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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 21, Issue 8 - Nov 2002
Volume 21, Issue 7 - Oct 2002
Volume 21, Issue 6 - Aug 2002
Volume 21, Issue 5 - Jul 2002
Volume 21, Issue 4 - May 2002
Volume 21, Issue 3 - Apr 2002
Volume 21, Issue 2 - Feb 2002
Volume 21, Issue 1 - Jan 2002
Volume 21, Issue 1E - 00 2002
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Internet Audio Broadcasting Technology Using MPEG-2 AAC Streaming
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 93~101
This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.
Optimization of Withdrawal Weighted SAW Ladder Filters
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 102~109
This paper proposes a new design algorithm to optimize the withdrawal weighted SAW ladder devices, Withdrawal weighting on SAW resonators gives enhanced bandwidth and transition band characteristics, so an optimization algorithm was developed for ladder type SAW filters. To illustrate the validity of the technique, this method was applied to the design of RF band CDMA TX SAW filters, and the design results were confirmed to show good agreement with experimental performance.
Acoustic Characteristics of Sand Sediment with Circular Cylindrical Pores in Water
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 110~119
Acoustic characteristics of water sediment were experimentally studied in laboratory. Water saturated sand sediment less than the grain size of 0.5 mm diameter is uniformly distributed in an acryl box (100 mm×100mm×42mm) with material thickness 1 mm. Pores in the acryl box are modeled as the structure of cylindrical pore tubes (diameter 3 mm and length 42 mm) filled with water. Cylindrical pore tubes have porosities 0%, 5%, 11%, 18% and 26 % controlled by the tube numbers. Transmitted acoustic waves through sand sediment specimen are analyzed as the functions of porosity and frequency from 0.3 MHz to 4 MHz. Transmitted acoustic waves are mixed with the first-kind wave from whole specimen and the second-kind wane from cylindrical pore tubes. For the center frequency 1 MHz, the first kind wave is dominant but for the center frequency 2.25 MHz, the second kind wave is dominant. In the case of the first-kind wave, as the porosity increases, the transmission coefficient decreases and the sound speed decreases to the sound speed of water. As the frequency increases, the transmission coefficient decreases but the sound speed is almost constant. In the case of the second-kind wave, as the porosity increases, the transmission coefficient increases but the sound speed is almost constant. The transmission coefficient and the sound speed are almost constant as a function of frequency.
The Performance Analysis of the Pseudo-decorrelator for WCDMA systems
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 120~127
As a new type of a linear decorrelating receiver, the Pseudo-Decorrelator was presented for asynchronous code division multiple access systems by the author. In this paper, the concept of the Pseudo-Decorrelator is extended to derive a multiuser receiver for WCDMA uplink systems over a Rayleigh fading multipath channel. Starting with the analysis of the multiple access components of the decision statistics, a non-square cross-correlation matrix for each bit is obtained. This cross-correlation matrix is then inverted, and the inverted matrix is applied to the decision statistics obtained from a conventional receiver. This receiver is near-far resistant and outperforms conventional receivers even for the cases in which synchronization errors, such as time delay errors and phase errors exist.
Electronic Music Glove Using Sound Card
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 128~133
We developed a electronic music glove (BMG) system that plays musical scores in real time processing. The EMG system interfaces with the signal coming from the controller to the sound card in the computer, The computer, according to the status of the fingers and foot switches, generates the sound signals and sends them to the speaker systems through the application C++ program and MIDI message. The EMG systems control the velocity and duration of sound and several musical performance expressions such as chorus, reverberation, rhythm, and volume. Finally, we implemented the digital drum set using: the EMG system as example.
Wind Turbine Performance and Noise Prediction by Using Free Wake Method
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 134~141
In this paper, a free wake analysis based on the curved vortex element and CVC wake model is used to predict the aerodynamic performance and noise for HAWT. Also for prediction of RPM, a maximum value through a quadratic regression was suggested. And for a noise prediction, the broadband noise prediction method based on experimental equation was used. The curved vortex element uses a BCVE and an SIVE instead of a straight vertex element. In the CVC wake model, the vortex strengths are assumed to be constant along a span and a vortex filament. The free wake structure made by the curved vortex element and CVC was substituted for a vortex lattice, so it has an advantage for the less calculation time and a depiction of accurate wake structure. For the verification of this program, calculated results are compared with Mr. Kim's experiment model and Zond Z-40FS for performance and with WTS-4 and USWP models for noise. Good agreements are obtained between the predicted and the measured data for the performance and far-field noise spectra.
A Study on the Characteristics of Attenuation and Propagation of Railway Noise at Various Ground Conditions
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 142~149
With rapid industrial development, a railway has become a main traffic means. But, the railway noise has caused much annoyance for the residents living nearby railway tracks. Therefore, execute regulation standard of railway noise has established for quite dwelling environment from January 2000. However efficient soundproofing measures are very difficult because of the lack of basic data and insufficiency of the existing research. In this point, this study attempts to survey the characteristics of attenuation and propagation of railway noise. And this study intends to get the basic data for establishment of a standard for domestic railway noise.
Broadband Transmission Noise Reduction Performance of Smart Panels Featuring Piezoelectric Shunt Damping and Passive Characteristics
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 150~159
The possibility of a broadband noise reduction of piezoelectric smart panels is experimentally studied. Piezoelectric smart panel is basically a plate structure on which piezoelectric patch with shunt circuits is mounted and sound absorbing material is bonded on the surface of the structure. Sound absorbing materials can absorb the sound transmitted at mid frequency region effectively while the use of piezoelectric shunt damping can reduce the transmission at resonance frequencies of the panel structure. To be able to tune the piezoelectric shunt circuit, the measured electrical impedance model is adopted. Resonant shunt circuit composed of register and inductor in stories is considered and the circuit parameters are determined based on maximizing the dissipated energy through the circuit. The transmitted noise reduction performance of smart panels is investigated using an acoustic tunnel. The tunnel is a square crosses sectional tunnel and a loud speaker is mounted at one side of the tunnel as a sound source. Panels are mounted in the middle of the tunnel and the transmitted sound pressure across the panels is measured. Noise reduction performance of a double smart panel possessing absorbing material and air gap shows a good result at mid frequency region except the first resonance frequency. By enabling the piezoelectric shunt damping, noise reduction is achieved at the resonance frequency as well. Piezoelectric smart panels incorporating passive method and piezoelectric shunt damping are a promising technology for noise reduction in a broadband frequency.
An Implementation of the Vocabulary Independent Speech Recognition System Using VCCV Unit
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 160~166
In this paper, we implement a new vocabulary-independent speech recognition system that uses CV, VCCV, VC recognition unit. Since these recognition units are extracted in the trowel region of syllable, the segmentation is easy and robust. And in the case of not existing VCCV unit, the units are replaced by combining VC and CV semi-syllable model. Clustering of vowel group and applying combination rule to the substitution model in the case of not existing of VCCV model lead to 5.2% recognition performance improvement from 90.4% (Model A) to 95.6% (Model C) in the first candidate. The recognition results that is 98.8% recognition rate in the second candidate confirm the effectiveness of the proposed method.
Design of EVRC LSP Codebooks with Korean
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 167~172
The EVRC (Enhanced Variable Rate Codec) is currently in service as a speech cosec in digital cellular systems in North America and Korea. In the EVRC, the LSP (Line Spectral Pairs) related to energy distribution of speech signals in the frequency domain are coded by weighted split vector quantization. Considering that the LSP codebooks might be trained with the language of the develop country of the codebooks or English, it is expected that codebooks trained with Korean provide the performance improvements in the communication in Korean. In this paper, the EVRC LSP codebooks are designed with korean adopting the LBG algorithm based vector quantization, and the performance improvement of the vector quantization and the accompanying speech quality improvement are demonstrated by spectral distortion, SNR and SegSNR measurements, respectively.
A Study on a Robust Voice Activity Detector Under the Noise Environment in the G,723.1 Vocoder
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 173~181
Generally the one of serious problems in Voice Activity Detection (VAD) is speech region detection in noise environment. Therefore, this paper propose the new method using energy, lsp varation. As a result of processing time and speech quality of the proposed algorithm, the processing time is reduced due to the accurate detection of inactive period, and there is almot no difference in the subjective quality test. As a result of bit rate, proposed algorithm measures the number of VAD=1 and the result shows predominant reduction of bit rate as SNR of noisy speech is low (about 5∼10 dB).
The Content Based Analysis According to the Composition of the Feature Parameters for the Auditory Data
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 182~189
In this paper, we research the content-based analysis and classification according to the composition of the feature parameters pool for the auditory signals to implement the auditory indexing and searching system. Auditory data is classified to the primitive various auditory types. we described the analysis and feature extraction method for the feature parameters available to the auditory data classification. And we compose the feature parameters pool in the indexing group unit, then compare and analysis the auditory data centering around the including level and indexing criterion into the audio categories. Based on this result, we composed the classification procedure and simulate the auditory data classification.
A Noble Decoding Algorithm Using MLLR Adaptation for Speaker Verification
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 190~198
In general, we have used the Viterbi algorithm of Speech recognition for decoding. But a decoder in speaker verification has to recognize same word of every speaker differently. In this paper, we propose a noble decoding algorithm that could replace the typical Viterbi algorithm for the speaker verification system. We utilize for the proposed algorithm the speaker adaptation algorithms that transform feature vectors into the region of the client' characteristics in the speech recognition. There are many adaptation algorithms, but we take MLLR (Maximum Likelihood Linear Regression) and MAP (Maximum A-Posterior) adaptation algorithms for proposed algorithm. We could achieve improvement of performance about 30% of EER (Equal Error Rate) using proposed algorithm instead of the typical Viterbi algorithm.
A Study on Speech Recognition Using the HM-Net Topology Design Algorithm Based on Decision Tree State-clustering
The Journal of the Acoustical Society of Korea, volume 21, issue 2, 2002, Pages 199~210
In this paper, we carried out the study on speech recognition using the KM-Net topology design algorithm based on decision tree state-clustering to improve the performance of acoustic models in speech recognition. The Korean has many allophonic and grammatical rules compared to other languages, so we investigate the allophonic variations, which defined the Korean phonetics, and construct the phoneme question set for phonetic decision tree. The basic idea of the HM-Net topology design algorithm is that it has the basic structure of SSS (Successive State Splitting) algorithm and split again the states of the context-dependent acoustic models pre-constructed. That is, it have generated. the phonetic decision tree using the phoneme question sets each the state of models, and have iteratively trained the state sequence of the context-dependent acoustic models using the PDT-SSS (Phonetic Decision Tree-based SSS) algorithm. To verify the effectiveness of the above algorithm we carried out the speech recognition experiments for 452 words of center for Korean language Engineering (KLE452) and 200 sentences of air flight reservation task (YNU200). Experimental results show that the recognition accuracy has progressively improved according to the number of states variations after perform the splitting of states in the phoneme, word and continuous speech recognition experiments respectively. Through the experiments, we have got the average 71.5%, 99.2% of the phoneme, word recognition accuracy when the state number is 2,000, respectively and the average 91.6% of the continuous speech recognition accuracy when the state number is 800. Also we haute carried out the word recognition experiments using the HTK (HMM Too1kit) which is performed the state tying, compared to share the parameters of the HM-Net topology design algorithm. In word recognition experiments, the HM-Net topology design algorithm has an average of 4.0% higher recognition accuracy than the context-dependent acoustic models generated by the HTK implying the effectiveness of it.