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 27, Issue 8 - Nov 2008
Volume 27, Issue 7 - Oct 2008
Volume 27, Issue 6 - Aug 2008
Volume 27, Issue 5 - Jul 2008
Volume 27, Issue 4 - May 2008
Volume 27, Issue 3 - Apr 2008
Volume 27, Issue 2 - Feb 2008
Volume 27, Issue 1 - Jan 2008
Selecting the target year
Prediction Method of Loudspeaker Driver Characteristics
Park, Soon-Jong ; Rho, Sung-Tak ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 325~332
The prediction method of TS parameters, frequency response, and electrical input impedance is proposed with physical properties of parts and results of electromagnetic FEA(Finite Element Analysis) in a loudspeaker driver design. In design for weight reduction and improvement of flux density asymmetry, the prediction results are well coincided with measurement ones. As the applications, it can be applied in design for improvement of the
harmonic distortion with flux density distribution analysis. The proposed method is expected to be utilized for reducing trial-and-error process in electromagnetic parts design. It can also be used for providing guidelines for parts selection in the early stages.
HRTF Interpolation Using a Spherical Head Model
Lee, Ki-Seung ; Lee, Seok-Pil ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 333~341
In this paper, a new interpolation model for the head related transfer function (HRTF) was proposed. In the method herein, we assume that the impulse response of the HRTF for each azimuth angle is given by linear interpolation of the time-delayed neighboring impulse responses of HRTFs. The time delay of the HRTF for each azimuth angle is given by sum of the sound wave propagation time from the ears to the sound source, which can be estimated by using azimuth angle, the physical shape of the underlying head and the distance between the head and sound source, and the refinement time yielding the minimum mean square error. Moreover, in the proposed model, the interpolation intervals were not fixed but varied, which were determined by minimizing the total number of HRTFs while the synthesized signals have no perceptual difference from the original signals in terms of sound location. To validate the usefulness of the proposed interpolation model, the proposed model was applied to the several HRTFs that were obtained from one dummy-head and three human heads. We used the HRTFs that have 5 degree azimuth angle resolution at 0 degree elevation (horizontal plane). The experimental results showed that using only
of the original HRTFs were sufficient for producing the signals that have no audible differences from the original ones in terms of sound location.
Design of Multipath Adaptive BISMO-Algorithm in the Underwater Communication
Im, Byung-Ook ; Shim, Tae-Bo ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 342~349
Multipath makes clear message transfer difficult in the underwater communication. To solve this problem, we propose a new method producing proper MFSP (Modulation Frequency Shift Period) which could be obtained by calculating time delay caused by different path from a transmitter to a receiver. At the transmitter end, messages were divided according to the size of the MFSP and transmitted accordingly alternating Frequency. At the receiver end, the received messages were demodulated in order to recover the original message by the adaptive BISMO algorithm which is constructed at the algorithm design stage. Adaptive MFSP and estimated BER (Bit Error Rate) were calculated through simulation test.
Beamforming Method for Target Range Estimation Using Near Field Shading Function
Choi, Joo-Pyoung ; Lee, Won-Cheol ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 350~356
In this paper, we propose shading functions to the appropriate focused beamforming for near-field target estimation. This near field shading functions are based on Chebychev and Manning windows. In order to obtain the optimum sensor weighting values with the help of the proposed shading technique, we assume that the sensor positions associated to the non-uniformly distributed array are precisely known. We calculate a series of sensor weighting values from the FFT operation of given shading functions in time domain. By applying the shading weights on the sensor array, we can see that the level of sidelobe becomes diminished and the performance of estimating range and azimuth gets improved. In addition, we propose a non-uniform structure in terms of frequency bands, which may minimize the attenuation of incoming signals.
A Study on Development of Acoustic Tweezer System Using Standing Waves and Very High Frequency Focused Beams
Yang, Jeong-Won ; Ha, Kang-Lyeol ; Kim, Moo-Joon ; Lee, Jung-Woo ; Shung, K.K. ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 357~364
For the purpose of possibility study on development of an acoustic tweezer using standing waves and very high frequency ultrasound focused beams, a system which can manipulate the position of particles in water has been constructed. It can move the particles to near focal point of a focused beam by the radiation force of standing waves, and then the particles would be trapped by the radiating force of the focused beam. The results show that micro sphere particles were trapped well at nodes of the standing waves and their position can be easily manipulated by frequency control. And, even though the radiation force by single focused beam pushes a particle away from the transducer, two focused confronted beams can trap it at near center.
Beat Control Method Using the Finite Element Analysis of an Equivalent Ring
Kim, Seock-Hyun ; Cui, Cheng-Xun ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 365~371
In this study, beat control method using an equivalent ring model is proposed to control beat period of a slightly asymmetric ring. Slight asymmetry in a ring generates mode pair and the interaction of the mode pair makes beat in vibration and sound. In a ring, as a simplified bell type structure, mode data are measured and an equivalent ring is determined so that the measured mode condition is satisfied. By the finite element analysis on the equivalent ring, changes of mode pair condition are predicted when local mass is attached or the local thickness is decreased. The predicted results are compared with the experimental result and the validity of the proposed method is verified.
Statistical Voice Activity Defector Based on Signal Subspace Model
Ryu, Kwang-Chun ; Kim, Dong-Kook ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 372~378
Voice activity detectors (VAD) are important in wireless communication and speech signal processing, In the conventional VAD methods, an expression for the likelihood ratio test (LRT) based on statistical models is derived in discrete Fourier transform (DFT) domain, Then, speech or noise is decided by comparing the value of the expression with a threshold, This paper presents a new statistical VAD method based on a signal subspace approach, The probabilistic principal component analysis (PPCA) is employed to obtain a signal subspace model that incorporates probabilistic model of noisy signal to the signal subspace method, The proposed approach provides a novel decision rule based on LRT in the signal subspace domain, Experimental results show that the proposed signal subspace model based VAD method outperforms those based on the widely used Gaussian distribution in DFT domain.
Speech Reinforcement Based on Soft Decision Under Far-End Noise Environments
Choi, Jae-Hun ; Chang, Joon-Hyuk ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 379~385
In this paper, we propose an effective speech reinforcement technique under the near-end and the far-end noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. Specifically, based on the estimated background noise spectrum of the near-end, we reinforce the far-end speech spectrum by incorporating the more general cases under the near-end with background noise. Also, we propose the novel approach to reinforce the actual speech signal except for the noise signal in the far-end noisy speech signal. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with the conventional method.
Adaptive Threshold for Speech Enhancement in Nonstationary Noisy Environments
Lee, Soo-Jeong ; Kim, Sun-Hyob ;
The Journal of the Acoustical Society of Korea, volume 27, issue 7, 2008, Pages 386~393
This paper proposes a new approach for speech enhancement in highly nonstationary noisy environments. The spectral subtraction (SS) is a well known technique for speech enhancement in stationary noisy environments. However, in real world, noise is mostly nonstationary. The proposed method uses an auto control parameter for an adaptive threshold to work well in highly nonstationary noisy environments. Especially, the auto control parameter is affected by a linear function associated with an a posteriori signal to noise ratio (SNR) according to the increase or the decrease of the noise level. The proposed algorithm is combined with spectral subtraction (SS) using a hangover scheme (HO) for speech enhancement. The performances of the proposed method are evaluated ITU-T P.835 signal distortion (SIG) and the segment signal to-noise ratio (SNR) in various and highly nonstationary noisy environments and is superior to that of conventional spectral subtraction (SS) using a hangover (HO) and SS using a minimum statistics (MS) methods.