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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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Development of an SH-SAW Sensor for Protein Measurement
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 1~7
We developed SH-SAW sensors to detect protein molecules in liquid solutions applying a particular antibody thin film on the delay line of transverse SAW devices. The antibody investigated was human-immune-globulin G (HigG) to hold the antigens (anti-HigG) in the protein solution. We fabricated the sensor generating 100 MHz with the piezoelectric single crystal LiTaO₃. We measured the frequency change of the sensor by adding the anti-body concentration on SAM (self assembled monolayer) deposited on the Au layer. The sensor showed stable response to the mass loading effects of the anti-HigG molecules with the sensitivity up to 10.8 ng/ml/Hz at noise level 400 Hz below.
Inverse Estimation of Geoacoustic Parameters in Shallow Water Using tight Bulb Sound Source
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 8~16
An inversion method is presented for the determination of the compressional wave speed, compressional wave attenuation, thickness of the sediment layer and density as a function of depth for a horizontally stratified ocean bottom. An experiment for estimating those properties was conducted in the shallow water of South Sea in Korea. In the experiment, a light bulb implosion and the propagating sound were measured using a VLA (vertical line array). As a method for estimating the geoacoustic properties, a coherent broadband matched field processing combined with Genetic Algorithm was employed. When a time-dependent signal is very short, the Fourier transform results are not accurate, since the frequency components are not locatable in time and the windowed Fourier transform is limited by the length of the window. However, it is possible to do this using the wavelet transform a transform that yields a time-frequency representation of a signal. In this study, this transform is used to identify and extract the acoustic components from multipath time series. The inversion is formulated as an optimization problem which maximizes the cost function defined as a normalized correlation between the measured and modeled signals in the wavelet transform coefficient vector. The experiments and procedures for deploying the light bulbs and the coherent broadband inversion method are described, and the estimated geoacoustic profile in the vicinity of the VLA site is presented.
Implementation of a Real-time Multipath Fading Channel Simulator Using a Hybrid DSP-FPGA Architecture
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 17~23
The mobile radio channel can be simulated as a complex-valued random process with narrow-band spectrum. This paper describes a real-time implementation of that process using a INS320C6414 digital signal processor and XC2VP30 Virtex FPGA. The simulator presented here is not only a comprehensive model of the flat fading but also frequency selective fading mobile channel conditions. To replicate the statistical characteristics of the multipath fading environment with the minimum computational burden, multi-rate techniques are employed to resolve practical problems such as variable sampling rate. The simulator produces accurate and consistent results due to digital implementation. It is very flexible and simple to program for various field conditions in mobile communications with a graphical user interface.
Analysis of the Ocean Acoustic Channel Using M-sequences in Ocean Acoustic Tomography
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 24~29
In ocean acoustic tomography (OAT), the pulse compression techniques using M-sequences are employed in the many studies for investigating the ocean structures. M-sequences can provide the good time and Doppler resolution in the process of demodulation using matched-filter. The signal-to-noise (SNR) performance at the output of receiver may be improved by manipulating received signal, i. e. coherently averaging. The processing time can be significantly reduced by using fast hadarmard transform (FHT) or fast Fourier transform (FFT). In this paper, we estimate the multipath arrival structures and delay times using the East Korean Sea experiment data and explore the compensation method for the detrimental effects on performance due to sampling rate error. We also analyze the characteristics of the ocean acoustic channels through scattering function, delay power profile, and time dispersions.
Modeling of Distance Localization by Using an Extended Auditory Parallax Model
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 30~39
This study aims at establishing a digital signal processing technique to control 3-D sound localization, especially focusing our ores on the role of information provided by Head-Related Transfer Function (HRTF). In order to clarify the cues to control the auditory distance perception, two conventional models named Hirsch-Tahara model and auditory parallax model were examined. As a result, it was shown that both models have limitations to universally explain the auditory distance perception. Hence, the auditory parallax model was extended so as to apply in broader cases of auditory distance perception. The results of the experiment by simulating HRTFs based on the extended parallax model showed that the cues provided by the new model were almost sufficient to control the perception of auditory distance from an actual sound source located within about 2m.
A Study on the Rectangular Distribution of far Field Sources in Equivalent Source Method
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 40~46
The equivalent source method (ESM) uses two groups of equivalent source positions. One group includes the first order images of the sound source inside the enclosure. The positions of the other group are usually on a spherical surface some distance outside the enclosure. A proper selection of the positions for the far field sources could greatly improve the performance of the modeling accuracy and reduce the number of the sources to achieve the required accuracy. This study uses optimally distributed far field source positions on the surface of enlarged version of the rectangular enclosure instead of using typical spherical distribution. Simulations using various sizes of the box shaped distribution are executed and optimal positions are searched using an optimization technique from the larger number of candidate positions. The results of using these far field source positions are compared and analyzed.
Efficient Speech Enhancement based on left-right HMM with State Sequence Decision Using LRT
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 47~53
We propose a new speech enhancement algorithm based on left-right Hidden Markov Model (HMM) with state decision using Log-likelihood Ratio Test (LRT). Since the conventional HMM-based speech enhancement methods try to improve speech quality for all states, they introduce huge computational loads inappropriate to real-time implementation. In the left-right HMM, only the current and the next state are considered for a possible state transition so to reduce the computational complexity. In this paper, we propose a method to decide the current state by using the LRT on the previous state. Experimental results show that the proposed method improves the speed up to 60% with 0.2∼0.4 dB degradation of speech quality compared to the conventional method.
Sound Enhancement of low Sample rate Audio Using LMS in DWT Domain
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 54~60
In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio, current digital audio is always restricted by sampling rate and bandwidth. This restriction normally results in low sample rate audio or calls for the data compression scheme such as MP3. However, they can only reproduce a lower frequency range than a regular CD quality because of the Nyquist sampling theory. Consequently they lose rich spatial information embedded in high frequency. The propose of this paper is to propose efficient high frequency enhancement of low sample rate audio using n adaptive filtering and DWT analysis and synthesis. The proposed algorithm uses the LMS adaptive algorithm to estimate the missing high frequency contents in DWT domain and it then reconstructs the spectrally enhanced audio by using the DWT synthesis procedure. Several experiments with real speech and audio are performed and compared with other algorithm. From the experimental results of spectrogram and sonic test, we confirm that the proposed algorithm outperforms the other algorithm and reasonably works well for the most of audio cases.
A Study on the Channel Normalized Pitch Synchronous Cepstrum for Speaker Recognition
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 61~74
In this paper, a contort- and speaker-dependent cepstrum extraction method and a channel normalization method for minimizing the loss of speaker characteristics in the cepstrum were proposed for a robust speaker recognition system over the channel. The proposed extraction method creates a cepstrum based on the pitch synchronous analysis using the inherent pitch of the speaker. Therefore, the cepstrum called the 〃pitch synchronous cepstrum〃 (PSC) represents the impulse response of the vocal tract more accurately in voiced speech. And the PSC can compensate for channel distortion because the pitch is more robust in a channel environment than the spectrum of speech. And the proposed channel normalization method, the 〃formant-broadened pitch synchronous CMS〃 (FBPSCMS), applies the Formant-Broadened CMS to the PSC and improves the accuracy of the intraframe processing. We compared the text-independent closed-set speaker identification on 56 females and 112 males using TIMIT and NTIMIT database, respectively. The results show that pitch synchronous km improves the error reduction rate by up to 7.7% in comparison with conventional short-time cepstrum and the error rates of the FBPSCMS are more stable and lower than those of pole-filtered CMS.
Gaussian Selection in HMM Speech Recognizer with PTM Model for Efficient Decoding
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 75~81
Gaussian selection (GS) is a popular approach in the continuous density hidden Markov model for fast decoding. It enables fast likelihood computation by reducing the number of Gaussian components calculated. In this paper, we propose a new GS method for the phonetic tied-mixture (PTM) hidden Markov models. The PTM model can represent each state of the same topological location with a shared set of Gaussian mixture components and contort dependent weights. Thus the proposed method imposes constraint on the weights as well as the number of Gaussian components to reduce the computational load. Experimental results show that the proposed method reduces the percentage of Gaussian computation to 16.41%, compared with 20-30% for the conventional GS methods, with little degradation in recognition.
A Study on Real Time Pitch Alteration of Speech Signal
The Journal of the Acoustical Society of Korea, volume 23, issue 1, 2004, Pages 82~89
This paper describes how to reduce the effect of an occupation threshold by that the transform of mixture components of HMM parameters is controlled in hierarchical tree structure to prevent from over-adaptation. To reduce correlations between data elements and to remove elements with less variance, we employ PCA (principal component analysis) and ICA (independent component analysis) that would give as good a representation as possible, and decline the effect of over-adaptation. When we set lower occupation threshold and increase the number of transformation function, ordinary WLLR adaptation algorithm represents lower recognition rate than SI models, whereas the proposed MLLR adaptation algorithm represents the improvement of over 2% for the word recognition rate as compared to performance of SI models.