• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.4
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• pp.857-866
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• 2009

In this paper, a noise cancellation-method using microphone array for digital hearing aids is proposed. The microphone array is located around the ear of a dummy. Speech sound is generated from the forward speaker positioned in the front of the dummy and noise sound is generated from the backward speaker. The speech and noise are mixed in the air space and entered into the microphones. VAD(voice activity detector) and ANC(adaptive noise cancellation) methods were used to eliminate noise in the sound of the microphones. 10 two-syllable words and 4 sentences were used for speech signals. Babble and car interior noise were used for noise signals. The performance of the proposed algorithm was evaluated by SNR(signal-to-noise ratio) and PESQ-MOS(perceptual evaluation of speech quality-mean opinion score). In babble noise condition, SNR was improved as much as $7.963{\pm}1.3620dB\;and\;3.968{\pm}0.6659dB$ for words and sentences respectively. In the case of car interior noise, SNR was improved as $10.512{\pm}2.0665dB\;and\;6.000{\pm}1.7642dB$ for words and sentences respectively. PESQ-MOS of the babble noise was improved as much as $0.1722{\pm}0.0861$ score for words and $0.083{\pm}0.0417$ score for sentences. And PESQ-MOS of the car interior noise was improved as $0.2661{\pm}0.0335$ score and $0.040{\pm}0.0201$ score for words and sentences respectively. It is verified that the proposed algorithm has a good performance in noise cancellation of microphone array for digital hearing aids.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2011.04a
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• pp.816-820
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• 2011

A method for increasing the difference of side-lobe level in spherical microphone array is presented. In array signal processing, it is known that narrow interval between sensors can increase the difference between main lobe and side-lobe of array response which eventually increase the source recognition capability. Recent commercial array being used, however, have shown certain limitation in using the number of sensors due to its costs and geometrical size of array. To overcome this problem, we have adapted MEMS sensors into spherical microphone array. To check out the improvement, two different types of spherical microphone array were designed. One array is composed with 32 regular instrument microphones and the other one is 85 MEMS sensors. Simulation and experiments were conducted on a sinusoidal noise source with two arrays. The time history data were analyzed with spherical harmonic decomposition and beamforming technique. 85 MEMS sensors array showed the improved side-lobe level suppression by more than 4 dB above the frequency content of 2 kHz compared to 32-sensor array.

• Journal of Korea Society of Digital Industry and Information Management
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• v.12 no.1
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• pp.43-50
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• 2016

This paper is on an acoustic echo canceller solving the double-talk problem by using the blind signal separation technology. 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. The blind signal separation extracts the near-end signal from dual microphones by the iterative computations using the 2nd order statistical character in the closed reverberation environment. By this method, the acoustic echo canceller operates irrespective of the double-talking. We verified performances of the proposed acoustic echo canceller in the computer simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods well, and then operates stably without diverging of the coefficients after ending the double-talking. The merits are in the simplicity and stability.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.363-364
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• 2006

The direction of arrival of the sound signal can be derived from the time differences at the microphone array and the motor controls the camera to point at the direction of the sound signal. You can get through to the homepage and confirm the camera image on a client computer which connects to the server computer through Internet.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.341-342
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• 2006

The direction of arrival of the sound signal can be derived from the time differences at the microphone array and the motor controls the camera to point at the direction of the sound signal. You can get through to the homepage and confirm the camera image on a client computer which connects to the server computer through Internet.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.76-82
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• 1997

The paper describes simulations and experimental results using a measuring system which utilizes the acoustic holographic method in order to exactly estimate an absolute position of a sound source. The measuring surface is installed to satisfy with a far field to the sound source and is composed of linear arrayed seven microphones. A measurement is simultaneously recorded by a reference microphone setting up a neighbour sound source and the linear arrayed seven microphones which are moved to the same interval. An absolute position of sound source is estimated by the cross-spectrum method to the received sounds between a reference and the measuring microphones. Phase differences of each microphone and time delays during scanning are compensated to the reference microphone and the measuring time of the first column. An optimal interval for each microphone in the measuring surface is decided by a numerical simulation. A source signal makes use of a sinusoid, and S/N ratio is 30dB in the experiment. The optimal microphone's interval in the simulation and the experiment is decided in order to satisfy with the Nyquist space sampling condition related to the wave length of 2kHz sinusoid. Mainlobe width of a estimated 3D hologram in the case of 2kHz source signal is decreased to 87% and 30% in comparison to 500Hz and 1kHz, and then a valid of simulation results is confirmed. Therefore, we verified a utilization of the study for a sound source estimation using 3ㅇ acoustic holographic method.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2008.11a
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• pp.687-688
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• 2008

• Journal of Korea Multimedia Society
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• v.18 no.5
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• pp.652-662
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• 2015

It is necessary to develop small, implantable bio-telemetry systems which can measure and transmit patients' bio-signals from internal body to external receiver. When measuring bio-signals, like electrical bio-signals, acoustic bio-signal measurement has also a big clinical usefulness. But, sound signal has larger frequency bandwidth than any other bio-signals. When considering these issues, a wireless telemetry system which has rapid data transmission rate proportional to wide frequency bandwidth is necessary to be developed. The bluetooth module is used to overcome the data rate limitation caused by the large frequency bandwidth. In this paper, a novel multimedia bluetooth biotelemetry system was developed which consists of transmitter module located in the body and receiver device located outside of the body. The transmitter consists of microphone, bluetooth, and wireless charging device. And the receiver consists of bluetooth and codec system. The sound inside the skin is captured by microphone and sent to receiver by bluetooth while charging. The wireless charging system constantly supplies the electric power to the system. To verify the performance of the developed system, an in vitro experiment has been performed. The results show that the proposed biotelemetry system has ability to acquire the sound signals under the skin.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.10
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• pp.35-41
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• 1993

In this paper, a new speech enhancement method for speech signal corrupted by environmental noise is proposed. Two signals are obtained from the microphone and from the accelerometer attached to the neck, respectively. Since two signals are generated from same source signal, both signals are closely correlated. And environmental noise has no effect on the accelerometer signal. The speech enhancement system identifies the optimum linear system between two signals on the basis of the dependence between the signals. The enhanced speech can be obtained by filtering the noise-free accelerometer signal. Since the characteristcs of the speech signal and environmental noise are changing with time, adaptive filtering system has to be used for characterizing the time-varing system. Simulation results show 7dB enhancement with 0dB speech signal level relative to the white noise.

• Journal of the Korean Society of Manufacturing Technology Engineers
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• v.23 no.4
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• pp.325-329
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• 2014

This paper proposes a method to predict chatter vibration generated in milling processes and to enhance machining quality and surface finish. Chatter vibration is a common problem in the milling of thin walls and floors. It causes a poor surface finish, or even marks, to appear on the final machined surface. Therefore, an effective method is necessary to predict chatter vibration in machine tools. In this investigation, chatter vibration is measured by an accelerometer, microphone, and Acoustic Emission (AE) sensor in a machining operation. Based on the results of the experiment, a microphone can be applied for the prediction of chatter vibration in milling processes.