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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 23, Issue 8 - Nov 2004
Volume 23, Issue 7 - Oct 2004
Volume 23, Issue 6 - Aug 2004
Volume 23, Issue 5 - Jul 2004
Volume 23, Issue 4 - May 2004
Volume 23, Issue 3 - Apr 2004
Volume 23, Issue 2 - Feb 2004
Volume 23, Issue 1 - Jan 2004
Volume 23, Issue 4E - 00 2004
Volume 23, Issue 3E - 00 2004
Volume 23, Issue 2E - 00 2004
Volume 23, Issue 1_E - 00 2004
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Experimental Analysis of the Damper of a Loudspeaker
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 192~196
A decision of the modulus of elasticity is made by using the degree of bending strength of materials for loudspeaker damper and the radius of corrugation lines and the radius of curvature of each corrugation as a geometrical element. And it is compared with experimental measurements. As a result. the elasticity of damper is proportional to the degree of bending strength and inversely proportional to the radius of corrugation lines and inversely proportional to the square of the radius of curvature. We made a small loudspeaker using a modified damper which take the form of inner small curvature and outer large curvature of each corrugation. This loudspeaker have the increased sensitivity in high frequency and also in low frequency region.
Influence of the Shear Property of Seabed Appearing in the Striation Pattern of the Spectrogram of Ship-radiated Noise Measured in a Shallow Sea
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 197~205
This paper represents the results of interpretation on the cause of sign changing of the striation slopes appearing in the range-frequency domain spectrogram of ship-radiated noise measured in a shallow sea. Striation patterns and dispersion characteristics simulated from a numerical model based on mode theory at various seabed conditions show that the sign changing of the striation slopes appearing in measured signal is caused by the shear property of seabed. more specifically by the shear property of the basement lying below the sediment which is estimated about 3±1m thick.
Steering Angle Error Compensation Algorithm Appropriate for Rapidly Moving Sources
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 206~213
This paper presents a steering angle error compensation (SAEC) algorithm that is appropriate for rapidly moving sources. The Proposed algorithm utilizes a modal covariance matrix from multiple frequency components instead of the multiple snapshots in a narrowband SAEC, and estimates the steering error by maximizing the wideband WVDR output power using a first-order Taylor series approximation of the modal steering vector in terms of the steering error. As such, the steering error can be compensated with short observation times. Several simulations using artificial and sea trial data are used to demonstrate the Performance of the proposed algorithm.
Efficient 3-D Near-field Source Localization Algorithm Using Uniform Circular Array
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 214~220
A computationally efficient algorithm is presented for 3-D near-field source localization using a uniform circular away (UCA). Algebraic relations are demonstrated between the incident angles (elevation angle and azimuth angle) under the far-field assumption and the actual near-field location (range. elevation angle, and azimuth angle). Using these relations as paths to follow to the peak of the 3-D MUSIC spectrum, the proposed algorithm replaces the 3-D search required in the conventional 3-D MUSIC with a 1-D path following after a 2-D initialization. thereby reducing the computational burden.
Subband Affine Projection Algorithm
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 221~227
This paper presents the subband affine projection algorithm(SAPA). The improved performance of SAPA is achieved by applying the affine projection algorithm to the subband adaptive structure. In this algorithm, the weight updating formula of adaptive filter is simply derived by using the orthogonal quadrature filter(OQF) as an analysis filter bank for subband filtering. The derived SAPA has the fast convergence speed and small computational complexity. The efficiency of the proposed algorithm for colored input signal is evaluated through some experiments.
A Study on Channel Mis-match Compensation Technique for Robust Speaker Verification System
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 228~234
In this paper, we proposed the compensation technique that overcomes the limitations of the conventional approaches through summing up the bias terms between world's codebook and individual codebook vectors of feature parameters. But, mean compensation without condition can bring higher false acceptance. Therefore, the proposed technique compensates the channel mis-match condition by weighted bias sum using nonlinear function regarding to the distortion between speech and silence. The simulation results show that the FRR (flase reject rate) is decreased 14.95% when the proposed algorithm was applied.
GMM-based Emotion Recognition Using Speech Signal
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 235~241
This paper studied the pattern recognition algorithm and feature parameters for speaker and context independent emotion recognition. In this paper, KNN algorithm was used as the pattern matching technique for comparison, and also VQ and GMM were used for speaker and context independent recognition. The speech parameters used as the feature are pitch. energy, MFCC and their first and second derivatives. Experimental results showed that emotion recognizer using MFCC and its derivatives showed better performance than that using the pitch and energy parameters. For pattern recognition algorithm. GMM-based emotion recognizer was superior to KNN and VQ-based recognizer.
A Study on Keyword Spotting System Using Pseudo N-gram Language Model
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 242~247
Conventional keyword spotting systems use the connected word recognition network consisted by keyword models and filler models in keyword spotting. This is why the system can not construct the language models of word appearance effectively for detecting keywords in large vocabulary continuous speech recognition system with large text data. In this paper to solve this problem, we propose a keyword spotting system using pseudo N-gram language model for detecting key-words and investigate the performance of the system upon the changes of the frequencies of appearances of both keywords and filler models. As the results, when the Unigram probability of keywords and filler models were set to 0.2, 0.8, the experimental results showed that CA (Correctly Accept for In-Vocabulary) and CR (Correctly Reject for Out-Of-Vocabulary) were 91.1% and 91.7% respectively, which means that our proposed system can get 14% of improved average CA-CR performance than conventional methods in ERR (Error Reduction Rate).
Automatic Pronunciation Generator Using Selection Procedure for Exceptional Pronunciation Words
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 248~252
Cultural, social, economic and other various environmental factors affect our language and different words and terminology are used and coined for different contexts, resulting in quantitative change of vocabulary. This paper presents an automatic pronunciation generator using selection procedure for exceptional pronunciation words from added text corpus, which reflects this dynamic nature of language. For our experiment, we used the text corpus released by ETRI for speech recognition. consisting or 53,750 sentences (740.497 Eojols), and obtained a 100% performance level of the proposed automatic pronunciation generator.
Voice Personality Transformation Using a Multiple Response Classification and Regression Tree
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 253~261
In this paper, a new voice personality transformation method is proposed. which modifies speaker-dependent feature variables in the speech signals. The proposed method takes the cepstrum vectors and pitch as the transformation paremeters, which represent vocal tract transfer function and excitation signals, respectively. To transform these parameters, a multiple response classification and regression tree (MR-CART) is employed. MR-CART is the vector extended version of a conventional CART, whose response is given by the vector form. We evaluated the performance of the proposed method by comparing with a previously proposed codebook mapping method. We also quantitatively analyzed the performance of voice transformation and the complexities according to various observations. From the experimental results for 4 speakers, the proposed method objectively outperforms a conventional codebook mapping method. and we also observed that the transformed speech sounds closer to target speech.
Real-time Implementation of AMR-WB Speech Codec Using TeakLite DSP
The Journal of the Acoustical Society of Korea, volume 23, issue 3, 2004, Pages 262~267
AMR-WB (Adaptive Multi Rate Wideband) speech codec, the most recent voice codec standardized by 3GPP, has the wider audio bandwidth of 50∼7000 Hz and operates on nine speech coding bit rates between 6.60 and 23.85 kbit/s. This Paper presents the real-time implementation of AMR-WB speech codec by using a 16 bit fixed-point TeakLite DSP. The implemented AMR-WB codec requires the complexity of 52.2 MIPS at 23.85 kbit/s mode and also needs the program memory of 17.9 kwords, data RAM of 11.8 kwords, and data ROM of 10.1kwords. It was verified through passing the all test vectors provided by 3GPP with maintaining bit exactness. Stable operations on the real-time testing board were also proved without any distortions and delays for the audio in/out.