• The Journal of the Acoustical Society of Korea
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• v.22 no.1
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• pp.42-47
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• 2003

This paper proposes the method to generate new test data using the sample shift of the start frame for SPRT(sequential probability ratio test) in speaker verification. The SPRT method is a effective algorithm that can reduce the test computational complexity. However, in making the decision procedure, SPRT can be executed on the assumption that the input samples are usually to be i.i.d. (Independent and Identically Distributed) samples from a probability density function (pdf), also it's not suitable method to apply for the short utterance. The proposed method can achieve SPRT regardless of the utterance length of the test data because it is method to generate the new test data through the sample shift of start frame. Also, the correlation property of data to be considered in the SPRT method can be effectively removed by employing the principal component analysis. Experimental results show that the proposed method increased the computational complexity of data for sample shift a little, but it has a good performance result more than a conventional method above the average 0.7% in EER (equal error rate).

• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.6
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• pp.1249-1256
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• 2019

In this paper, we propose a new algorithm to remove acoustic feedback and noise in hearing aids. Instead of using the conventional FIR structure, this algorithm is a deep learning algorithm using neural network adaptive prediction filter to improve the feedback and noise reduction performance. The feedback canceller first removes the feedback signal from the microphone signal and then removes the noise using the Wiener filter technique. Noise elimination is to estimate the speech from the speech signal containing noise using the linear prediction model according to the periodicity of the speech signal. In order to ensure stable convergence of two adaptive systems in a loop, coefficient updates of the feedback canceller and noise canceller are separated and converged using the residual error signal generated after the cancellation. In order to verify the performance of the feedback and noise canceller proposed in this study, a simulation program was written and simulated. Experimental results show that the proposed deep learning algorithm improves the signal to feedback ratio(: SFR) of about 10 dB in the feedback canceller and the signal to noise ratio enhancement(: SNRE) of about 3 dB in the noise canceller than the conventional FIR structure.

• The Journal of the Acoustical Society of Korea
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• v.42 no.6
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• pp.491-499
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• 2023

Scattering by Sea surface roughness occurs due to sea level roughness, communication performance deteriorates by causing frequency spread in communication signals and time variation in communication channels. In order to compare the difference in time variation of underwater acoustic communication channel according to the surface roughness, an experiment was performed in a tank owned by Hanyang University Ocean Acoustics Lab. Artificial surface roughness was created in the tank and communication signals with three bandwidths were used (8 kHz, 16 kHz, 32 kHz). The measured surface roughness was converted into a Rayleigh parameter and used as a roughness parameter, and statistical analysis was performed on the time-varying channel characteristics of the surface path using Doppler spread and correlation time. For the Doppler spread of the surface path, the Weighted Root Mean Square Doppler spread (w_fσ_ν) that corrected the effect of the carrier frequency and bandwidth of the communication signal was used. Using the correlation time of the surface path and the energy ratio of the direct path and the surface path, the correlation of total channels was simulated and compared with the measured correlation time of total channels. In this study, we propose a method for efficient communication signal design in an arbitrary marine environment by using the time-varying characteristics of the sea surface path according to the sea surface roughness.

• The Journal of the Acoustical Society of Korea
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• v.26 no.2
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• pp.61-68
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• 2007

The frequency response of a microphone, which indicates the frequency range that a microphone can output within the approved level, is one of the most significant standards used to measure the characteristics of a microphone. At present, conventional methods of measuring the frequency response are complicated and involve the use of expensive equipment. To complement the disadvantages, this paper suggests a new algorithm that can measure the frequency response of a microphone in a simple manner. The algorithm suggested in this paper generates the Optimized Aoshima's Time Stretched Pulse(OATSP) signal from a computer via a standard speaker and measures the impulse response of a microphone by convolution the inverse OATSP signal and the received by the microphone to be measured. Then, the frequency response of the microphone to be measured is calculated using the signals. The performance test for the algorithm suggested in the study was conducted through a comparative analysis of the frequency response data and the measures of frequency response of the microphone measured by the algorithm. It proved that the algorithm is suitable for measuring the frequency response of a microphone, and that despite a few errors they are all within the error tolerance.

• Tunnel and Underground Space
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• v.21 no.5
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• pp.349-358
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• 2011

This paper introduces a new P wave arrival time determination algorithm of acoustic emission (AE) suitable to identify P waves with low signal-to-noise ratio generated in rock masses around the high-level radioactive waste disposal repositories. The algorithms adopted for this paper were amplitude threshold picker, Akaike Information Criterion (AIC), two step AIC, and Hinkley criterion. The elastic waves were generated by Pencil Lead Break test on a granite sample, then mixed with white noise to make it difficult to distinguish P wave artificially. The results obtained from amplitude threshold picker, AIC, and Hinkley criterion produced relatively large error due to the low signal-to-noise ratio. On the other hand, two step AIC algorithm provided the correct results regardless of white noise so that the accuracy of source localization was more improved and could be satisfied with the error range.

• Journal of the Korean Society for Nondestructive Testing
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• v.21 no.1
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• pp.54-61
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• 2001

A software package to classify acoustic emission (AE) signals using the wavelet transform and the neural network was developed Both of the continuous and the discrete wavelet transforms are considered, and the error back-propagation neural network is adopted as m artificial neural network algorithm. The signals acquired during the 3-point bending test of specimens which have artificial defects on weld zone are used for the classification of the defects. Features are extracted from the time-frequency plane which is the result of the wavelet transform of signals, and the neural network classifier is tamed using the extracted features to classify the signals. It has been shown that the developed software package is useful to classify AE signals. The difference between the classification results by the continuous and the discrete wavelet transforms is also discussed.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.181-189
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• 2011

According to developments of digital signal processing, 3D sound come into focus on multimedia systems. Many studies on 3d sound have proposed lots of clues to create realistic sounds. But these clues are only focused on binaural systems which two ears are normal. If we make the 3d sound using those clues at monaural systems, the performance goes down dramatically. In order to use the clues for monaural systems, we have studies algorithms such as duplex theory. In duplex theory, the sounds that we listen are affected by human's body, pinna and shoulder. So, we can enhance sound localization performances using its characteristics. In this paper, we propose a new method to use psychoacoustic theory that creates realistic 3D audio at monaural systems. To improve 3d sound, we calculate the excitation energy rates of each symmetric HRTF and extract the weights in each bark range. Finally, they are applied to emphasize the characteristics related to each direction. Informal listening tests show that the proposed method improves sound localization performances much better than the conventional methods.

• Journal of the Korean Society of Fisheries and Ocean Technology
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• v.34 no.1
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• pp.85-95
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• 1998

The broadband ultrasonic transducers have been designed to use in obtaining the broadband echo signals from fish schools in relation to the identification of fish species. The broadening of bandwidth was achieved by attaching double acoustic matching layers on the front face of a Tonpilz transducer consisted of an aluminum head, a piezoelectric ring, a brass tail and to evaluate the performance characteristics, such as the transmitting voltage response(TVR) of transducers. The constructed transducers were tested experimentally and numerically by changing the parameters such as impedances and thicknesses of the head, tail and matching layers, in the water tank. Also, the developed transducer was excited by a chirp signal and the received chirp waveforms were analyzed. According to the measured TVR results, the available 3 dB bandwidth of the transducer with double matching layers of an $Al_O_3/epoxy$ composite of 7 mm thick and a polyurethane window of 18 mm thick was 7.3 kHz with a center frequency of 38.8 kHz, and the maximum and the minimum values of the TVR in this frequency region were 135.7 dB and 132.7 dB re $1\;{\mu}Pa/V$ at 1 m, respectively. Also, the available 3 dB bandwidth of the transducer with double matching layers of an $Al_O_3/epoxy$ composite of 11 mm thick and a polyurethane window of 15 mm thick was 6.2 kHz with a center frequency of 38.6 kHz, and the maximum TVR value in the frequency region was 136.3 dB re $1\;{\mu}Pa/V$ at 1 m. Reasonable agreement between the experimental results and the numerical results for the TVR of the developed transducers was achieved. The frequency dependant characteristics of experimentally observed chirp signals closely matched to the measured TVR results. These results suggest that there is potential for increasing the bandwidth by varying other parameters in the transducer design and the material of the acoustic matching layers.

• Journal of rehabilitation welfare engineering & assistive technology
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• v.11 no.2
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• pp.133-141
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• 2017

The hybrid acoustic sensor for implantable hearing aid has the structure in which a sound pressure based acoustic sensor (ECM) and a vibration based acceleration sensor are combined. This sensor combines the low frequency sensitivity of an acoustic sensor with the high frequency sensitivity of an acceleration sensor, allowing the acquisition of a wide range of sound from low to high frequency. In this paper, an acceleration sensor for use in a hybrid acoustic sensor has been proposed. The acceleration sensor captures the vibration of the tympanic membrane generated by the acoustic signal. The size of the proposed acceleration sensor was determined to diameter of 3.2 mm considering the anatomical structure of the tympanic membrane and the standard of ECM. In order to make the hybrid acoustic sensor have high sensitivity and wide bandwidth characteristics, the aim of the resonance frequency of the acceleration sensor is to be generated at about 3.5 kHz. The membrane of the acceleration sensor derives geometric structure through mathematical model and finite element analysis. Based on the analysis results, the membrane was implemented through a chemical etching process. In order to verify the frequency characteristics of the implemented membrane, vibration measurement experiment using external force was performed. The experiment results showed mechanical resonance of the membrane occurred at 3.4 kHz. Therefore, it is considered that the proposed acceleration sensor can be utilized for a hybrid acoustic sensor.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.57-62
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• 2000

In recent, the innovative improvement of a internet and computing environment make users desire the capability of processing information in real time. In this paper we implement a PC-to-PC real time multi-user communication system on the internet environment using the efficient algorithm for a real time processing and the MFEG-4 CELP codec which can be used for a low bit-rate coding from 6 to 24kbps. The implemented system produces a compressed bit-streams with the MPEG-4 CEU Mode-I 18200bps mode. There is 5 frames for a package and 1 frame has 160 samples. We can use this system to communicate with 4 users simultaneously in real time. The system is designed and examined on the Windows operating system.