• The Journal of the Acoustical Society of Korea
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• v.37 no.1
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• pp.31-38
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• 2018

Based on the measured data in the south western sea of Jeju Island during the SAVEX15(Shallow Water Acoustic Variability EXperiment 2015), the effect of internal waves on the PPD (Predictive Probability of Detection) of a sonar system was analyzed. The southern west sea of Jeju Island has complex flows due to internal waves and USC (Underwater Sound Channel). In this paper, sonar performance is predicted by probabilistic approach. The LFM (Linear Frequency Modulation) and MLS (Maximum Length Sequence) signals of 11 kHz - 31 kHz band of SAVEX15 data were processed to calculate the TL (Transmission Loss) and NL (Noise Level) at a distance of approximately 2.8 km from the source and the receiver. The PDF (Probability Density Function) of TL and NL is convoluted to obtain the PDF of the SE (Signal Excess) and the PPD according to the depth of the source and receiver is calculated. Analysis of the changes in the PPD over time when there are internal waves such as soliton packet and internal tide has confirmed that the PPD value is affected by different aspects.

• Journal of Navigation and Port Research
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• v.39 no.4
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• pp.293-297
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• 2015

Underwater acoustic communications has been limited use for military purposes in the past. However, the fields of underwater applications expend to detection, submarine and communication in recent. The excessive multipath encountered in underwater acoustic communication channel is creating inter symbol interference, which is limiting factor to achieve a high data rate and bit error rate performance. To improve the performance of a received signal in underwater communication, many researchers have been studied for channel coding scheme with excellent performance at low SNR. In this paper, we applied BCJR decoder based ( 2,1,7 ) convolution codes and to compensate for the distorted data induced by the multipath, we applying the turbo equalization method. Through the underwater experiment on the Gyeungcheun lake located in Mungyeng city, we confirmed that turbo equalization structure of BCJR has better performance than hard decision and soft decision of Viterbi decoding. We also confirmed that the error rate of decoder input is less than error rate of $10^{-1}$, all the data is decoded. We achieved sucess rate of 83% through the experiment.

• Journal of the Korean Society for Precision Engineering
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• v.22 no.8 s.173
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• pp.143-150
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• 2005

To verify applicability of multi-dimensional spectral analysis (MDSA) fur noise source identification two different approaches which are frequency response and coherent function have been investigated. The coherence function approach appears able to separate the correlated system when the noise sources were coherent. In this study, we identify contribution of structure-borne-noise of vehicle HVAC system using MDSA method. Firstly, to identify the applicability of MDSA method, 4-inputs of vehicle HVAC system were the signals measured by accelerometers attached on the selected noise sources which were composed of blower, evaporator, heater and duct. While 1-output which was driver's position sound was the SPL signals measured by a remote microphone, when the blower motor was operating. We identify efficiency of systems modeled with four Inputs/single output through ordinary coherence function (OCF) and partial coherence function (PCF). As a result of experiment, the blower accounted for $62-88\%$ of the overall level of sound energy density. Also, according to the analysis of acoustic signal and vibration signals measurement, an investigation of the noise source identification in the vehicle HVAC is presented. With the sound intensity method, the major sources of the vehicle HVAC radiation are verified. Also the method of improving the noise reduction is proposed by attaching damping patch access to blower motor and noise reduction is verified.

• The Journal of the Acoustical Society of Korea
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• v.30 no.7
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• pp.390-395
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• 2011

This paper covers the studies on common voice parameters through the voice analysis of female main news anchors on weekday evening by the station, and differences of relative voices and sounds among stations. To examine voice characteristics, 6 voice parameters were analyzed and it showed anchors of each station had distinctive characteristics of voices and phonations over all fields except the speech rate, and there were also differences in sound systems. As major analysis parameters, basic pitch, tone of the 1st formant and pitch ratio, level of closeness by pitch bandwidth, type of sentence closing through average pitch position within pitch bandwidth, average speech rate, and acoustic tone analysis by energy distribution by frequency band were used. Analyzed values and results could be referred to and utilized in the criteria of phonation characteristics for domestic female news anchors.

• The Journal of the Acoustical Society of Korea
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• v.38 no.6
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• pp.651-660
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• 2019

The present study aims to find out the necessary speech sound level which can satisfy with the speech intelligibility in a noisy classroom environments. For this, auralized materials were made to undertake listening tests with 27 people. Speech intelligibility tests were carried out using both Consonant-Vowel-Consonant (CVC) and Phonetically Balanced Words (PBW) methods. Signal to noise ratio was changed by 5 dB for each test. As a result, it was found that speech intelligibilities are increasing with larger Signal to Noise Ratio (SNR). It was also found that there is a lot of difference of speech intelligibilities by SNR for syllables (CVC) with the Reverberation Time (RT) of 1.5 s. However, any significant difference was not found for words (PBW) in the case with RTs of below 0.8 s. Also, it was revealed through the 2-way analysis of variance (ANOVA) test that SNR is the only attentive factor which can affect the Korean speech intelligibilities for both PBW and CVC methods. Therefore, RTs below 0.8 s could be the acoustic criteria for classroom which can minimize the effects of noise. In the case with RTs larger than 0.8 s, much larger SNR is needed to give sufficient speech intelligibility.

• The Journal of the Acoustical Society of Korea
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• v.19 no.7
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• pp.66-76
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• 2000

Acoustic performance of loudspeakers for sound reproduction has been qualitatively evaluated by using the listening test by juries in the development and final evaluation stages. However, the subjective evaluation method has many problems in the viewpoint of reliability and repeatability that are mainly related to the jury performance, as well as time and economy. In this reason, objective techniques should be tried to evaluate the acoustic performance of loudspeakers as well as the conventional subjective test. The object of this study is to find if there is any correlation between the statistically treated in test results and the measured results based on the loudness of reproduced sound signals. For the four-step statistical analysis, the analysis of variance (ANOVA) and Tukeys method are employed for dealing with the data from the listening test. For the objective evaluation, Zwickers loudness considering the human hearing characteristics is calculated for the measured sound signal emitted from each loudspeaker and the objective ratings such as fidelity rating (FR) and softness rating (SR) is suggested. The correlation between two ratings has been demonstrated for an actual set of loudspeakers using FR, SR and correlation coefficient. The method in this study can be useful in statistically evaluating commercial or prototype loudspeakers without using very time-consuming and expensive subjective testing.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.1
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• pp.111-120
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• 2017

In this paper, we propose novel feature extraction algorithm for classification of seabed sediment. In previous researches, acoustic reflection coefficient has been used to classify seabed sediments, which is constant in terms of frequency. However, attenuation of seabed sediment is a function of frequency and is highly influenced by sediment types in general. Hence, we developed a feature vector by using attenuation variation with respect to frequency. The attenuation variation is obtained by using reflected signal from the second sediment layer, which is generated by broadband chirp. The proposed feature vector has advantage in number of dimensions to classify the seabed sediment over the classical scalar feature (reflection coefficient). To compare the proposed feature with the classical scalar feature, dimension of proposed feature vector is reduced by using linear discriminant analysis (LDA). Synthesised acoustic amplitudes reflected by seabed sediments are generated by using Biot model and the performance of proposed feature is evaluated by using Fisher scoring and classification accuracy computed by maximum likelihood decision (MLD). As a result, the proposed feature shows higher discrimination performance and more robustness against measurement errors than that of classical feature.

• 한국해양학회지
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• v.22 no.3
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• pp.143-152
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• 1987

The digital data were obtained using Kennedy 9000 magnetic tape deck which was connected to the SMS960 side scan sonar during the field operations. The data of three consecutive survey tracks near Seongsan-po, Cheju were used for the development of this study. The softwares were mainly written in Fortran-77 using VAX 11/780 MINI-COMPUTER (CPU Memory; 4MB). The established mapping system consists of the pretreatment and the digital processing of seafloor image data. The pretreatment was necessary because the raw digital data format of the field magnetic tapes was not compatible to the VAX system. Therefore the raw data were read by the personal computer using the Assembler language and the data format was converted to IBM compatible, and next data were communicated to the VAX system. The digital processing includes geometrical correction for slant range, statistical analysis and cartography of the seafloor image. The sound speed in the water column was assumed 1,500 m/sec for the slant range correction and the moving average method was used for the signal trace smoothing. Histograms and cumulative curves were established for the statistical analysis, that was purposed to classify the backscattering strength from the sea-bottom. The seafloor image was displayed on the color screen of the TEKTRONIX 4113B terminal. According to the brief interpretation of the result image map, rocky and sedimentary bottoms were very well discriminated. Also it was shown that the backscattered acoustic pressurecorrelateswith the grain size and sorting of surface sediments.

• Journal of Biosystems Engineering
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• v.33 no.1
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• pp.30-37
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• 2008

Dynamic excitation and response analysis is an acceptable method to determine some of physical properties of agricultural product for quality evaluation. There is a difference in the internal viscoelasticity between sound and defective fruits due to the difference of geometric structures, thereby showing different vibration characteristics. This study was carried out to develop a portable piezoelectric film-based glove sensor system that can separate internally damaged watermelons from sound ones using an acoustic impulse response technique. Two piezoelectric sensors based on polyvinylidene fluoride (PVDF) films to measure an impact force and vibration response were separately mounted on each glove. Various signal parameters including number of peaks, energy ratio, standard deviation of peak to peak distance, zero-crossing rate, and integral value of peaks were examined to develop a regression-estimated model. When using SMLR (Stepwise Multiple Linear Regression) analysis in SAS, three parameters, i.e., zeros value, number of peaks, and standard deviation of peaks were selected as usable factors with a coefficient of determination ($r^2$) of 0.92 and a standard error of calibration (SEC) of 0.15. In the validation tests using twenty watermelon samples (sound 9, defective 11), the developed model provided good capability showing a classification accuracy of 95%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1B
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• pp.89-98
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• 2000

This paper presents an active noise cancelling model using wavelet transform and subband filter banks based on adaptive filter. The analysis filter banks decompose input and error signals into QMF filter banks of lowpass and highpass bands. Each filter bank uses wavelet filter with dyadic tree structure. The decomposed input and error signals are iterated by adaptive filter coefficients of each subband using filtered-X LMS algorithm. The synthesis filter banks make output signal of wideband with perfect reconstruction to prepare adaptive filter output signals of each subband. The analysis and synthesis niter hants use conjugate quadrature filters for Pefect reconstruction. Also, The delayed LMS algorithm model for on-line identification of error path transfer characteristics is used gain and acoustic time delay factors. The proposed adaptive active noise cancelling modelis suggested by system retaining the computational and convergence speed advantage using wavelet subband filter banks.