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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
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The Journal of the Acoustical Society of Korea
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Journal DOI :
The Acoustical Society of Korea
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Volume & Issues
Volume 22, Issue 8 - Nov 2003
Volume 22, Issue 7 - Oct 2003
Volume 22, Issue 6 - Aug 2003
Volume 22, Issue 5 - Jul 2003
Volume 22, Issue 4 - May 2003
Volume 22, Issue 3 - Apr 2003
Volume 22, Issue 2 - Feb 2003
Volume 22, Issue 1 - Jan 2003
Volume 22, Issue 1E - 00 2003
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Noise Analyses of VVVF Inverter and DC/DC Converter for Maglev Train
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 337~344
In DC/DC converter and VVVF inverter, which are the must dominant noise sources of Maglev train, noise is radiated from core and coil excited by MS(Magnetostriction). The main noise source of DC/DC converter is transformer whose spectrum shows strong peaks associated with harmonics of exciting frequency, On the other hand, LIM/VVVF noise is dominated by the harmonics of switching frequency, whereas harmonics of exciting frequency are not significant. As switching frequency is increased in VVVF inverter, it is shown that the harmonics are shifted to higher frequency range. If switching frequency is increased from 700㎐ to 2 ㎑, It is measured that noise can be reduced by 5 to 6 ㏈. Since complete mathematical description of MS phenomena is far beyond the present technology, vibration spectrum is investigated qualitatively in this paper, where effect of increasing switching frequency is confirmed.
A Basic Study on the Improvement of Leakage Error of the Acoustic Intensity
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 345~350
Acoustic intensity is usually estimated by the cross-spectrum of acoustic pressure at two adjacent microphones. The cross-spectrum calculated by digital Fourier transform technique will unavoidably have leakage error since the period of signal will not be usually coincident with record length. Therefore, the acoustic intensity estimated by the conventional FFT analyzer will show distorted value. In this paper, the expression of the Fourier transformed data of a harmonic signal with a single frequency is formulated when there is leakage error. The method to eliminate the effect of leakage error from the contaminated data is also proposed. Some numerical examples show the validation of the proposed method.
Acoustic Echo Cancellation Using Independent Component Analysis
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 351~359
In this paper, we proposed a method for acoustic echo cancellation based on independent component analysis. When the large acoustic noise is picked up by the microphone, the performance of echo cancellation decreased. We used two microphones that received echo signal which is linearly mixed with the noise, then separated the echo signals from the received signals with independent component analysis algorithm. The separated echo signal is used for the reference signal of adaptive algorithm which leads to better performance of the echo cancellation. Computer simulation results show the validity of the proposed method.
Improving Speaker Enrolling Speed for Speaker Verification Systems Based on Multilayer Perceptrons by Using a Qualitative Background Speaker Selection
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 360~366
Although multilayer perceptrons (MLPs) present several advantages against other pattern recognition methods, MLP-based speaker verification systems suffer from slow enrollment speed caused by many background speakers to achieve a low verification error. To solve this problem, the quantitative discriminative cohort speakers (QnDCS) method, by introducing the cohort speakers method into the systems, reduced the number of background speakers required to enroll speakers. Although the QnDCS achieved the goal to some extent, the improvement rate for the enrolling speed was still unsatisfactory. To improve the enrolling speed, this paper proposes the qualitative DCS (QlDCS) by introducing a qualitative criterion to select less background speakers. An experiment for both methods is conducted to use the speaker verification system based on MLPs and continuants, and speech database. The results of the experiment show that the proposed QlDCS method enrolls speakers in two times shorter time than the QnDCS does over the online error backpropagation(EBP) method.
An Audio Watermarking Method Using the Attribute of the Tonal Masker
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 367~374
In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.
A Study on Lip-reading Enhancement Using Time-domain Filter
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 375~382
Lip-reading technique based on bimodal is to enhance speech recognition rate in noisy environment. It is most important to detect the correct lip-image. But it is hard to estimate stable performance in dynamic environment, because of many factors to deteriorate Lip-reading's performance. There are illumination change, speaker's pronunciation habit, versatility of lips shape and rotation or size change of lips etc. In this paper, we propose the IIR filtering in time-domain for the stable performance. It is very proper to remove the noise of speech, to enhance performance of recognition by digital filtering in time domain. While the lip-reading technique in whole lip image makes data massive, the Principal Component Analysis of pre-process allows to reduce the data quantify by detection of feature without loss of image information. For the observation performance of speech recognition using only image information, we made an experiment on recognition after choosing 22 words in available car service. We used Hidden Markov Model by speech recognition algorithm to compare this words' recognition performance. As a result, while the recognition rate of lip-reading using PCA is 64%, Time-domain filter applied to lip-reading enhances recognition rate of 72.4%.
A New Method of Selecting Cohort for Speaker Verification
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 383~387
This paper deals with the method of speaker verification based on the conventional cohort of fixed size. In particular, a new cohort of variable size, which makes use of the distance between speaker models, is proposed: The density of neighboring speaker models within the fixed distance from each speaker is taken into account in the proposed method. The high density leads to the increase of cohort size, thus improving the speaker verification rate. On the other hand, the low density leads to its decrease, thus reducing the amount of computations. The simulation results show that the proposed method outperforms the conventional one, achieving a reduction in the EER.
A Study on Phoneme Likely Units to Improve the Performance of Context-dependent Acoustic Models in Speech Recognition
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 388~402
In this paper, we carried out the word, 4 continuous digits. continuous, and task-independent word recognition experiments to verify the effectiveness of the re-defined phoneme-likely units (PLUs) for the phonetic decision tree based HM-Net (Hidden Markov Network) context-dependent (CD) acoustic modeling in Korean appropriately. In case of the 48 PLUs, the phonemes /ㅂ/, /ㄷ/, /ㄱ/ are separated by initial sound, medial vowel, final consonant, and the consonants /ㄹ/, /ㅈ/, /ㅎ/ are also separated by initial sound, final consonant according to the position of syllable, word, and sentence, respectively. In this paper. therefore, we re-define the 39 PLUs by unifying the one phoneme in the separated initial sound, medial vowel, and final consonant of the 48 PLUs to construct the CD acoustic models effectively. Through the experimental results using the re-defined 39 PLUs, in word recognition experiments with the context-independent (CI) acoustic models, the 48 PLUs has an average of 7.06%, higher recognition accuracy than the 39 PLUs used. But in the speaker-independent word recognition experiments with the CD acoustic models, the 39 PLUs has an average of 0.61% better recognition accuracy than the 48 PLUs used. In the 4 continuous digits recognition experiments with the liaison phenomena. the 39 PLUs has also an average of 6.55% higher recognition accuracy. And then, in continuous speech recognition experiments, the 39 PLUs has an average of 15.08% better recognition accuracy than the 48 PLUs used too. Finally, though the 48, 39 PLUs have the lower recognition accuracy, the 39 PLUs has an average of 1.17% higher recognition characteristic than the 48 PLUs used in the task-independent word recognition experiments according to the unknown contextual factor. Through the above experiments, we verified the effectiveness of the re-defined 39 PLUs compared to the 48PLUs to construct the CD acoustic models in this paper.
Rapid Speaker Adaptation Based on Eigenvoice Using Weight Distribution Characteristics
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 403~407
Recently, eigenvoice approach has been widely used for rapid speaker adaptation. However, even in the eigenvoice approach, Performance improvement using very small amount of adaptation data is relatively small in comparison with that using somewhat large adaptation data because the reliable estimation of weights of eigenvoice is difficult. In this paper, we propose a rapid speaker adaptation method based on eigenvoice using the weight distribution characteristics to improve the performance on a small adaptation data. In the Experimental results on vocabulary-independent word recognition task (using PBW 452 database), the weight threshold method alleviates the problem of relatively low performance for a tiny small adaptation data. When single adaptation word is used, word error rate is reduced about 9-18% by the weight threshold method.
The Enhancement of the Acoustic Image by Combining Bases of Support for SFR (Spatial Frequency Response)
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 408~417
In this paper, we have studied the enhancement of the acoustic image by combining bases of support for SFR (Spatial Frequency Response) taken at multi-frequencies. The scanning acoustic microscope system have been constructed using the quadrature detector that is able to measure the amplitude and phase of the reflected signal simultaneously. Both real and quadrature components of reflected signal have been acquired at 4.4 ㎒ to 5.6 ㎒ reliably and accurately. In this experimental result, better depth resolution can be obtained by numerically combining images taken at several different frequencies. Image intensity have been better about 3.4 times at multi-frequency than one at a single frequency.
Experimental Results of an Underwater Acoustic Communications Using BFSK Modulation
The Journal of the Acoustical Society of Korea, volume 22, issue 5, 2003, Pages 418~424
In this paper we analyzed the performance of data transmission using BFSK modulation. The system performances were evaluated by the experiments in water tank. As a result we showed the influences of reverberation due to the multipath. In order to simplify the experiment procedure the channel coding etc. were omitted. The experimental result shows that the maximum transmission data rate in used water tank is about 800 bps. We also verified that the reverberation effect m reduced using a deconvolution with a measured channel impulse response. This method improved the bit rate by about 100 bps than simple noncoherent demodulator at bit error rate of 10/sup -3/.