• 대한기계학회논문집A
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• 제29권12권
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• pp.1597-1604
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• 2005

The identification of a moving noise source is important in reducing the source power of the transport systems such as airplanes or high speed trains. However, the direct measurement using a microphone running with noise source is usually difficult due to wind noise, white the source motion distorts the frequency characteristics of the pass-by sound measured at a fixed point. In this study the relationship between the spectra of the source and the pass-by sound signal is analyzed for an acoustic source moving at a constant velocity. Spectrum of the sound signal measured at a fixed point has an integral relationship with the source spectrum. Nevertheless direct conversion of the measured spectrum to the source spectrum is ill-posed due to the singularity of the integral kernel. Alternatively a differential equation approach is proposed, where the source characteristics can be recovered by solving a differential equation relating the source signal to the distorted measurement in time domain. The parameters such as the source speed and the time origin, required beforehand, are also determined only from the frequency-phase relationship using an auxiliary measurement. With the help of the regularization method, the source signal is successfully recovered. The effects of the parameter errors to the estimated frequency characteristics of the source are investigated through numerical simulations.

• 한국음향학회지
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• 제14권3호
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• pp.28-36
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• 1995

본 논문에서는 능동소음제어를 위하여 망각형 지연 LMS(least mean square) 알고리듬을 이용한 이중루프제어 모델을 제안하였다. 제안한 이중루푸제어 모델은 계산양을 줄이기 위해 이득과 음향시간지연인자를 이용하여 온라인으로 라우드스피커 특성 및 오차음경로를 추정한다. 이중루프구조를 통한 오차신호의 제어는 보다 견실한 제어시스템이 된다. 음향시간지연을 추정하기 위한 필터의 입력신호는 입력 마이크로폰 신호와 적응필터의 차를 사용한다. 망각형 지연 LMS 알고리듬은 비정상상태에서 계수들의 발산을 방지하기 위해 사용한다. 실제 소음신호에 대하여 제안한 이중루프제어 모델은 소음레벨이 평균 12.9dB정도 감쇠되었다.

• 한국멀티미디어학회논문지
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• 제2권1호
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• pp.99-108
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• 1999

본 논문에서는 계산량이 크게 증가하지 않으면서 음향 반향 제거기의 성능을 향상시키는 잡음에 강건한 알고리즘을 제안한다. 제안하는 알고리즘은 NLMS 알고리즘을 기반으로 적응 상수를 기준 입력신호와 마이크 입력신호의 전력합으로 정규화한다. 마이크 입력으로 근단화자의음성신호와 잡음이 입력되면 그 전력만큼 적응 상수가 작아지기 때문에 이들 주변 잡음신호에 의한 계수 오조정을 줄일 수있다. 따라서 제안하는 알고리즘은 동시통화와 주변 잡음이 있는 상황에서 반향 제거 기능을 잘 수행한다. 제안하는 알고리즘의 수렴 속도는 반향 신호가 10~20 dBSPL 정도 감쇠되기 때문에 NLMS 알고리즘에 비해서 크게 느려지지 않는다. 본 논문에서는 제안하는 알고리즘의 성능을 분석하고 기존의 알고리즘과 비교한다.

• 대한음성학회지:말소리
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• 제50호
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• pp.71-83
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• 2004

Recently, broadcast news speech recognition has become one of the most attractive research areas. If we can transcribe automatically the broadcast news and store their contents in the text form instead of the video or audio signal itself, it will be much easier for us to search for the multimedia databases to obtain what we need. However, the desirable speech signal in the broadcast news are usually affected by the interfering signals such as the background noise and/or the music. Also, the speech of the reporter who is speaking over the telephone or with the ill-conditioned microphone is severely distorted by the channel effect. The interfered or distorted speech may be the main reason for the poor performance in the broadcast news speech recognition. In this paper, we investigated some methods to cope with the problems and we could see some performance improvements in the noisy broadcast news speech recognition.

• 한국생산제조학회지
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• 제6권4호
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• pp.101-107
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• 1997

In order to detect and suppress chatter in turning process, a stability control methodology was studied through manipulation of spindle speeds regarding to chatter frequencies, The chatter frequency was identified by monitoring and signal processing of sound pressure during turing on a lathe. The stability control methodology can select stable spindle speeds without knowing a prior knowledge of machine compliances and cutting dynamics. Reliability of the developed stability control methodology was verified through turing experiments on an engine lathe. Experimental results show that a microphone is an excellent sensor for chatter detection and control .

• 한국자동차공학회논문집
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• 제6권1호
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• pp.36-42
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• 1998

In this study, active muffler was designed and was manufactured for exhaust noise reduction of commercial vehicle, then experiment of real vehicle was conducted. In a manufactured active muffler, because the flow of exhaust noise in tail pope outlet are become a plane wave, the global reduction if radiation noise from outlet and the placement of error microphone to avoid the exhaust gas of high temperature could be implemented. In control algorithm, reduction of noise of engine driving frequency and harmonic frequency can be archieved using proposed reference signal including a fixed speed state(2,000rpm, 3,000rpm, 4,000rpm) and a run-up speed state(2,000rpm$\rightarrow$4,000rpm) is accomplished with the active muffler installed in vehicle.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 춘계학술대회논문집
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• pp.837-842
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• 2004

The positions of rotating sound sources have been localized by experiments with the Doppler effects removed. In order to do-Dopplerize the sound signals emitted from moving sources, two kinds of signal reconstruction methods were applied. One is the forward propagation method and the other is the backward propagation method. Forward propagation method analyze the source emission time based on the instantaneous distance between sensors and the assumed source position, then the signals are reconstructed with respect to the emission time. On the other hand, the backward method uses time delay to do-Dopplerize the acquired data for the received time of reference. In both techniques, the reconstructed signal data were processed using beamforming algorithm to produce power distributions at the frequency of interest. Experiments have been carried out for varying frequencies, rotating speeds and the object distances. Forward propagation method has shown better performance in locating source position than the backward propagation method.

• Journal of Biosystems Engineering
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• 제22권1호
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• pp.41-48
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• 1997

A nondestructive quality inspection technique using acoustic impulse response method was studied to investigate the feasibility of egg shell inspection. An experimental system was built to generate impact force, to measure the response signal and to analyze the frequency spectrum. This system includes an impulse generating unit, an egg holding seat, a microphone with preamplifier, and a digital oscilloscope connected to Personal Computer by RS-232C interface. The factors such as impulse generating method, egg holding method, and sensor location were evaluated by analyzing the power spectrum density of the measured signal. The results obtained are summarized as follows : 1. From the sampled eggs, the proper conditions for detecting damaged eggs were found as followings; ceramic for the impact ball material, rubber for egg seat material, 20 degrees for an impact angle of pendulum, 10mm for the distance between egg and sensor, the sharp side for impacting part, and 180 degrees for the location of sensor. 2. Examination of the Fourier transformed analysis in beth normal and damaged eggs revealed that those factors such as the resonant frequency, a number of peak frequencies and the magnitude of power spectrum were important to detect damaged eggs.

• 한국컴퓨터정보학회논문지
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• 제20권12호
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• pp.83-90
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• 2015

In this paper, we propose a new method for multi transmission method of data among near smart devices. Existing methods require the use of an extra application service where the operating system differs between smart devices. In contrast, the proposed method makes use of the smart device's inner speaker and microphone to confirm the transmission signal. Then, real sharing data is transmitted via WiFi or LTE. Therefore, the proposed method overcomes operating system issues with existing methods. Besides, BUMP technology works in a similar way to the method we propose, it only supports one-to-one transmission. To evaluate the efficacy of the new method, we tested one-to-many data transmission in an experiment: the results showed a 96% success rate. As a result, we believe that the proposed method is an effective multi transmission method of data among near smart devices.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 ITC-CSCC -1
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• pp.225-228
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• 2000

It is well known that a Frequency Domain Adaptive Filter (FDAF) converges faster than a Time Domain Adaptive Filter (TDAF) even when the input signal is colored such as a speech signal. We have proposed the FDAF using the Modified Discrete Fourier Transform Pair (MDFTP) and its realization and effectiveness has been confirmed through the computer simulations. In this paper, we apply the FDAF using the MDFTP to the noise cancellation system. The proposed system is based on the Adaptive Line Enhancer (ALE) and utilizes single microphone; therefore it is suitable for the portable electronic equipment. Moreover, we propose to utilize the MDFT for detecting of the pitch in the speech because the number of data points in the MDFT must be equal to the pitch to confirmed that the noise can be removed to near the level of SNR.