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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
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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 28, Issue 8 - Nov 2009
Volume 28, Issue 7 - Oct 2009
Volume 28, Issue 6 - Aug 2009
Volume 28, Issue 5 - Jul 2009
Volume 28, Issue 4 - May 2009
Volume 28, Issue 3 - Apr 2009
Volume 28, Issue 2 - Feb 2009
Volume 28, Issue 1 - Jan 2009
Selecting the target year
Seafloor Sediment Classification Using Nakagami Probability Density Function of Acoustic Backscattered Signals
Bok, Tae-Hoon ; Paeng, Dong-Guk ; Park, Yo-Sup ; Kong, Gee-Soo ; Park, Soo-Chul ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 165~173
The physical properties of a seafloor sediment have been used as a basic data for the ocean survey. Conventional methods such as a coring, a drilling, and a grabbing have been used to explore the physical properties but these methods have a number of shortcomings as it is time consuming, expensive and spatially limited. To overcome these limitations, seafloor sediment classification using acoustic signals has been studied actively. In this paper, we obtained the backscattered signal from the seafloor sediment using an echo sounder which is one kind of seafloor topography equipment. Nakagami probability density function of the backscattered signals from the seafloor sediment was computed and a Nakagami parameter was compared with the physical properties of the seafloor sediment. We have confirmed that Nakagami parameter, m is correlated with the physical properties of a seafloor sediment. This study will be utilized as a basic data of the seafloor sediment research.
Derivation of Coherent Reflection Coefficient at Mid and Low Frequency for a Rough Surface
Chu, Young-Min ; Seong, Woo-Jae ; Byun, Sung-Hoon ; Kim, Sea-Moon ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 174~186
When we apply a propagation model to the ocean with boundaries, we can calculate reflected wave using reflection coefficient suggested by Rayleigh assuming the boundaries are flat. But boundaries in ocean such as sea surface and sea bottom have an irregular rough surface. To calculate the reflection loss for an irregular boundary, it is needed to compute the coherent reflection coefficient based on an experimental formula or scattering theory. In this article, we derive the coherent reflection coefficients for a fluid-fluid interface using perturbation theory, Kirchhoff approximation and small-slope approximation respectively. Based on each formula, we can calculate coherent reflection coefficients for a rough sea surface or sea bottom, and then compare them to the Rayleigh reflection coefficient to analyze the reflection loss for a random rough surface. In general, the coherent reflection coefficient based on small-slope approximation has a wide valid region. Comparing it with the coherent reflection coefficients derived from the Kirchhoff approximation and perturbation theory, we discuss a valid region of them.
The Effects of Ocean Surface Bubbles on Sound Wave Transmission
Im, Byun-Kook ; Shim, Tae-Bo ; Kim, Young-Gyu ; Park, Joung-Soo ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 187~197
The bubbles are created by waves, raindrops, water collision, vessels sailing at sea, life activities of various marine organisms in the ocean and other sources. The bubbles affect the intensity and sound speed of acoustic waves in the ocean. We indirectly observed bubbles in order to understand the creation of and the effects of bubbles on sound waves, using an Acoustic Bubble Spectrometer (ABS) and CTD, from 04:00 to 17:00, 19 September, 2007. We also analyzed the correlation of wind speed and the generation of bubbles, the amount of bubbles, and the sound speed variation at 50, 60, and 70 kHz. Finally, We simulated the way how bubbles affect sound transmission based on the analysis results.
Estimation of Cavitation Bubble Distribution Using Multi-Frequency Acoustic Signals
Kim, Dae-Uk ; La, Hyoung-Sul ; Choi, Jee-Woong ; Na, Jung-Yul ; Kang, Don-Hyug ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 198~207
Distribution of cavitation bubbles relative to change of the sound speed and attenuation in the water was estimated using acoustic signal from 20 to 300 kHz in two cases that cavitation bubbles exist and do not exist. To study generation and extinction property of cavitation bubble, bubble distribution was estimated in three cases: change of rotation speed (3000-4000 rpm), surface area of blade (
) and elapsed time (30-120 sec). As a result, the radii of the generated bubbles ranged from 10 to
, and bubble radius of
was accounted for 45 and 25% of the total number of cavitation bubbles, respectively. And generation bubble population correlated closely with the rotating speed of the blades but did not correlate with the surface area of blade. It was observed that 80% of total bubble population disappeared within 2 minutes. Finally, acoustic data of bubble distribution was compared with optical data.
Interpretation of Ground Wave Using Ray Method in Pekeris Waveguide
Choi, Jee-Woong ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 208~212
Ground wave is an acoustic wave propagating at a sediment sound speed in the case that sediment sound speed is constant with depth, which is explained by modal dispersion effects. In this paper, the ground wave in time domain is simulated using the ray-based approach, which is possible because the modal dispersion can be explained by the guiding of energy caused by reflection and refraction in the waveguide geometry. For a Pekeris waveguide, the ground wave can be interpreted as a sequence of head waves, called a head wave sequence [Choi and Dahl, J. Acoust. Soc. Am. 119, 3660-3668 (2006)]. The ground wave is simulated by convolution of the source signal with a channel impulse response of the head wave sequence, which is compared with simulated signals obtained via a Fourier synthesis of a complex parabolic equation (PE) field.
Analysis of Time Reversal Transmission Performance for Underwater Communications
Kim, Hyeon-Su ; Kwon, Yang-Soo ; Lee, Il-Shin ; Chung, Jae-Hak ; Kim, Seong-Il ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 213~221
The time reversal mirror (TRM) method for underwater communications has been developed to improve transmission performance with low complexity. However, digital communication parameters for TRM have not been researched deeply. This paper demonstrates that the TRM scheme obtains spatial diversity gain similar to multiple antennas, and proposes design methodologies of symbol interval, frame duration and transmission protocol for time reversal mirror transmission. Simulation results show that spatial diversity gain is achieved and the effect of ISI decreases as the number of transducer increases.
Comparison of Sound Spectrums of Pyeonjong Remains at the King Sejong Memorial Museum and Pyeonjong Replica
Yoo, June-Hee ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 222~228
This study compared the sound spectrums of pyeonjong remains and pyeonjong replica to investigate tuning ways of bells. pyeonjong remains, exhibited at King Sejong Memorial Museum and pyeonjong replica, played at the National Center for Korean Traditional Performance Arts were analyzed. To get mode frequencies and mode shapes, pyeonjong replicas' sound spectrums were analyzed and modal analyses by TV holography were performed. Also pyeonjong remains' sound spectrum were analyzed. Nominal frequencies on the pyeonjong replica and remains showed differences in a range between 9.8 cent and 203 cent. Two facts were inferred as causes of the differences, the tuning conditions of pyeonjong remains were not good and C4 in western tempered scale was preferred as the sound standard of Kukak, whangjong. Relative ratio of higher mode frequencies to the nominal frequencies were calculated to figure out tonal differences between two pyeonjongs. The differences in relative ratio of higher mode frequencies except (3,0)a and (3,0)b modes were significants as well as beyond the just noticeable difference. These results implied that the tonal differences between two pyeonjongs could exist. More pyeonjong remains are needed to be investigated to confirm this result in addition to the analyses of alloy components and bell structure of pyeonjong remains and replica.
Sound Engine for Korean Traditional Instruments Using General Purpose Digital Signal Processor
Kang, Myeong-Su ; Cho, Sang-Jin ; Kwon, Sun-Deok ; Chong, Ui-Pil ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 229~238
This paper describes a sound engine of Korean traditional instruments, which are the Gayageum and Taepyeongso, by using a TMS320F2812. The Gayageum and Taepyeongso models based on commuted waveguide synthesis (CWS) are required to synthesize each sound. There is an instrument selection button to choose one of instruments in the proposed sound engine, and thus a corresponding sound is produced by the relative model at every certain time. Every synthesized sound sample is transmitted to a DAC (TLV5638) using SPI communication, and it is played through a speaker via an audio interface. The length of the delay line determines a fundamental frequency of a desired sound. In order to determine the length of the delay line, it is needed that the time for synthesizing a sound sample should be checked by using a GPIO. It takes
for the Gayageum and
for the Taepyeongso, respectively. It happens that each sound sample is synthesized and transferred to the DAC in an interrupt service routine (ISR) of the proposed sound engine. A timer of the TMS320F2812 has four events for generating interrupts. In this paper, the interrupt is happened by using the period matching event of it, and the ISR is called whenever the interrupt happens,
. Compared to original sounds with their spectra, the results are good enough to represent timbres of instruments except 'Mu, Hwang, Tae, Joong' of the Taepyeongso. Moreover, only one sound is produced when playing the Taepyeongso and it takes
for the real-time playing. In the case of the Gayageum, players usually use their two fingers (thumb and middle finger or thumb and index finger), so it takes
for the real-time playing.
A Study of Development for Korean Phonotactic Probability Calculator
Lee, Chan-Jong ; Lee, Hyun-Bok ; Choi, Hun-Young ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 239~244
This paper is to develop the Korean Phonotactic Probability Calculator (KPPC) that anticipates the phonotactic probability in Korean. KPPC calculates the positional segment frequecncy, position-specific biphone frequency and position-specific triphone frequency. And KPPC also calculates the Neighborhood Density that is the number of words that sound similar to a target word. The Phonotactic Calculator that was developed in University of Kansas can be analyzed by the computer-readable phonemic transcription. This can calculate positional frequency and position-specific biphone frequency that were derived from 20,000 dictionary words. But KPPC calculates positional frequency, positional biphone frequency, positional triphone frequency and neighborhood density. KPPC can calculate by korean alphabet or computer-readable phonemic transcription. This KPPC can anticipate high phonotactic probability, low phonotactic probability, high neighborhood density and low neighborhood density.
Variable Step Size LMS Algorithm Using the Error Difference
Woo, Hong-Chae ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 245~250
In communications and signal processing area, a number of least mean square adaptive algorithms have been used because of simplicity and robustness. However the LMS algorithm is known to have slow and non-uniform convergence. Various variable step size LMS adaptive algorithms have been introduced and researched to speed up the convergence rate. A variable step size LMS algorithm using the error difference for updating the step size is proposed. Compared with other algorithms, simulation results show that the proposed LMS algorithm has a fast convergence. The theoretical performance of the proposed algorithm is also analyzed for the steady state.
Discrimination Between Natural and Artificial Seismic Sounds by Using 20 MSVQ Algorithm
Yoon, Sang-Hoon ; Song, Young-Hwan ; Bae, Myung-Jin ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 251~259
This paper proposes an identification technique to discriminate natural and artificial seismic sounds by using the 20 MSVQ algorithm with the data measured by using a hydrophone. Spectrum band energy and MFCC were used as representative parameters for sake of discriminating natural and artificial seismic sounds, and the orders of characterized parameters were determined through experiments. As a result of using 20 MSVQ algorithm with the 2 characterized parameters, MFCC had 99.9% and the spectrum energy parameter had 83.9% percent of success. It was verified that it is extremely accurate when seismic sounds were discriminated by using the method suggested by this paper.
Design of Acoustic Source Array Using the Concept of Holography Based on the Inverse Boundary Element Method
Cho, Wan-Ho ; Ih, Jeong-Guon ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 260~267
It is very difficult to form a desired complex sound field at a designated region precisely as an application of acoustic arrays, which is one of important objects of array systems. To solve the problem, a filter design method was suggested, which employed the concept of an inverse method using the acoustical holography based on the boundary element method. In the acoustical holography used for the source identification, the measured field data are employed to reconstruct the vibro-acoustic parameters on the source surface. In the analogous problem of source array design, the desired field data at some specific points in the sound field was set as constraints and the volume velocity at the surface points of the source plane became the source signal to satisfy the desired sound field. In the filter design, the constraints for the desired sound field are set, first. The array source and given space are modelled by the boundary elements. Then, the desired source parameters are inversely calculated in a way similar to the holographic source identification method. As a test example, a target field comprised of a quiet region and a plane wave propagation region was simultaneously realized by using the array with 16 loudspeakers.
Performance Enhancement for Speaker Verification Using Incremental Robust Adaptation in GMM
Kim, Eun-Young ; Seo, Chang-Woo ; Lim, Yong-Hwan ; Jeon, Seong-Chae ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 268~272
In this paper, we propose a Gaussian Mixture Model (GMM) based incremental robust adaptation with a forgetting factor for the speaker verification. Speaker recognition system uses a speaker model adaptation method with small amounts of data in order to obtain a good performance. However, a conventional adaptation method has vulnerable to the outlier from the irregular utterance variations and the presence noise, which results in inaccurate speaker model. As time goes by, a rate in which new data are adapted to a model is reduced. The proposed algorithm uses an incremental robust adaptation in order to reduce effect of outlier and use forgetting factor in order to maintain adaptive rate of new data on GMM based speaker model. The incremental robust adaptation uses a method which registers small amount of data in a speaker recognition model and adapts a model to new data to be tested. Experimental results from the data set gathered over seven months show that the proposed algorithm is robust against outliers and maintains adaptive rate of new data.
Speech and Music Discrimination Using Spectral Transition Rate
Yang, Kyong-Chul ; Bang, Yong-Chan ; Cho, Sun-Ho ; Yook, Dong-Suk ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 273~278
In this paper, we propose the spectral transition rate (STR) as a novel feature for speech and music discrimination (SMD). We observed that the spectral peaks of speech signal are gradually changing due to coarticulation effect. However, the sound of musical instruments in general tends to keep the peak frequencies and energies unchanged for relatively long period of time compared to speech. The STR of speech is much higher than that of music. The experimental results show that the STR based SMD method outperforms a conventional method. Especially, the STR based SMD gives relatively fast output without any performance degradation.
Noise Rabust Speaker Verification Using Sub-Band Weighting
Kim, Sung-Tak ; Ji, Mi-Kyong ; Kim, Hoi-Rin ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 279~284
Speaker verification determines whether the claimed speaker is accepted based on the score of the test utterance. In recent years, methods based on Gaussian mixture models and universal background model have been the dominant approaches for text-independent speaker verification. These speaker verification systems based on these methods provide very good performance under laboratory conditions. However, in real situations, the performance of speaker verification system is degraded dramatically. For overcoming this performance degradation, the feature recombination method was proposed, but this method had a drawback that whole sub-band feature vectors are used to compute the likelihood scores. To deal with this drawback, a modified feature recombination method which can use each sub-band likelihood score independently was proposed in our previous research. In this paper, we propose a sub-band weighting method based on sub-band signal-to-noise ratio which is combined with previously proposed modified feature recombination. This proposed method reduces errors by 28% compared with the conventional feature recombination method.
Double-Talk Detection Based on Soft Decision for Acoustic Echo Suppression
Park, Yun-Sik ; Chang, Joon-Hyuk ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 285~289
In this paper, we propose a novel double-talk detection (DTD) technique based on soft decision in the frequency domain. In the proposed method, global near-end speech presence probability (GNSPP) considering the statistical model assumption and voice activity detection (VAD) decision of the near-end and far-end signal are applied to the DTD algorithm in the frequency domain instead of the traditional hard decision scheme using cross-correlation coefficients. The performance of the proposed algorithm is evaluated by the objective test under various environments, and yields better results compared with the conventional scheme.
Efficient TTS Database Compression Based on AMR-WB Speech Coder
Lim, jong-Wook ; Kim, Ki-Chul ; Kim, Kyeong-Sun ; Lee, Hang-Seop ; Park, Hae-Young ; Kim, Moo-Young ;
The Journal of the Acoustical Society of Korea, volume 28, issue 3, 2009, Pages 290~297
This paper presents an improved adaptive multi-rate wideband (AMR-WB) algorithm for the efficient Text-To-Speech (TTS) database compression. The proposed algorithm includes unnecessary common bit-stream (CBS) removal and parameter delta coding combined with speaker-dependent huffman coding to reduce the required bit-rate without any quality degradation. We also propose lossy coding schemes to produce the maximum bit-rate reduction with negligible quality degradation. The proposed lossless algorithm including CBS removal can reduce bit-rate by 12.40% without quality degradation compared with the 12.65 kbps AMR-WB mode. The proposed lossy algorithm can reduce bit-rate by 20.00% with 0.12 PESQ degradation.