• Transactions of the Korean Society of Mechanical Engineers A
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• v.29 no.12 s.243
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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.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.28-36
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• 1995

In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.

• Journal of Korea Multimedia Society
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• v.2 no.1
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• pp.99-108
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• 1999

In this paper, a new robust adaptive algorithm is proposed to improve the performance of AEC without computational burden. The proposed adaptive algorithm is based on NLMS algorithm, and its step-size is varied with the reference input signal power and the desired signal power. Its step-size is normalized by the sum of the powers of the reference input signal and the desired signal. When the near-end speaker's speech and noise are applied into the microphone, the step-size becomes small and the misalignment of coefficients are reduced. The convergence speed is comparable to NLMS algorithm at AEC application because the echo signals are attenuated about 10∼20 dBSPL. The characteristics of this algorithm is also analyzed and compared with conventional ones in this paper.

• MALSORI
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• no.50
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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.

• Journal of the Korean Society of Manufacturing Technology Engineers
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• v.6 no.4
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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 .

• Transactions of the Korean Society of Automotive Engineers
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• v.6 no.1
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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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.05a
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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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• v.22 no.1
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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.

• Journal of the Korea Society of Computer and Information
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• v.20 no.12
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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.

• Proceedings of the IEEK Conference
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• 2000.07a
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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.