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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 32, Issue 6 - Nov 2013
Volume 32, Issue 5 - Sep 2013
Volume 32, Issue 4 - Jul 2013
Volume 32, Issue 3 - May 2013
Volume 32, Issue 2 - Mar 2013
Volume 32, Issue 1 - Jan 2013
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Target Signal Simulation in Synthetic Underwater Environment for Performance Analysis of Monostatic Active Sonar
Kim, Sunhyo ; You, Seung-Ki ; Choi, Jee Woong ; Kang, Donhyug ; Park, Joung Soo ; Lee, Dong Joon ; Park, Kyeongju ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 455~471
DOI : 10.7776/ASK.2013.32.6.455
Active sonar has been commonly used to detect targets existing in the shallow water. When a signal is transmitted and returned back from a target, it has been distorted by various properties of acoustic channel such as multipath arrivals, scattering from rough sea surface and ocean bottom, and refraction by sound speed structure, which makes target detection difficult. It is therefore necessary to consider these channel properties in the target signal simulation in operational performance system of active sonar. In this paper, a monostatic active sonar system is considered, and the target echo, reverberation, and ambient noise are individually simulated as a function of time, and finally summed to simulate a total received signal. A 3-dimensional highlight model, which can reflect the target features including the shape, position, and azimuthal and elevation angles, has been applied to each multipath pair between source and target to simulate the target echo signal. The results are finally compared to those obtained by the algorithm in which only direct path is considered in target signal simulation.
An Algorithm for Submarine Passive Sonar Simulator
Jung, Young-Cheol ; Kim, Byoung-Uk ; An, Sang-Kyum ; Seong, Woo-Jae ; Lee, Keun-Hwa ; Hahn, Joo-Young ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 472~483
DOI : 10.7776/ASK.2013.32.6.472
Actual maritime exercise for improving the capability of submarine sonar operator leads to a lot of cost and constraints. Sonar simulator maximizes the capability of sonar operator and training effect by solving these problems and simulating a realistic battlefield environment. In this study, a passive sonar simulator algorithm is suggested, where the simulator is divided into three modules: maneuvering module, noise source module, and sound propagation module. Maneuvering module is implemented in three-dimensional coordinate system and time interval is set as the rate of vessel changing course. Noise source module consists of target noise, ocean ambient noise, and self noise. Target noise is divided into modulated/unmodulated and narrowband/broadband signals as their frequency characteristics, and they are applied to ship radiated noise level depending on the vessel tonnage and velocity. Ocean ambient noise is simulated depending on the wind noise considering the waveguide effect and other ambient noise. Self noise is also simulated for flow noise and insertion loss of sonar-dome. The sound propagation module is based on ray propagation, where summation of amplitude, phase, and time delay for each eigen-ray is multiplied by target noise in the frequency domain. Finally, simulated results based on various scenarios are in good agreement with generated noise in the real ocean.
Design of a Multimode Type Ring Vector Sensor
Lim, Youngsub ; Joh, Cheeyoung ; Seo, Heeseon ; Roh, Yongrae ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 484~493
DOI : 10.7776/ASK.2013.32.6.484
Typical underwater acoustic sensors can measure the scalar quantity of sound-pressure-magnitude with the limitation of being unable to identify the direction of an incoming wave. This paper proposes a method to detect the direction of the sound wave with a ring sensor. The sensor of the proposed structure has a piezoceramic ring divided into eight elements, and distinguishes the direction of the sound wave by properly combining the output voltages of the piezoceramic elements. Further, through the analysis of the effects of the structural parameters like the ring radius and length, and piezoceramic thickness, we have suggested the way to improve the sensitivity of the vector sensor.
Performance of Convolution Coding Underwater Acoustic Communication System on Frequency Selectivity Index
Seo, Chulwon ; Park, Jihyun ; Park, Kyu-Chil ; Shin, Jungchae ; Jung, Jin Woo ; Yoon, Jong Rak ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 494~501
DOI : 10.7776/ASK.2013.32.6.494
The convolution code(CC) of code rate 1/2 as a forward error correction (FEC) in Quadrature Phase Shift Keying (QPSK) is applied to decrease bit error rate (BER) by background noise and multipath in shallow water acoustic channel. Ratio of transmitting signal bandwidth to channel coherence bandwidth is defined as frequency selectivity index. BER and bit energy-to-noise ratio gain of transmitted signal according to frequency selectivity index are evaluated. In the results of indoor water tank experiment, BER is well matched theoretical results at frequency selectivity index less than about 1.0. And bit energy-to-noise ratio gain is also matched theoretical value of 5 dB. BER is effectively decreased at frequency selective multipath channel with frequency selectivity index higher than 1.0. But bit energy-to-noise ratio greater than a certain size in terms of CC weaving is effective in reducing bit errors. In the results, the defined frequency selectivity index in this study could be applied to evaluate a performance of CC in multipath channel. Also it could effectively reduced BER in a low speed underwater acoustic communication system without an equalizer.
Design and Fabrication of 2D Array Ultrasonic Transducers with a Conductive Backer
Woo, Jeongdong ; Roh, Yongrae ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 502~508
DOI : 10.7776/ASK.2013.32.6.502
In this paper, 2D array transducers using a conductive backer similar to 1-3 composites have been designed, fabricated, and evaluated. The conductive backer was based on well known manufacturing process of 1-3 composites with affordable ingredients. The 2D array transducer had 4,096 elements designed to have 3.5 MHz center frequency and a fractional bandwidth over 60 %. Fabricated prototype of the transducer satisfied the specifications in the center frequency and bandwidth. Performance over the entire elements was so uniform that the standard deviation was less than 0.81 dB. Thus applicability of the conductive backer proposed in this work to 2D array transducers was verified.
A Study on the Audio Compensation System
Jeoung, Byung-Chul ; Won, Chung-Sang ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 509~517
DOI : 10.7776/ASK.2013.32.6.509
In this paper, we researched a method that makes a good acoustic-speech system using a digital signal processing technique with dynamic microphone as a transducer. Good acoustic-speech system should deliver the original sound input to electric signal without distortion. By measuring the frequency response of the microphone, adjustment factors are obtained by comparing measured data and standard frequency response of microphone for each frequency band. The final sound levels are obtained using the developed adjustment factors of frequency responses from the microphone and speaker to match the original sound levels using the digital signal processing technique. Then, we minimize the changes in the frequency response and level due to the variation of the distance from source to microphone, where the frequency responses were measured according to the distance changes.
Enhanced Normalized Subband Adaptive Filter with Variable Step Size
Chung, Ik Joo ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 518~524
DOI : 10.7776/ASK.2013.32.6.518
In this paper, we propose a variable step size algorithm to enhance the normalized subband adaptive filter which has been proposed to improve the convergence characteristics of the conventional full band adaptive filter. The well-known Kwong`s variable step size algorithm is simple, but shows better performance than that of the fixed step size algorithm. However, in case that large additive noise is present, the performance of Kwong`s algorithm is getting deteriorated in proportion to the amount of the additive noise. We devised a variable step size algorithm which does not depend on the amount of additive noise by exploiting a normalized adaptation error which is the error subtracted and normalized by the estimated additive noise. We carried out a performance comparison of the proposed algorithm with other algorithms using a system identification model. It is shown that the proposed algorithm presents good convergence characteristics under both stationary and non-stationary environments.
Low Delay IntMDCT Using Power Complementary Window
Lee, Sang-Hwan ; Lee, In-Sung ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 525~531
DOI : 10.7776/ASK.2013.32.6.525
In this paper, we propose to apply low delay algorithm using power complementary window to Integer Modified Discrete Cosine Transform (IntMDCT). Conventional transform, the Modified Discrete Cosine Transform (MDCT) usually produces floating point values for integer input values. This causes the expansion of the data. To refine on this, IntMDCT that produces integer values even for integer input values have emerged. However, IntMDCT has a problem of the algorithm delay, such as MDCT. Delay has became a key issue in environments for the purpose of real-time communications. In order to reduce the delay, the proposed algorithm was applied and the results of the performance evaluation show that delay of IntMDCT has reduced by halfexisting delay.
Receiving Signal Level Measurement Based Weighting Method for Broadband Energy Detection
Kang, TaeSu ; Kim, Youngshin ; Kim, Yong Guk ; Moon, Sang-Taeck ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 532~540
DOI : 10.7776/ASK.2013.32.6.532
In this paper, we propose the modified SED (Subband Energy Detection) which can assign weights adapting to the receiving signal level for the broadband energy detection in the passive SONARs. SED which is one of the broadband processing mainly employed by passive SONARs to detect a target is more robust against interference like multi signals or a clutter than CED (Conventional Energy Detection), but it degrades detection performance to assign weights independent of extracted extrema level of the receiving signal. Therefore, in this paper, the weighting method which can efficiently assigns rewards or penalties adapting to extracted extrema level of the receiving signal is proposed. In order to evaluate the performance of proposed method, we conducted experiments by using simulation and real ocean acoustic signal which is acquired from Yellow Sea. From the experiments, our proposed method has shown better performance than conventional SED.
Audio Mixer Algorithm for Enhancing Speech Quality of Multi-party Audio Telephony
Ryu, Sang-Hyeon ; Kim, Hyoung-Gook ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 541~547
DOI : 10.7776/ASK.2013.32.6.541
The speech quality of multi-party audio telephony between two, three or more participants is decreased by audio volume imbalance, audio volume saturation and noise level increase. To solve this issue, this paper proposes an advanced audio mixing algorithm for software-based multi-point control unit. Our approach is based on the combined voice activity detection and gain control technique that consists of a set of algorithms that classify audio signals, estimate audio volumes, adjust gain factors and mix audio signals of all channels. The proposed audio mixing algorithm is computationally efficient, delivers high-quality speech, and is suitable for use in any practical multi-party audio telephony.
An Adaptive Time Delay Estimation Method Based on Canonical Correlation Analysis
Lim, Jun-Seok ; Hong, Wooyoung ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 548~555
DOI : 10.7776/ASK.2013.32.6.548
The localization of sources has a numerous number of applications. To estimate the position of sources, the relative delay between two or more received signals for the direct signal must be determined. Although the generalized cross-correlation method is the most popular technique, an approach based on eigenvalue decomposition (EVD) is also popular one, which utilizes an eigenvector of the minimum eigenvalue. The performance of the eigenvalue decomposition (EVD) based method degrades in the low SNR and the correlated environments, because it is difficult to select a single eigenvector for the minimum eigenvalue. In this paper, we propose a new adaptive algorithm based on Canonical Correlation Analysis (CCA) in order to extend the operation range to the lower SNR and the correlation environments. The proposed algorithm uses the eigenvector corresponding to the maximum eigenvalue in the generalized eigenvalue decomposition (GEVD). The estimated eigenvector contains all the information that we need for time delay estimation. We have performed simulations with uncorrelated and correlated noise for several SNRs, showing that the CCA based algorithm can estimate the time delays more accurately than the adaptive EVD algorithm.
Frequency Band Selection Exited Linear Prediction Wideband Speech/Audio Coding Using SBR
Jang, Sunghoon ; Lee, Insung ;
The Journal of the Acoustical Society of Korea, volume 32, issue 6, 2013, Pages 556~562
DOI : 10.7776/ASK.2013.32.6.556
This paper is aimed to improve performance of Band-Selection speech/audio Coder reconstucted band spectrum that is not sent by the comfort noise. To improve the performance, we use the Spectral Band Replication(SBR) technique instead of substitution of Comfort noise. To synthesize SBR signal, the SBR algorithm is referenced in selected signals and the spectrum synthesized by SBR is injected to non-selected band. Each sub-band spectrum has been energy-weighted by real audio signal. We propose the enhanced the Band-Selection Coder that utilizes synthesized SBR signal from selected signal instead of comfort noise.