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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
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 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
Music Programming Language Composition Using Csound
Yeo Young-Hwan ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 365~370
The present study is purposed to establish a systematic theory for user-friendly approach to the creation of using a programming language using Csound. Csound is a world-wide computer music programming language and a software synthesizer specialized for prominent sound designers developed by Barry Vercoe at the Media Laboratory in M.I.T. The introduction and the main body of this paper suggested as the starting point of creating electronic music and musical sound the time of combination of music with natural sound or sound from specific media from the viewpoint of traditional Western music. and presents a systematic method composed of the principle of the operation of Csound and basic data samples.
Study on Measurements of the In-Plane Vibration Intensity In a Beam With a Damped End
Kim Chang-Yeol ; Kil Hyun-Gwon ; Hong Suk-Yoon ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 371~378
The objective or this paper is to measure the in-plane vibration intensity of a beam with a damped end that means the magnitude and direction of vibration power. Three experimental methods have been implemented to measure the in-plane vibration intensity over the beam. The first method is the accelerometer array method using two accelerometers. The second method is the frequency response function method using the only one accelerometer. The third method is the reference accelerometer method using a fixed reference accelerometer and another moving accelerometer. Those methods have been used to measure the spatial distribution of in-plane vibration intensity over the beam. The results obtained with those methods have been compared with each other. The results have been compared with an input power. It showed that the frequency response function method and the reference accelerometer method as well as the accelerometer array method can be effectively used to measure the in-plane vibration intensity in beams.
Vibration Modeling and Optimal Design of Differential Electromagnetic Transducer for Implantable Middle Ear Hearing Devices using the FEA
Kim Min-Kyu ; Lim Hyung-Gyu ; Han Chan-Ho ; Song Byung-Seop ; Park Il-Yong ; Cho Jin-Ho ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 379~386
Among various kinds of hearing aids which have been developed so far. the conventional air conduction hearing aids have some problems such as the acoustic distortion, an howling effect due to acoustic feedback. Another type of hearing aid. the cochlear implant system can be applied to the profound imparied person. However. it shows the disadvantage that there is no possibility of recovery of the acoustic organ such as ossicle. On the other hand. the implantable middle ear heaving device directly vibratos the ossicular chain and has better sound qualify. good cosmetics for appearance. and wide frequency responses so that it can overcome the defects or the conventional hearing aids. In this paper, a mathematical modeling and a momentum equation derivation of the DET has been performed. For the optimization of the structure dimension generating maximal vibrating force of the DET. the computer simulation using a finite element analysis (FEA) software has been performed. Also. the vibrating transducer has been designed to make the frequency characteristics or the transducer be similar to those of the normal middle ear. Through the experimental results, the measured vibration characteristics of the DET has been evaluated to verify the performance for the application to implantable middle ear hearing devices.
Serial Transmission of Audio Signals for Multi-channel Speaker Systems
Kwon, Oh-Kyun ; Song, Moon-Vin ; Lee, Seung-Won ; Lee, Young-Won ; Chung, Yun-Mo ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 387~394
In this paper, we propose a new transmission technique of audio signals for the serial connection of the speakers of multiple-channel audio systems. Analog audio signals from a multi-channel audio system are converted into digital signals with signal processing steps and transferred to each speaker through a serial line. The signal processing steps contain data compression and packet generation in association with audio signal characteristics. Each speaker gets its corresponding digital audio signals from the transmitted packets and converts the signals into analog audio signals to make sounds with the speaker All the proposed functions in this paper are modeled in VHDL. implemented with FPGA chips, and tested for actual multi-channel audio systems.
High-Band Codec for Bandwidth Scalable Wideband Speech Codec
Kim Youngvo ; Jeong Byounghak ; Son Chang-Yong ; Sung Ho-Sang ; Park Hochong ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 395~401
In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.
Matching Pursuit Estimation and Quantizer Design for Sinusoidal Model-based Coder
Ahn Yeong-Uk ; Jeong Gyu-Hyeok ; Kim Jong-Hak ; Yang Yong-Ho ; Lee In-Sung ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 402~409
In this paper. we propose a coding method using a matching pursuit algorithm in a strongly periodic highband signal. Also. we propose an efficient quantizer for the estimated parameters : spectral magnitude and phase. Based on the error concealment principle and sinusoidal model. the MP algorithm requires the high-precision pitch period estimation. To estimate more accurate pitch period. the refined pitch obtained from lowband speech is used. which increases the efficiency of bit allocation. The spectral magnitude parameters are quantized by the method which is combined with MDCT (Modified Discrete Cosine Transform) and multi-stage structure. The spectral phase quantizer uses the
modular characteristic of phases and the weighted function by spectral magnitudes. To evaluate the efficiency of the proposed method. we applied it to analysis-by-synthesis system. Furthermore we suggest the possibillity of scalable wideband speech codecs based on band-split structure.
A Study on the Mixed Model Approach and Symbol Probability Weighting Function for Maximization of Inter-Speaker Variation
Chin Se-Hoon ; Kang Chul-Ho ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 410~415
Recently, most of the speaker verification systems are based on the pattern recognition approach method. And performance of the pattern-classifier depends on how to classify a variety of speakers' feature parameters. In order to classify feature parameters efficiently and effectively, it is of great importance to enlarge variations between speakers and effectively measure distances between feature parameters. Therefore, this paper would suggest the positively mixed model scheme that can enlarge inter-speaker variation by searching the individual model with world model at the same time. During decision procedure, we can maximize inter-speaker variation by using the proposed mixed model scheme. We also make use of a symbol probability weighting function in this system so as to reduce vector quantization errors by measuring symbol probability derived from the distance rate of between the world codebook and individual codebook. As the result of our experiment using this method, we could halve the Detection Cost Function (DCF) of the system from
Pre-Processing for Performance Enhancement of Speech Recognition in Digital Communication Systems
Seo, Jin-Ho ; Park, Ho-Chong ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 416~422
Speech recognition in digital communication systems has very low performance due to the spectral distortion caused by speech codecs. In this paper, the spectral distortion by speech codecs is analyzed and a pre-processing method which compensates for the spectral distortion is proposed for performance enhancement of speech recognition. Three standard speech codecs. IS-127 EVRC. ITU G.729 CS-ACELP and IS-96 QCELP. are considered for algorithm development and evaluation, and a single method which can be applied commonly to all codecs is developed. The performance of the proposed method is evaluated for three codecs, and by using the speech features extracted from the compensated spectrum. the recognition rate is improved by the maximum of
compared with that using the degraded speech features.
A Microphone Array Beamformer for the Performance Enhancement of Speech Recognizer in Car
Han Chul-Hee ; Kang Hong-Goo ; Hwang Youngsoo ; Youn Dae-Hee ;
The Journal of the Acoustical Society of Korea, volume 24, issue 7, 2005, Pages 423~430
In this paper. a microphone array beamforming algorithm that reduces the signal distortion caused by reverberation and near-field effect in car environment is proposed. When reverberation or near-field effect is present, an optimum beamformer should be constructed with a steering vector consisting of transfer functions between source and microphones, but it is generally difficult to estimate transfer functions on-line without knowledge of the source signal. Instead, a sub-optimal beamforming algorithm that reduces signal distortion is proposed. It is constructed with steering vectors consisting of relative transfer functions between reference sensor and other sensors. In order to evaluate the performance of the proposed algorithm. we had recorded noisy speech database in a car, and performed speech recognition experiments with HMM Toolkit (HTK) released by Cambridge University. The recognition rate of the proposed algorithm was 15 percents higher than that of the conventional far-field beamformers in best case.