• International Journal of Advanced Culture Technology
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• v.12 no.3
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• pp.206-212
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• 2024

This study presents five selection rules for training algorithms to extract audio sources from noise. The five rules are Dynamics, Roots, Tonal Balance, Tonal-Noisy Balance, and Stereo Width, and the suitability of each rule for sound extraction was determined by spectrogram analysis using various types of sample sources, such as environmental sounds, musical instruments, human voice, as well as white, brown, and pink noise with sine waves. The training area of the algorithm includes both melody and beat, and with these rules, the algorithm is able to analyze which specific audio sources are contained in the given noise and extract them. The results of this study are expected to improve the accuracy of the algorithm in audio source extraction and enable automated sound clip selection, which will provide a new methodology for sound processing and audio source generation using noise.

• The Journal of the Acoustical Society of Korea
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• v.11 no.3
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• pp.74-82
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• 1992

• Journal of the Institute of Convergence Signal Processing
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• v.13 no.1
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• pp.1-5
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• 2012

In this study, we constructed a virtual auditory display(VAD) that enables listener to move in a room freely. The VAD system was installed in a soundproof room($4.7m(W){\times}2.8m(D){\times}3.0m(H)$). The system consisted of a personal computer, a sound presentation device, and a three-dimensional ultrasound sensor system. This system acquires listener's location and position from a three-dimension ultrasonic sensor system covering the entire room. Localization was realized by convolving the sound source with head related transfer functions(HRTFs) on personal computer(PC). The calculated result is generated through a LADOMi(Localization Auditory Display with Opened ear-canal for Mixed Reality). The HRTFs used in the experiment were measured for each listener with loudspeakers constantly 1.5m away from the center of the listener' s head in an anechoic room. To evaluate the system performance, we experimented a search task of a sound source position in the condition that the listener is able to move all around the room freely. As a result, the positioning error of presented sound source was within 30cm in average for all listeners.

• Korean Journal of Fisheries and Aquatic Sciences
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• v.14 no.4
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• pp.265-268
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• 1981

The direction and spectra of underwater sound wave were a remarkable contrast to the sound wave in the air because of the difference of transmissive medium. The linear hydrophone array of passive system has so far been applied to find out the direction and spectra of underwater sound wave from the sources for many purposes. The conventional methods are generally classified into two systems such as, the system which varying frequency responses, other parameters and pattern of signal like an adaptive array controlled by internal feedback, and another system which obtaining maximum of S/N ratio by giving a appropriate delay and a weighting coefficient in the output of each hydrophone. The array device of passive system can easily change the amplitude and the phase of signal by separately controlled hydrophone. And here we introduce a method that the spectral analyzing and the direction finding can be simultaneously carried out using a linear array of hydrophones. By making a circular convolution of output of signal from each hydrophone with appropriate rectangular weighting coefficient on the array, a sharp response of single lobe directivity and the spectral analyzing by time averaging were simultaneously obtained. In tile computer simulation of the array system with the length of 250cm and the interhydrophone distance of l0cm the power levels of sound signals received from given array direction were 16dB higher than those from the other directions when processing with rectangular weightings, and 8dB higher when processing with rectangular sound signals and rectangular weightings.

• Journal of the Korean Society for information Management
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• v.36 no.3
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• pp.131-148
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• 2019

This study constructed a cognitive model of information processing to understand the topic of a sound material and its characteristics. It then proposed methods to generate sound summaries, by incorporating anterior-posterior N400/P600 components of event-related potential (ERP) response, into the language representation of the cognitive model of information processing. For this end, research hypotheses were established and verified them through ERP experiments, finding that P600 is crucial in screening topic-relevant shots from topic-irrelevant shots. The results of this study can be applied to the design of classification algorithm, which can then be used to generate the content-based metadata, such as generic or personalized sound summaries and video skims.

• Korean Journal of Audiology
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• v.24 no.4
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• pp.174-179
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• 2020

Background and Objectives: The integration of auditory-visual speech information improves speech perception; however, if the auditory system input is disrupted due to hearing loss, auditory and visual inputs cannot be fully integrated. Additionally, temporal coincidence of auditory and visual input is a significantly important factor in integrating the input of these two senses. Time delayed acoustic pathway caused by the signal passing through digital signal processing. Therefore, this study aimed to investigate the effects of hearing loss and hearing aid digital delay circuit on sound-induced flash illusion. Subjects and Methods: A total of 13 adults with normal hearing, 13 with mild to moderate hearing loss, and 13 with moderate to severe hearing loss were enrolled in this study. Subsequently, the sound-induced flash illusion test was conducted, and the results were analyzed. Results: The results showed that hearing aid digital delay and hearing loss had no detrimental effect on sound-induced flash illusion. Conclusions: Transmission velocity and neural transduction rate of the auditory inputs decreased in patients with hearing loss. Hence, the integrating auditory and visual sensory cannot be combined completely. Although the transmission rate of the auditory sense input was approximately normal when the hearing aid was prescribed. Thus, it can be concluded that the processing delay in the hearing aid circuit is insufficient to disrupt the integration of auditory and visual information.

• The Journal of the Acoustical Society of Korea
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• v.43 no.3
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• pp.314-323
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• 2024

Recently, a score-penalty method has been used for the acoustic passive tracking of marine mammals. The interesting aspect of this technique lies in the loss function, which has a penalty term representing the mismatch between the measured signal and the modeled signal, while the traditional time-domain matched-field processing is positively considering the match between them. In this study, we apply the score-penalty method into the depth estimation of a passive target with a known source waveform. Assuming deep ocean environments with uncertainties in the sound speed profile, we evaluate the score-penalty method, comparing it with the time-domain matched field processing method. We shows that the score-penalty method is more accurate than the time-domain matched field processing method in the ocean environment with weak mismatch of sound speed profile, and has better efficiency. However, in the ocean enviroment with strong mismatch of the sound speed profile, the score-penalty method also fails in the depth estimation of a target, similar to the time-domain matched-field processing method.

• Journal of KIISE:Software and Applications
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• v.28 no.10
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• pp.699-705
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• 2001

As it became possible to generate 3D sound on a PC environment due to the advances in computing performance and digital signal processing technology, 3D sound technology gains its focus in the multimedia area Specifically a two-channel based 3D sound technology is being studied by many researchers because of its space efficiency and economical structure. While the positional sound effect is simple in its implementation, the moving sound effect has many problems to be resolved as there are only discrete measured point of HRTF database. In this paper, we propose the method of generating smooth moving sound in a two-channel based 3D sound technique with respect to generating smooth trajectory, and the interpolation method of discrete measured HRTF data. We perform the tests in the PC environment and prove the utility of the proposed method.

• Journal of the Korean Institute of Navigation
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• v.24 no.5
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• pp.443-450
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• 2000

Before some experiments were carried out with analog bandpass filter which used for filtering the noise included in sound source signal. And this filter was constituted by condenser, register and operational amplifier. Hut these elements made the phase characteristics to differentiate in each sensing channel and cause a little of measurement error. We made new measurement system that was substituted digital filter for the analog filter in order to develop the optimal system which could find the time delay between each sensors with high accuracy. This paper describes the new system's constitution and the function of each parts. Specially three digital filters were designed and applied to the digital signal processing Part. And a series of experiments were carried out with the source＇s distance 9.53meters and the random bearing interval within the limits of $0^{\circ}$ ~ $180^{\circ}$. As a result, we have recognized that the accuracy of measurements were differentiated by the methods what kind of digital filter were adopted. And we have confirmed the facts that IIR LPF was suitable for sound source's bearing measurement and FIR LPF reduced the range measurement error.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.4
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• pp.167-174
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• 2002

This study describes a fetal heart rate(FHR) estimation algorithm using phonogram. Using a phonogram amplifier, various fetal heart sounds are collected in a university hospital. The FHR estimation algorithms consists of a lowpass filter, decimation, envelop detection, pitch detection, and post-processing. The post-processing is the FHR decision procedure using all informations of fetal heart rates. Using the algorithm and other parameters of fetal heart sound, a fetal monitoring software was developed. This can display the original signals, the FFT spectra, FHR and its trajectory. Even though the fetal phonogram amplifier detects the fetal heart sounds well, the sound quality is not so good as the ultrasonography. In case of very week fetal heart sound, autocorrelation of it showed clear periodicity. But two main peaks in one period is an obstacle in pitch detection and peaks are not so vivid. The proposed FHR estimation algorithm showed very accurate and stable results. Since the developed software displays multiple parameters in real time and has convenient functions, it will be useful for the phonogram-style fetal monitoring device.