We propose a channel-attentive Mel frequency cepstral coefficient (CAMFCC) extraction method to improve the recognition performance of speech that is partially corrupted in the frequency domain. This method introduces weighting terms both at the filter bank analysis step and at the output probability calculation of decoding step. The weights are obtained for each frequency channel of filter bank such that the more reliable channel is emphasized by a higher weight value. Experimental results on TIDIGITS database corrupted by various frequency-selective noises indicated that the proposed CAMFCC method utilizes the uncorrupted speech information well, improving the recognition performance by 11.2% on average in comparison to a multi-band speech recognition system.

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.35-46
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• 2000

In this paper, we propose a subband acoustic echo canceller using the weighted Overlap-add adaptive filter bank to prevent the decrease of convergence speed in full-band US processing, and make it possible to realize the adaptive filter in block-parallel processing, this paper introduces the weighted overlap-add technique for subband echo canceller. Moreover, we propose a new double-talk detector which employs dedicated filter in addition to the energy comparison method simultaneously. The computer simulation results show that the performance of the proposed subband adaptive echo canceller double-talk detection

• The Journal of the Acoustical Society of Korea
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• v.16 no.6
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• pp.12-18
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• 1997

Beamforming method has a limited spatial resolution because of finite aperture size, so that the estimated source distributions are smoothed within the resolution. Especially for low frequency noise such as mechanical noise, this limitation often diminishes the direct use of beamforming method. In this study, the relation between smoothed beamforming and radiation power distribution of plate has been addressed. By adjustment of aperture size of array, the smoothed beamforming power shows radiation power distribution of plate. Numerical simulations are carried for simply supported plate.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.36-42
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• 1997

This paper purposes the measurement method of vibration intensity in building structure which is a method of measuring the intensity and the flow of vibration energy. We derived basic theory and measuring theory for a simple beam, and comparison of the experimental results with calculated results. As a result, according to the calculated value from acceleration distribution and the measurement result from the method of vibration intensity under the condition except near field of measurement zone. The measured results, show that this method is useful for measuring the vibration energy flow in building structure.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2E
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• pp.60-65
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• 2009

The present study aims to investigate the effects of sound diffusers and absorbers on the sound clarity in classrooms. In order to do this, computer simulations were carried out to find the effective area of treatment which could enhance the sound clarity in the room. Acoustic measurements were undertaken in a lecture room with several conditions changing the surface of walls and ceilings with diffusers and absorbers. Diffusion and absorption treatments were applied to the side walls, rear wall and the ceiling of the classroom. SPL, RT, D50, RASTI were measured at 9 measurement points with one sound source and MLS was used as the sound source signal. The results show that higher sound clarity was obtained when diffusers were applied to rear walls and ceiling rather than side walls. Also, it was confirmed that absorption increased sound clarity more effectively with smaller amount in comparison with diffusers. It was also concluded that the effects of sound diffusers and absorbers on the sound clarity could be obtained distinctly at the rear area of the classroom.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.383-388
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• 2016

We develop two methods to select a constant in the RLS (Recursive Least Squares) with the convex regularization. The RLS with the convex regularization was proposed by Eksioglu and Tanc in order to estimate the sparse acoustic channel. However the algorithm uses the regularization constant which needs the information about the true channel response for the best performance. In this paper, we propose two methods to select the regularization constant which don't need the information about the true channel response. We show that the estimation performance using the proposed methods is comparable with the Eksioglu and Tanc's algorithm.