• The Journal of the Acoustical Society of Korea
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• v.42 no.1
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• pp.32-41
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• 2023

Recently, studies on reverberation suppression using Non-negative Matrix Factorization (NMF) have been actively conducted. The NMF method uses a cost function based on the Kullback-Leibler divergence for optimization. And some constraints are added such as temporal continuity, pulse length, and energy ratio between reverberation and target. The tendency of constraints are controlled by hyperparameters. Therefore, in order to effectively suppress reverberation, hyperparameters need to be optimized. However, related studies are insufficient so far. In this paper, the reverberation suppression performance according to the three hyperparameters of the NMF was analyzed by using sea experimental data. As a result of analysis, when the value of hyperparameters for time continuity and pulse length were high, the energy ratio between the reverberation and the target showed better performance at less than 0.4, but it was confirmed that there was variability depending on the ocean environment. It is expected that the analysis results in this paper will be utilized as a useful guideline for planning precise experiments for optimizing hyperparameters of NMF in the future.

• The Journal of the Acoustical Society of Korea
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• v.42 no.5
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• pp.422-428
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• 2023

Ultrafast ultrasound imaging has been applied to various imaging approaches, including shear wave elastography, ultrafast Doppler, and super-resolution imaging. However, these methods are still challenging in real-time implementation for three Dimension (3D) or portable applications because of their massive data rate required. In this paper, we proposed an adaptive quantization method that effectively reduces the data rate of large Radio Frequency (RF) data. In soft tissue, ultrasound backscatter signals require a high dynamic range, and thus typical quantization used in the current systems uses the quantization level of 10 bits to 14 bits. To alleviate the quantization level to expand the application of ultrafast ultrasound imaging, this study proposed a depth-sectional quantization approach that reduces the quantization errors. For quantitative evaluation, Field II simulations, phantom experiments, and in vivo imaging were conducted and CNR, spatial resolution, and SSIM values were compared with the proposed method and fixed quantization method. We demonstrated that our proposed method is capable of effectively reducing the quantization level down to 3-bit while minimizing the image quality degradation.

• Journal of the Institute of Convergence Signal Processing
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• v.23 no.2
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• pp.84-90
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• 2022

Machine tool state monitoring is a process that automatically detects the states of machine. In the manufacturing process, the efficiency of machining and the quality of the product are affected by the condition of the tool. Wear and broken tools can cause more serious problems in process performance and lower product quality. Therefore, it is necessary to develop a system to prevent tool wear and damage during the process so that the tool can be replaced in a timely manner. This paper proposes a method for diagnosing five tool states using a deep learning-based hierarchical convolutional neural network to change tools at the right time. The one-dimensional acoustic signal generated when the machine cuts the workpiece is converted into a frequency-based power spectral density two-dimensional image and use as an input for a convolutional neural network. The learning model diagnoses five tool states through three hierarchical steps. The proposed method showed high accuracy compared to the conventional method. In addition, it will be able to be utilized in a smart factory fault diagnosis system that can monitor various machine tools through real-time connecting.

• The Journal of the Acoustical Society of Korea
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• v.42 no.6
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• pp.627-636
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• 2023

Due to the COVID-19 pandemic, the global population has experienced drastic changes, one of which is the increase in remote work. Given the ongoing possibility of exposure to infectious diseases and various other circumstances, the expansion of remote work is anticipated. To enhance the efficiency of remote work and address its existing limitations, this study surveyed the perceptions of indoor environments among individuals who worked from home during the COVID-19 pandemic. The study examined how the characteristics of individuals influenced their perceptions of indoor environments. It was found that the number of occupants and rooms, size of the house, and noise sensitivity affected the perceptions of outdoor noise, neighbor noise, and indoor noise caused by cohabitants. The findings can be used as foundational data for designing multipurpose housing that can be utilized not only for residential purposes but also for work and educational settings in the future.

• The Journal of the Acoustical Society of Korea
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• v.25 no.8
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• pp.356-361
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• 2006

We present the performance of differential space-time block decoders. which does not require the channel state information. We suggest the structure of the multiple blocks differential space time decoders. which does not require the channel state information, and analyze the Performances. In quasi-static flat fading channels. the Performance of multiple blocks differential detection (MD-STC) outperforms that of 2 blocks(D-STC) by 1.5dB. But in the time-selective fading channels due to Doppler frequency $(f_d)$, the performance of MD-STC degrades as the vehicular speed is greater than 200km/hr in 802.16e systems, where the data transmission rate is 144kbps.

• The Journal of the Acoustical Society of Korea
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• v.29 no.3
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• pp.155-164
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• 2010

This paper proposes a robust detection method for ship wake search using two wake sensors. A long trailing wake in the rear of a surface ship is generated along the track of surface ships. In this paper, we assume that the nearer the surface ship, the stronger wake strength is and a two-sensor based wake homing torpedo can sense for the wake strength. On this assumption we propose a simple wake detection and search method using information of wake strength. Experimental results using monte-carlo simulation demonstrate that the proposed method yields better performance in search time than previous method, which uses a single sensor. Our method is shown faster by about 45 seconds than previous method to achieve the same performance. Also, it can improve the detection performance of torpedo in the case of short wake length.

• The Journal of the Acoustical Society of Korea
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• v.28 no.5
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• pp.441-446
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• 2009

This paper proposes an automatic indexing algorithm of golf video using audio information. In the proposed algorithm, the input audio stream is demultiplexed into the stream of video and audio. By means of Adaboost-cascade classifier, the continuous audio stream is classified into announcer's speech segment recorded in studio, music segment accompanied with players' names on TV screen, reaction segment of audience according to the play, reporter's speech segment with field background, filed noise segment like wind or waves. And golf swing sound including drive shot, iron shot, and putting shot is detected by the method of impulse onset detection and modulation spectrum verification. The detected swing and applause are used effectively to index action or highlight unit. Compared with video based semantic analysis, main advantage of the proposed system is its small computation requirement so that it facilitates to apply the technology to embedded consumer electronic devices for fast browsing.

• The Journal of the Acoustical Society of Korea
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• v.29 no.3
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• pp.209-215
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• 2010

Recently portable terminal is received attention by wireless networks and mass capacity ROM. In this result, TTS(Text to Speech) system is inserted to portable terminal. Nevertheless high quality synthesis is difficult in portable terminal, users need high quality synthesis. In this paper, we proposed Distributed TTS (DTTS) that was composed of server and terminal. The DTTS on corpus based speech synthesis can be high quality synthesis. Synthesis system in server that generate optimized speech concatenation information after database search and transmit terminal. Synthesis system in terminal make high quality speech synthesis as low computation using transmitted speech concatenation information from server. The proposed method that can be reducing complexity, smaller power consumption and efficient maintenance.

• The Journal of the Acoustical Society of Korea
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• v.25 no.4
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• pp.151-158
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• 2006

In this paper, we proposed a new audio fingerprint extraction method, based on Philips' music retrieval algorithm, which uses the energy difference of neighboring filter-bank and probabilistic characteristics of music. Since Philips method uses too many filter-banks in limited frequency band, it may cause audio fingerprints to be highly sensitive to additive noises and to have too high correlation between neighboring bands. The proposed method improves robustness to noises by reducing the number of filter-banks while it maintains the discriminative power by representing the energy difference of bands with 2 bits where the quantization levels are determined by probabilistic characteristics. The correlation which exists among 4 different levels in 2 bits is not only utilized in similarity measurement. but also in efficient reduction of searching area. Experiments show that the proposed method is not only more robust to various environmental noises (street, department, car, office, and restaurant), but also takes less time for database search than Philips in the case where music is highly degraded.

• The Journal of the Acoustical Society of Korea
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• v.43 no.2
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• pp.184-199
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• 2024

Residual crosstalk has been considered as a major drawback of conventional time-reversal processing in the case of simultaneous multiple focusing. In this paper, the Gram-Schmidt process is applied to time-reversal processing to mitigate crosstalk in ocean waveguides for multiple probe sources. Experimental data-based numerical simulations confirm that nulls can be placed at multiple locations, and it is shown that different signals can be simultaneously focused at different probe source locations, ensuring distortionless responses in terms of active time-reversal processing. This focusing property is also shown to be much less affected by a reduction in the number of receivers than the adaptive time-reversal mirror method. The proposed method is shown to be effective in eliminating crosstalk in passive multi-input multi-output communications using sea-going data.