• The Journal of the Korea Contents Association
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• v.10 no.12
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• pp.27-33
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• 2010

In this study, we conducted the classification experiments with GMM (Gaussian Mixture Model) from combining the extracted features by using microphone, Gyro sensor and Acceleration sensor in 9 different environment types. Existing studies of Context Aware wanted to recognize the Environment situation mainly using the Environment sound data with microphone, but there was limitation of reflecting recognition owing to structural characteristics of Environment sound which are composed of various noises combination. Hence we proposed the additional application methods which added Gyro sensor and Acceleration sensor data in order to reflect recognition agent's movement feature. According to the experimental results, the method combining Acceleration sensor data with the data of existing Environment sound feature improves the recognition performance by more than 5%, when compared with existing methods of getting only Environment sound feature data from the Microphone.

• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1573-1580
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• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.46 no.12
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• pp.975-985
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• 2018

The drones have been developed for military purposes and are now used in many fields such as logistics, communications, agriculture, disaster, defense and media. As the range of use of drones increases, cases of abuse of drones are increasing. It is necessary to develop anti-drone technology to detect the position of unwanted drones using the physical phenomena that occur when the drones fly. In this paper, we estimate the DOA(direction of arrival) of the drone by using the acoustic signal generated when the drone is flying. In addition, the dynamics model of the drones was applied to the unscented kalman filter to improve the microphone array detection performance and reduce the error of the position estimation. Through simulation, the drone detection performance was predicted and verified through experiments.

• Journal of Sensor Science and Technology
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• v.28 no.6
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• pp.371-376
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• 2019

We have developed an acoustic resonance inspection system to inspect the welding defects in the mechanical parts fabricated using friction stir welding method. The inspection system was consisted of a DAQ board, a microphone sensor, an impact hammer, and controlled by a PC software. The system was developed to collect and analyze the sound signal generated by hitting the sample with an impact hammer to determine whether it is defective. In this study, 100% welded good samples were compared with 95%, 90%, and 85% welded samples, respectively. The variation of the completeness in welding did not affect the visual appearance in the samples. As a result of analyzing the natural frequencies of the good samples, the five natural frequency peaks were identified. In the case of the defective samples, the frequency change was observed. The welding failure detection time was fast enough to be only 0.7 seconds. Employing our welding defect inspection system to the actual industrial field will maximize the efficiency of quality inspection and thus improve the productivity.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4E
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• pp.52-60
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• 2001

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.

• International journal of advanced smart convergence
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• v.1 no.2
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• pp.20-25
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• 2012

The oscillometric method and Korotkoff sound method are the most common ways to measure the blood pressure. A new automatic blood pressure measurement device, which uses both oscillometric method and Korotkoff method, was developed. A pressure sensor was used to obtain cuff pressure and oscillation signal, and a microphone was used to detect Korotkoff sounds. Forty-five measurements from fifteen subjects were used for analysis. Correlation coefficients between the traditional auscultatory method and Korotkoff sound method were 0.9820 and 0.9721 for the systolic and diastolic blood pressure values, respectively. Standard deviations of differences for the systolic and diastolic blood pressure values were 1.3019 and 1.4495, respectively. Correspondingly, correlation coefficients between the traditional auscultatory method and oscillometric method using newly developed algorithm were 0.9651 and 0.9136 for the systolic and diastolic blood pressure values, with the standard deviations of 1.42 and 1.73, respectively. The results showed that the newly developed algorithm for oscillometirc method provide accurate blood pressure values, moreover, Korotkoff sound method using microphone provides even higher accuracy. Therefore, a new automatic device which utilizes both oscillometric method and Korotkoff sound method would provide the accurate and reliable blood pressure values.

• Proceedings of the KIEE Conference
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• 2003.11c
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• pp.878-881
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• 2003

To acquire an appointed speaker's clear voice signal from inspect-camera, picture-conference or hands free microphone eliminating interference noises needs to be preceded speaker's position automatically. Presumption of sound source position's basic algorithm is about measuring TDOA(Time Difference Of Arrival) from reaching same signals between two microphones. This main project uses ADF(Adaptive Delay Filter) [4] and CPS(Cross Power Spectrum) [5] which are one of the most important analysis of TDOA. From these analysis this project proposes presumption of real time sound source position and improved model NI-ADF which makes possible to presume both directions of sound source position. NI-ADF noticed that if auditory sense of humankind reaches above to some specified level in specified frequency, it will accept sound through activated nerve. NI-ADF also proposes practicable algorithm, the presumption of real time sound source position including both directions, that when microphone loads to some specified system, it will use sounds level difference from external system related to sounds of diffraction phenomenon. In accordance with the project, when existing both direction adaptation filter's algorithm measures sound source, it increases more than twice number by measuring one way. Preserving this weak point, this project proposes improved algorithm to presume real time in both directions.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.40-50
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• 1994

In this paper, the sensitivity compensation method for three-dimensional acoustic intensity probe in the higher frequency range has been studied. The measurement error in the higher frequency range is generated from the phase mismatch between microphone's signals of the probe. If the wavelength of sound signal measured is less than those of the distance between microphones of the probe, that is, the higher frequency of the sound signal, the bigger measurement error is generated. In this study, we proposed the compensation methods for one-dimensional acoustic intensity probe with two-microphones, and the efficiency of those methods were investigated by numerical calculation of computer. It was most effective method to compensate the phase mismatch between microphone for the acoustic intensity probe was investigated for the sound estimated. and the efficiency of this method in a three-dimensional probe was investigated for the sound wave travelling in the arbitrary direction by numerical calculation of computer. In this result, the efficiency was proved that, for the measurement error of 1dB or less with the three-dimensional probe of 60mm space, the frequency should be less than 1.2kHz without the error compensation method, but the frequency increased up to 2.8kHz with the error compensation method.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.101-105
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• 1997

The active noise control method that was developed for long duct has some problems to be applied for short duct. To overcome this problem, we apply the SISO(Single Input Single Output) algorithm for the active noise control of short duct using zero acoustic impedance boundary. The SISO algorithm can input noise signal and error signal with one microphone simultaneously. The real-time controller was implemented using TMS320E25 DSP(Digital Signal Processing) chip and it's performance was evaluated by experiment. As a result, we obtain total 4.7dBA noise reduction for 0.80m short duct.