• 한국음향학회:학술대회논문집
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• 한국음향학회 1991년도 학술발표회 논문집
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• pp.129-132
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• 1991

In this paper, the active noise controll system in duct is analyzed with real time implementation. The primary noise signal detected by microphone is modeled using adaptive algorithm and the secondary signal which has the same amplitude and 180$^{\circ}$phase shift with the primary noise signal is generated in the controller. We used the DSP56001 as a real-time processor and LMS algorithm as a adaptive technique, the experimental results shows that our system can reduce the noise level in duce to 15~40[db].

• 전기학회논문지
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• 제58권4호
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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.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2011년도 춘계학술대회 논문집
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• pp.816-820
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• 2011

본 논문은 구형 마이크로폰 어레이의 부엽 레벨의 차를 증가시키기 위한 방법에 대한 연구 내용을 다루었다. 일반적인 어레이 신호처리에서 마이크로폰을 조밀하게 배치함으로써 어레이 응답에서의 주엽과 부엽 간의 차이를 늘릴 수 있고 어레이의 소음원 판별능력을 증가시킨다. 최근 사용되고 있는 상용 에레이들은 제작 단가와 어레이의 크기 때문에 센서의 수를 늘리는데 한계를 보이고 있다. 이런 문제를 극복하기 위해 본 연구에서는 MEMS 센서를 이용하여 구형 어레이에 적용하였다. 구형 마이크로폰 어레이를 이용한 시뮬레이션과 실험을 통해 정현파 소음원을 측정하였다. 실험을 위해 32 개의 일반 측정용 마이크로폰을 이용한 어레이와 85 개의 MEMS 마이크로폰을 이용한 구형 어레이를 제작하였다. 구형 조화 분해기법과 빔형성기법을 이용하여 측정 데이터를 분석하였다. 2 kHz 이상의 소음원에 대하여 MEMS 마이크로폰 어레이가 4 dB 이상의 부엽 저감 능력을 가지는 것을 확인하였다.

• 디지털산업정보학회논문지
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• 제12권1호
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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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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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.

• 한국음향학회지
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• 제16권5호
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• pp.76-82
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• 1997

본 논문은 음원의 절대위치를 정확하게 추정할 수 있는 음향홀로그래픽법에 관하여 계산기상의 시뮬레이션 및 측정시스템을 이용한 실험과결과에 대하여 서술한다. 이 연구에서는 원거리 음장을 만족하도록 측정면을 설정하여 7개의 마이크로폰을 직선으로 배열한다. 음원의 측정은 음원면에 근접한 위치에 한 개의 기준 마이크로폰을 설치하고 측정면의 마이크로폰들을 등간격으로 스캐닝하면서 각지점의 음을 동시 기록한다. 수음한 기준음과 측정음간의 크로스 스펙트럼 알고리즘에 의하여 음원의 절대위치를 측정한다. 그리고 각 마이크로폰의 위상차는 기준 마이크로폰을 대상으로 위상보상 하였으며, 측정시의 시간지연은 제 1열 측정시점을 기준으로 시간보상을 행하였다. 측정면에 설정한 마이크로폰들의 최적 간격은 수치 시뮬레이션에 의하여 정한다. 음원신호는 정현파를 이용하고 S/N비를 30dB의 조건하에서 각각 실험을 행하였다. 시뮬레이션과 실험에서 결정한 최적 마이크로폰 간격은 2KHz인 정현파 음원을 기준으로 하여 공간상의 나이키스트 조건을 만족하도록 설정하였다. 무향실에서 측정한 실험결고, 500Hz 와 1KHz의 신호원에 대한 음원이 2KHz인 경우의 추정된 3차원 홀로그램의 주극폭이 각각 87%와 30%씩 감소하였고, 그 결과 수치 시뮬레이션의 타당성을 확인할 수 있었다. 그러므로 본 연구에서 제안하는 3차원 음향 홀로그래픽법을 이용한 음원위치 추정에 관한 연구의 유용성을 검증하였다.

• 한국정밀공학회:학술대회논문집
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• 한국정밀공학회 2008년도 추계학술대회 논문집
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• pp.687-688
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• 2008

• 한국멀티미디어학회논문지
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• 제18권5호
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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.

• 전자공학회논문지B
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• 제30B권10호
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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.