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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
The Journal of the Acoustical Society of Korea
Journal Basic Information
Journal DOI :
The Acoustical Society of Korea
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Volume & Issues
Volume 24, Issue 8 - Nov 2005
Volume 24, Issue 7 - Oct 2005
Volume 24, Issue 6 - Aug 2005
Volume 24, Issue 5 - Jul 2005
Volume 24, Issue 4 - May 2005
Volume 24, Issue 3 - Apr 2005
Volume 24, Issue 2 - Feb 2005
Volume 24, Issue 1 - Jan 2005
Selecting the target year
High Resolution Wideband Local Polynomial Approximation Beamforming for Moving Sources
Park Do-Hyun ; Park Gyu-Tae ; Lee Jung-Hoon ; Lee Su-Hvoung ; Lee Kyun-Kyung ;
The Journal of the Acoustical Society of Korea, volume 24, issue 1, 2005, Pages 1~10
This paper presents a wideband LPA (local polynomial approximation) beamforming algorithm that is appropriate for wideband moving sources. The Proposed wideband LPA algorithm adopts STMV (steered minimum variance) method that utilizes a steered covariance matrix obtained from multiple frequency components in one data snapshot, instead of multiple data snapshots in one frequency bin. The wideband LPA cost function is formed using STMV weight vector. The Proposed algorithm searches for the instantaneous DOA and angular velocity that maximize the wideband LPA cost function. resulting in a higher resolution performance than that of a DS LPA beamforming algorithm. Several simulations using artificial data and sea trial data are used to demonstrate the performance of the Proposed algorithm.
Study on the Design of S/PDIF BC which Can Operate without PLL
Park Ju-Sung ; Kim Suk-Chan ; Kim Kyoung-Soo ;
The Journal of the Acoustical Society of Korea, volume 24, issue 1, 2005, Pages 11~20
In this paper, we deal with the research about a S/PDIF (Sony Philips Digital Interface) receiver which can operate without PLL (Phase Locked Loop) circuits. Although a S/PDIF receiver is used in most audio devices and audio processors in these days. yet there are only few domestic researches about S/PDIF. Currently used commercial DACs (Digital-to-Analog Converters) which can decode S/PDIF signals, have a PLL circuit inside them. The PLL makes it possible to extract clock information from S/PDIF digital signal and to synchronize a clock signal with input signals. But the PLL circuit makes many diffculties in designing the SOC (System On Chips) of VLSIs (Vew Large Scale Integrated Ciruits) because it is an "analog circuit". We proposed a S/PDIF receiver which doesn't have PLL circuits and only has Pure digital circuits. The key idea of the proposed S/PDIF receiver. is to use the ratio between a 16 MHz basic input clock and S/PDIF signals. After having decoded hundreds thousands S/PDIF inputs, it went to prove that a S/PDIF receiver can be designed with pure digital circuits and without any analog circuits such as PLL circuits. We have confidence that the proposed S/PDIF receiver can be used as an IP (Intellectual Property) for the SOC design of the digital circuits
The Effect of Stage Ceiling Height on the Acoustic Characteristics of Concert Halls
Shin Dong-Jae ; Jeon Jin-Yong ; Seo Hyung-Gyoon ;
The Journal of the Acoustical Society of Korea, volume 24, issue 1, 2005, Pages 21~28
In this Paper, the effect of stage ceiling height on the acoustic characteristics of rectangular concert halls are investigated. To find out the acoustic properties of audience area, A simple Boston Symphony Hall(BSH) model which is typically rectangular shaped was applied for computer simulation. A newly built rectangular concert hall with 400 seats was also chosen for a scale model
study and its computer simulation varing the stage ceiling height and the volume. The results show that RT increased as the stage ceiling was lowered and the difference rate of RT by its variance is from -0.09 to -0.06[sec/m].
An Implementation of Automatic Genre Classification System for Korean Traditional Music
Lee Kang-Kyu ; Yoon Won-Jung ; Park Kyu-Sik ;
The Journal of the Acoustical Society of Korea, volume 24, issue 1, 2005, Pages 29~37
This paper proposes an automatic genre classification system for Korean traditional music. The Proposed system accepts and classifies queried input music as one of the six musical genres such as Royal Shrine Music, Classcal Chamber Music, Folk Song, Folk Music, Buddhist Music, Shamanist Music based on music contents. In general, content-based music genre classification consists of two stages - music feature vector extraction and Pattern classification. For feature extraction. the system extracts 58 dimensional feature vectors including spectral centroid, spectral rolloff and spectral flux based on STFT and also the coefficient domain features such as LPC, MFCC, and then these features are further optimized using SFS method. For Pattern or genre classification, k-NN, Gaussian, GMM and SVM algorithms are considered. In addition, the proposed system adopts MFC method to settle down the uncertainty problem of the system performance due to the different query Patterns (or portions). From the experimental results. we verify the successful genre classification performance over
for both the k-NN and SVM classifier, however SVM classifier provides almost three times faster classification performance than the k-NN.
A Design of Lowpass Active Filter for ADLS Tx/Rx Stage
Lee Geun-Ho ;
The Journal of the Acoustical Society of Korea, volume 24, issue 1, 2005, Pages 38~42
CMOS analog lowpass filters using speech signal bandwidth for a Asymmetrical Digital Subscriver Line(ADSL) modem are presented. Designed active lowpass filters are composed of the CMOS complementary high-swing cascode stage which can increase transconductance of an active element. As a result, their cutoff frequency are 138kHz and 1,100kHz respectively. A low-voltage high-swing cascode integrator which improved on a gain and unit gain frequency used to design the filters. The designed filters are verified by HSPICE simulation with the
Parameter and a single 2.5V power supply.
Performance Improvement of Perceptual Filter Using Noise Energy Control
Seo Joung-Kook ; Cha Hyung-Tai ;
The Journal of the Acoustical Society of Korea, volume 24, issue 1, 2005, Pages 43~51
In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a Performance of perceptual filter using noise energy control. Most of the algorithms which were proposed by the other researchers usually applied a filter using the noise energy acquired from a silent range. In this case. the improvement rate of tone quality decreases if the noise energy is changed by the magnitude or environment variation in a signal frame. But the Proposed method Provides the means to find a food estimated noise through energy control of the estimated noise which is obtained from a silent range. Also we can get the enhancement of tone qualify in low frequency band unlike other methods. To show the performance of the Proposed algorithm, various input signals which had a different signal-to-noise ratio (SNR) such as 5dB, l0dB, 15dB and 20dB were used to test the proposed algorithm. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR). noise-to-mask ration (NMR) and mean opinion score (MOS) test.
Real-time Implementation or AMR-WB Speech Coder Using TMS320C5509 DSP
Choi Song-ln ; Jee Deock-Gu ;
The Journal of the Acoustical Society of Korea, volume 24, issue 1, 2005, Pages 52~57
The adaptive multirate wideband (AMR-WB) speech coder has an extended audio bandwidth from 50 Hz to 7 kBz and operates on nine speech coding bit-rates from 6.6 to 23.85 kbit/s. In this Paper, we present the real-time implementation of AMR-WB speech coder using 16bit fixed-point TMS320C5509 that has dual MAC units. Firstly, We implemented AMR-WB speech coder in C 1anguage level using intrinsics, and then performed optimization in assembly language. The computational complexity of the implemented AMR-WB coder at 23.85 kbit/s is 42.9 Mclocks. And this coder needs the program memory of 15.1 kwords, data ROM of 9.2 kwords and data RAM of 13.9 kwords.
Minimum Classification Error Training to Improve Discriminability of PCMM-Based Feature Compensation
Kim Wooil ; Ko Hanseok ;
The Journal of the Acoustical Society of Korea, volume 24, issue 1, 2005, Pages 58~68
In this paper, we propose a scheme to improve discriminative property in the feature compensation method for robust speech recognition under noisy environments. The estimation of noisy speech model used in existing feature compensation methods do not guarantee the computation of posterior probabilities which discriminate reliably among the Gaussian components. Estimation of Posterior probabilities is a crucial step in determining the discriminative factor of the Gaussian models, which in turn determines the intelligibility of the restored speech signals. The proposed scheme employs minimum classification error (MCE) training for estimating the parameters of the noisy speech model. For applying the MCE training, we propose to identify and determine the 'competing components' that are expected to affect the discriminative ability. The proposed method is applied to feature compensation based on parallel combined mixture model (PCMM). The performance is examined over Aurora 2.0 database and over the speech recorded inside a car during real driving conditions. The experimental results show improved recognition performance in both simulated environments and real-life conditions. The result verifies the effectiveness of the proposed scheme for increasing the performance of robust speech recognition systems.