Go to the main menu
Skip to content
Go to bottom
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
Editor in Chief :
Volume & Issues
Volume 34, Issue 6 - Nov 2015
Volume 34, Issue 5 - Sep 2015
Volume 34, Issue 4 - Jul 2015
Volume 34, Issue 3 - May 2015
Volume 34, Issue 2 - Mar 2015
Volume 34, Issue 1 - Jan 2015
Selecting the target year
Acoustic Wireless Communication from Smart Phone to Hearing Aid
Jarng, Soon Suck ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 91~97
DOI : 10.7776/ASK.2015.34.2.091
In this paper, wireless communication from a smart phone to a hearing aid using audible frequency band sound was considered. 1kHz single channel carrier frequency was applied for amplitude shift keying (ASK) binary data modulation, and the result of the transmission and reception was tested. The overall system process was precisely explained and experimentally evaluated. The result will be applied for realizing a digital hearing aid with remote control feature by a smart phone.
Possibility of False Target Signals Induced by Reverberation Due to Internal Waves in Shallow Water
Lee, Sung Chun ; Kim, Sunhyo ; Choi, Jee Woong ; Kang, Donhyug ; Park, Joung Soo ; Park, Kyeongju ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 98~107
DOI : 10.7776/ASK.2015.34.2.098
It is investigated that there exists the possibility of the false target signals induced by reverberation in an active sonar system due to the internal waves in shallow water. The rays down-refracted from the internal waves may generate strong bottom-reverberation signals, which can result in false target signals. Sound waves emitted from a source propagate 3-dimensionally. Therefore, the study of internal waves on the reverberation should be studied for azimuthal direction as well as 2-dimensional (r-z) plane. Internal-wave modelling was conducted, based on solitons which were predicted with the various conditions such as, the range of source-soliton, horizontal widths of soliton. Variable depth sonar (VDS) was assumed as a source, of which the depth was located in the minimum sound speed layer in a simulation environment. Finally, the simulation on the reverberation level with time was made based on ray-based reverberation model, and the results implied that several false-target signals could be displayed on the PPI(Plan Position Indicator) scope simultaneously with range from source to soliton, and the horizontal width of soliton.
Modeling of Sound-structure Interactions for Designing a Piezoelectric Micro-Cantilever Acoustic Vector Sensor
Yang, Seongkwan ; Kim, Junsoo ; Moon, Wonkyu ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 108~116
DOI : 10.7776/ASK.2015.34.2.108
An acoustic vector sensor is a device that is capable of measuring the direction of wave propagation and the acoustic pressure. In this paper, the modeling of micro-cantilever sensor for the vector sensor are proposed by consideration of acoustic phenomenon in water. Two models based on unimorph structure are proposed in this paper and corresponding transfer function which describes the relation between input pressure wave and output voltage depending on incidence angle and frequency of pressure wave is derived based on lumped model. It has been shown that very thin and flexible micro-cantilever can be used to measure directly the particle velocity component in water.
Design and Development Research of a Parametric Array Transducer for High Directional Underwater Communication
Hwang, Yonghwan ; Je, Yub ; Moon, Wonkyu ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 117~129
DOI : 10.7776/ASK.2015.34.2.117
A parametric array is a nonlinear phenomenon that generates a narrow beam of low-frequency sound using the nonlinearity of the medium. The low-frequency sound so generated has a low sound pressure compared with that of sound generated directly. Consequently, a transducer that can generate a primary wave with high directivity and level is required. This study designed, fabricated, and evaluated a multi-resonance transducer as a parametric array source. The designs of the unit transducers and array transducer were based on an analysis model. The design process was repeated to fabricate the optimum transducer. The fabricated transducer array can generate a 189 dB, 190 dB primary wave level at 6.3 m and a 134 dB difference frequency wave using the parametric array phenomenon. The difference frequency wave has a frequency of 15 kHz and high directivity with an
half power beam width in a
A Filtered-x Affine Projection Sign Algorithm with Improved Convergence Rate for Active Impulsive Noise Control
Lee, En Jong ; Kim, Jeong Rae ; Chung, Ik Joo ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 130~137
DOI : 10.7776/ASK.2015.34.2.130
In this paper, we propose a new Modified Filtered-x Affine Projection Sign Algorithm(MFxAPSA) to improve the convergence speed of the conventional MFxAPSA which has been proposed for active control of impulsive noise. Under the impulsive noise environment, the adaptive algorithms based on the second order moment such as the Filtered-x Least Mean Square(FxLMS) show slow convergence speed or diverge because the noise source tends to have infinite variance. The MFxAPSA is the algorithm derived by applying the Affine Projection Sign Algorithm(APSA) to active noise control. The APSA has an advantage that it does not need the calculation for the inverse matrix, so it may be suitable for the active noise control that requires low computational burden. The proposed MFxAPSA also has APSA`s advantage and furthermore, better performance than the conventional MFxAPSA. We carried out a performance comparison of the proposed MFxAPSA with the conventional MFxAPSA. It is shown that the proposed MFxAPSA has the faster convergence speed than the conventional MFxAPSA.
Changes of Acoustic Reflex Thresholds and Speech-In-Noise Perception Using Personal Listening Device Under Subway Interior Noise
Han, Woojae ; Chun, Hyungi ; Ma, Sunmi ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 138~145
DOI : 10.7776/ASK.2015.34.2.138
Although it is well-known that environmental noise can lead to hearing loss in individuals, the true extent of subway noise effects in the general population remains poorly understood. The purpose of the present study is to see changes of acoustic reflex thresholds and speech perception scores when passengers listen to music presented from their personal listening device in the subway. Forty subjects with normal hearing participated being divided into two groups, experimental and control groups. As a baseline, all subjects were measured by acoustic reflex thresholds in five test frequencies and Korean speech perception in noise (KSPIN) test at 0 and -5 dB SNR. In the experiment, the control group read newspaper or magazine in the subway noise, whereas the experimental group listened to music presented from their smartphone under the subway noise through speakers at 73.45 dBA for 60 min. After completing the experiment, two groups also conducted both acoustic reflex thresholds and KSPIN tests in the same condition as the baseline. The results showed that there was a significant difference of correct percent in speech-in-noise test between experimental and control groups and of that between two signal-to-noise ratios, which means the double noise exposure of 60 min might cause some degradation of speech perception when noise increases compared to only subway noise condition that was not statistically significant difference. We concluded that a risk of some degraded speech perception ability would be expected when passengers have a habit of listening to music in the subway noisy situation for a long duration.
Comparison of Acoustic Performance Depending on the Location of Sound Absorptive and Diffuser in Small Auditoriums Using 1/10 Scale Models
Kim, Tae-Hee ; Park, Chan-Jae ; Park, Ji-Hoon ; Haan, Chan-Hoon ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 146~156
DOI : 10.7776/ASK.2015.34.2.146
This study investigated how the location of sound absorptive materials and sound diffusers affects the acoustic performance of small auditoriums. It was conducted for a standard model established with the averaged dimension of 36 auditoriums which had opened since 2000 in Daehak-ro, Seoul. In this study, the installation area of finishing materials was calculated upon a back wall which had the smallest installation effective area of finishing materials. To analyze the changes of acoustic performance according to installation location of finishing materials, experiments were carried out using the 1/10 down scale models for 8 cases which were made by classifying the installation location of ceiling and side wall into the front, middle and rear part.The used acoustic parameters were reverberation time (RT), early decay time (EDT), clarity (C80), definition (D50) and speech transmission index (STI). In result, the index related to the amount of reverberant sound (RT, EDT) showed the great changes when evaluating it through just noticeable difference (JND), but the one related to clarity (C80, D50, STI) hardly indicated the changes. In case to obtain short reverberation time, it was most effective to control reverberation time through the side walls when installing sound absorptive and diffusive materials, and side wall front was the location which could get the shortest reverberation time.
Audio Fingerprinting Using a Robust Hash Function Based on the MCLT Peak-Pair
Lee, Jun-Yong ; Kim, Hyoung-Gook ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 157~162
DOI : 10.7776/ASK.2015.34.2.157
In this paper, we propose an audio fingerprinting using robust hash based on the MCLT (Modulated Complex Lapped Transform) peak-pair. In existing methods, the robust audio fingerprinting is not generated if various distortions occurred; time-scaling, pith-shifting and equalization. To solve this problem, we used the spectrum of the MCLT, an adaptive thresholding method for detection of prominent peaks and the novel hash function in the audio fingerprinting. Experimental results show that the proposed method is highly robust in various distorted environments and achieves better identification rates compared to other methods.
A Robust Frequency-Domain Multi-Reference Narrowband Adaptive Noise Canceller
Kim, Seong-Woo ; Seo, Ji-Ho ; Ryu, Young-Woo ; Park, Young-Cheol ; Youn, Dae Hee ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 163~170
DOI : 10.7776/ASK.2015.34.2.163
In this paper, it is shown that the performance of the frequency-domain multi-reference narrowband noise canceller is determined by the narrowband component to the broadband disturbance power ratio in the reference signals. To overcome this problem, a new narrowband ANC is proposed, where the update of the adaptive filter is determined based on SNR of the reference inputs being measured using the magnitude squared coherence (MSC) between the primary and the reference signals. Simulation results show that the proposed ANC has superior performance over the conventional one.
Text Independent Speaker Verficiation Using Dominant State Information of HMM-UBM
Shon, Suwon ; Rho, Jinsang ; Kim, Sung Soo ; Lee, Jae-Won ; Ko, Hanseok ;
The Journal of the Acoustical Society of Korea, volume 34, issue 2, 2015, Pages 171~176
DOI : 10.7776/ASK.2015.34.2.171
We present a speaker verification method by extracting i-vectors based on dominant state information of Hidden Markov Model (HMM) - Universal Background Model (UBM). Ergodic HMM is used for estimating UBM so that various characteristic of individual speaker can be effectively classified. Unlike Gaussian Mixture Model(GMM)-UBM based speaker verification system, the proposed system obtains i-vectors corresponding to each HMM state. Among them, the i-vector for feature is selected by extracting it from the specific state containing dominant state information. Relevant experiments are conducted for validating the proposed system performance using the National Institute of Standards and Technology (NIST) 2008 Speaker Recognition Evaluation (SRE) database. As a result, 12 % improvement is attained in terms of equal error rate.