• Proceedings of the KIEE Conference
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• 1999.07g
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• pp.3221-3223
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• 1999

OAEs are low-level sounds produced by cochlea as part of the normal hearing process. OAEs can be measured with a microphone placed in the outer ear-cannel. We can diagnose cochlea's condition by using OAEs. To diagonose it's condition, however, is difficult by reason of OAEs' tiny. Thus, It need to a method using latency which essential diagnosing a time-element of OAEs. This study proposes a latency detection algorithm using 7-QMF for more effective detection of latency, 7-QMF designed by wavelet theory can process signal without a losses of information. The latency-detector based on 7-QMF is superior to former method.

• The Journal of Korea Robotics Society
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• v.17 no.3
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• pp.273-281
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• 2022

In this paper, we propose a robot sound recognition system that detects various sound events. The proposed system is designed to detect various sound events in real-time by using a microphone on a robot. To get real-time performance, we use a VGG11 model which includes several convolutional neural networks with real-time normalization scheme. The VGG11 model is trained on augmented DB through 24 kinds of various environments (12 reverberation times and 2 signal to noise ratios). Additionally, based on random forest algorithm, a decision logic is also designed to generate event signals for robot applications. This logic can be used for specific classes of acoustic events with better performance than just using outputs of network model. With some experimental results, the performance of proposed sound recognition system is shown on real-time device for robots.

• The Journal of the Korea institute of electronic communication sciences
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• v.12 no.3
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• pp.419-426
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• 2017

This paper describes an acoustic echo canceller with double-talk using BSS(: Blind Signal Separation) based on the geometric concept. The acoustic echo canceller may be deteriorated or diverged during the double-talk period. So we use the blind signal separation to detect the double talking by separating the near-end speech signal from the mixed microphone signal. In the closed reverberation environment, the blind signal separation extracts the near-end signal from unknown signals with the transformation and rotation based on the geometric concept. By this method, the acoustic echo canceller operates irrespective of double-talking. We verified performances of the proposed acoustic echo canceller by computer simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods thoroughly, and operates stably in the normal state without diverging of coefficients after ending the double-talking.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7C
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• pp.413-420
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• 2011

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.05a
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• pp.81-84
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• 2007

In this paper, fiber optic sound and vibration monitoring sensor which is latticed shape structure based on Sagnac interferometer is fabricated and tested in laboratory conditions. To detect external vibrations surface mounted fibers on the latticed steel wire fence with a dimension of 170cm by 180cm is used. To detect external sound frequency the tightened fiber optic itself wire netting fence with a dimension of 50cm by 50cm is used. Experiments for the detection of the excited vibration and sound signals were performed. A small vibrator induced external vibration signal and it is applied to the latticed structure in the range of 100Hz to several kHz. External sound signal applied to the fiber optic sensor net using non-directional sound speaker. The detected optical signals were compared and analyzed to the detected both accelerometer and microphone signals in the time and frequency domain. Based on the experimental results, distributed fiber optic sensor using Sagnac interferometer detected effectively external vibration and sound signal and had a good performance. This system can be expanded to the monitoring of a significant system and to the structural health monitoring system.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.200-204
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• 2005

The use of beamforming method and de-Dopplerization technique was applied in studying the rotating sound sources. Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. Two main issues of the signal reconstruction in time domain are addressed herein: First, to remove Doppler effect from the measured data and to restore the original emission data of the moving source. The difference of the time domain beamforming from the frequency domain beamforming was mentioned. Also, the time domain beamforming method is deployed in the test and the comparisons were made to the frequency domain results. The time domain signal reconstruction was numerically simulated prior to the application. To validate the de-Dopplerization Performance, the rotating Point sources were examined and localized by the use of a phased array of microphone. The application of prop-rotor was conducted in a hovering condition. The results of reconstructing time signals of rotating sources and its locations were shown in the power distribution maps. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies of interest.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.34 no.10
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• pp.1507-1512
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• 2010

In this study, the plasma induced by $CO_2$ laser lap welding of 6t Zn coated steel used for ship building was measured using photodiodes and a microphone. Then, the welding phenomenon with gap clearance of lap joint was compared with RMS-treated signal. Thus, we found that intensity of the RMS-treated signal increased with Zn vaporization; further, the presence of defects results in rapid variations with the RMS value as a function of lap-joint parameters. Besides, the FFT value of the raw signal with variations of changing welding parameters was calculated, and then the calculated FFT frequency value was set as the bandwidth of digital filter for a more accurate in-process monitoring. The RMS values were acquired by filtering the raw signal. By matching the weld beads and the calculated RMS values, we confirmed that there is a strong relationship between the signals and the defects.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1004-1016
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• 2007

In this paper we proposed a Weight-Update-Control Adaptive Noise Canceller which improves speech when environmental noise is stationary and it is hard to acquire a reference signal. Adaptive Noise Canceller(ANC) needs a reference signal, but it is not easy to measure pure noise without voice for reference in factory. Because there are mixed various mechanical noise and workers' voice. Therefore ANC is not suitable to reduce background noise. So we proposed the method that uses an arbitrary constant as an input signal and inputs microphone signal to the reference signal. The noise is eliminated using updated weights in non-speech range. In speech range the weight is fixed and the modified voice is acquired then voice is restored through transversal filter. The proposed method is based on facts that the factory noise is stationary and the noise is not changed in short conversation range. As a result of simulation using MATLAB, we confirmed that the proposed method is effective for reducing factory noise and has high signal to noise ratio(SNR).

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.3
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• pp.671-677
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• 2012

A black-box program was implemented in order to monitor abnormal symptoms of human body irregularly occurring during sleep. The system consists of sensor probing body signals, auxiliary devices such as the alarm, lamp, network camera, and signal monitoring computer. Various types of sensors, PPG, ECG, EEG, temperature, respiration sensor, G-sensor, and microphone were used to more exactly identify the causes of abnormal symptoms. If a symptom occurs, the system records the patient's condition to provide information being utilized in the treatment. The sensors are attached on some locations of body being proper to check a specific type of abnormal reaction. Based on the normal range and type of measurement data, criteria of signal levels were set to distinguish abnormal reaction. An abnormal signal being probed, the program starts to operate the lamp, alarm, and network camera at the same time and stores the signal and video data.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.4
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• pp.111-119
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• 2012

Time Stretched Pulse (TSP) is used for transmitting and analyzing the impulse signal over the designated spatial place. However, if transfer functions of transmitter and receiver are unknown, performance investigation of free field in temporal domain is barely possible due to the overlap between the direct and indirect signal from the space. Generally, the free field or hemi-free field is evaluated by the Annex A of ISO 3745 in which utilizing the inverse square law with one-third octave band signals. In this paper, the author performs analysis of free field via applying TSP with inverse square law and the results are compared with the one-third octave band signals. According to the analysis of deviation between the corresponding signal and inverse square law model, the proposed TSP method provides the comparable performance index to the one-third octave band signal with reduced measuring time. Provided that the pre-whitening can be implementable by employing the speaker and microphone transfer function, further analyses from TSP compression are able to be performed such as multipath separation from time domain data. The anechoic chamber used in this experiment is verified conformance with ISO 3745 for free field and hemi-free field condition for limited frequency of the signal.