• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.6 no.1
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• pp.23-28
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• 2013

In this paper, background noise reduction method using dual microphone is proposed in mobile environment. Each Signal, reference and primary, would be replaced by microphone input signals, which were measured by reference and primary microphones, and then, noise reduction was performed using FDAF. After then, residual and background noise would be estimated and reduced by MMSE-LSA. For consistent noise reduction performance, result of VAD that could be caculated by PLD between two microphones was used.

• Journal of Biomedical Engineering Research
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• v.29 no.1
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• pp.59-65
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• 2008

Nowadays, implantable hearing aids have been developed to solve the problems of conventional hearing aids. In case of fully implantable hearing aids, an implantable microphone is necessary to receive sound signal beneath the skin. Normally, an implantable microphone has poor frequency response characteristics in high frequency bands of acoustic signal due to the high frequency attenuation effect of skin after implantation to human body. In this paper, the implantable microphone is designed to reduce the high frequency attenuation effect of a skin by putting its resonance frequency at the attenuated range through a finite element analysis (FEA) simulation. The designed implantable microphone through the simulated results has been fabricated by manufacturing process using bio-compatible materials. By the several in-vitro experiments with pig skin, it has been verified that the designed implantable microphone has a resonance frequency around the starting part of the attenuated range and reduces the attenuation effect.

• ETRI Journal
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• v.36 no.5
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• pp.772-782
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• 2014

Two-microphone binary mask speech enhancement (2mBMSE) has been of particular interest in recent literature and has shown promising results. Current 2mBMSE systems rely on spatial cues of speech and noise sources. Although these cues are helpful for directional noise sources, they lose their efficiency in diffuse noise fields. We propose a new system that is effective in both directional and diffuse noise conditions. The system exploits two features. The first determines whether a given time-frequency (T-F) unit of the input spectrum is dominated by a diffuse or directional source. A diffuse signal is certainly a noise signal, but a directional signal could correspond to a noise or speech source. The second feature discriminates between T-F units dominated by speech or directional noise signals. Speech enhancement is performed using a binary mask, calculated based on the proposed features. In both directional and diffuse noise fields, the proposed system segregates speech T-F units with hit rates above 85%. It outperforms previous solutions in terms of signal-to-noise ratio and perceptual evaluation of speech quality improvement, especially in diffuse noise conditions.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.18 no.1
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• pp.26-32
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• 2007

Objective and Background: Laryngeal stroboscope is an useful equipment in evaluation of vocal cord vibration and in early detection of mucosal lesion including invasive cancer of the vocal cord. Recently Lee et al. (2006) developed portable stroboscope using voice as synchronization signal. It has been frequently impaired ability to synchronize the flashes even in normal female. Authors tried to investigate various methods including vibration pick-up, microphone, laryngeal microphone, and contact microphone for development of simple and accurate method like electroglottograph signal. The purpose of this study was to estimate wheher the vibration pick-up is available and is consistent with the signal of EGG. Subjects and Methods: Authors compared the signals between EGG and noncontact method such as voice, contact methods including vibration pick-up, laryngeal microphone, and contact microphone in normal twenty adults (male 10 and female 10). The number of peak in one cycle was compared with the number of the peak in EGG, and the percent of phase difference in the peak was compared with EGG Also, authors tried to investigate which site of vibration pick-up was most effective for synchronization of stobo flashes. Three site including anterior neck below the cricoid cartilage, thyroid ala, and suprahyoid region were analysed. Results: Among various methods for synchronization of strobo flashes, vibration pick-up was most effective method in peak detection. And anterior neck below cricoid cartilage was the most available site of the vibration pick-up. Conclusion: Authors suggest that vibration pick-up is most available and effective method for synchronization of strobo flashes.

• Proceedings of the KIEE Conference
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• 1998.07b
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• pp.654-656
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• 1998

This paper deal with the implementation of Active Noise Control (ANC) in a short duct. In case of ANC in the air duct, input microphone, control speaker, error microphone are used. But we can't use input microphone because of the characteristics of short duct. It is difficult to avoid howl. So we propose single-channel adaptive feedback ANC which is composed only error microphone and control speaker without input microphone. FXLMS algorithm is used to compensate for the time delay of the error path. Experimental results show that the controller reduce noise signal sufficiently.

• KEPCO Journal on Electric Power and Energy
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• v.1 no.1
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• pp.99-104
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• 2015

In this study, it has been initiated to investigate the specific abnormal vibration signal that has been captured in the power equipment. Array Microphone can be used in order to detect the direction and the position of the noise source. It is possible to track the abnormal mechanical noise in the power plant by utilizing the program and the microphone array system developed from this research. Array microphone system can be operated as a constant monitoring system.

• The Journal of the Acoustical Society of Korea
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• v.24 no.2
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• pp.68-77
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• 2005

The purpose of a spatial audio system is to give a listener an impression as if he were present in a recorded environment when its sound is reproduced. For this purpose a dummy head microphone is generally used. Because of its human-like shape, dummy head microphone can reproduce spatial images through headphone reproduction. However, its shape and size are restriction to public use and it is difficult to convert the output signal of dummy head microphone into a multi-channel signal for multi-channel environment. So, in this paper, we propose a multi-channel 3D microphone technology. The multi-channel 3D microphone acquire a spatial audio using five microphones around a horizontal plane of a rigid sphere and through post processing, it can reproduce various reproduction signals for headphone, stereo, stereo dipole, 4ch and 5ch reproduction environments. Because of complex computation, we implemented H/W based post processing system. To verily the Performance of the multi-channel 3D microphone, localization experiments were Performed. The result shows that a front/back confusion, which is the one of common limitations of conventional dummy head technology, can be reduced dramatically.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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• pp.38-46
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• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.

• Journal of Advanced Marine Engineering and Technology
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• v.31 no.8
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• pp.990-997
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• 2007

In the last 5 years, collisions of fishing vessels have recorded about 54.6% of the total marine accidents. Specially about 64.0% of these collisions were caused by navigator's negligence of watch keeping during works. The purpose of this paper is to develop vessel detecting system that is able to receive the whistle blast of other vessel and make a warning sound and light when the fishermen can not confirm the approaching another vessel on account of fishing works. It is designed to receive the whistle signal blast by a weather tight microphone. The signal is processed by analog active filter in order to enhance the SNR(Signal to noise ratio). And this microprocessor-based system is programmed to do ADC(Analog to digital converting), FFT analysis, controls of warning sound and light.