• Journal of the Institute of Convergence Signal Processing
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• v.7 no.3
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• pp.130-135
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• 2006

There have been experiments dealing with the possibility of the actualization of the ANC system by means of operating the DSP adaptation filter. This filter is composed of various filters(including X-LMS algorithm, Filter-U algorithm, and Full-Feedback-Filter-U algorithm) that use ventilation fans and loudspeakers as a primary source in a circular duct as an experimental device. In this operation, the ANC system using the X - LMS algorithm was found to be more effective in reducing noise than without such system. When applying the input signal of the DSP board Full Feedback-Filtered-U algorithm system while having in mind that the additionally installed second control signal was gone through feedback and mixed into the detection microphone installed near the ventilation fan when using the first ventilation fan, the system was not emitted, but maintained stable during the operation of the control filter. At this point, noise tended to decrease at a maximum of l0dB compared to other algorithms at the frequency band of 170-250Hz.

• Journal of the Korea Institute of Military Science and Technology
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• v.18 no.2
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• pp.115-124
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• 2015

A methodology is proposed to estimate a trajectory of a flying target and its velocity using the time and frequency analysis of the acoustic signal. The measurement of sound emitted from a flying acoustic source with a microphone above a ground shall receive both direct and ground-reflected sound waves. For certain frequency contents, the destructive interference happens in received signal waveform reflected path lengths are in multiple integers of direct path length. This phenomenon is referred to as the acoustical mirror effect and it can be observed in a spectrogram plot. The spectrogram of acoustic measurement for a flying vehicle measurement shows several orders of destructive interference curves. The first or second order of curve is used to find the best approximate path by using nonlinear least-square method. Simulated acoustic signal is generated for the condition of known geometric of a sensor and a source in flight. The estimation based on cepstrogram analysis provides more accurate estimate than spectrogram.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.35-43
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• 1999

In this paper, a new robust adaptive algorithm and a post-processing method are proposed to improve the performance of AEC without computational burden. 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. To reduce the residual echoes, a new post-processing method, which is co-operated with the proposed noise-robust adaptive algorithm, is proposed in this paper. The method is based on the correlation of the desired signal and the estimation error signal. The residual echoes are attenuated as proportional to the correlation normalized with the power of desired signals. The normalized correlation plays a role as Wiener filter for residual echoes. In the double-talk situation, the estimation error signals, that are residual echoes, dominantly include the near-end speaker's speech and the normalized correlation closes to 1. Therefore, the near-end speaker's speech can be transmitted without being attenuated. When the desired signals consists of only the acoustic echoes, the residual echoes are mostly attenuated and canceled by the proposed post-processor. The computation of AEC using the proposed post-processor is comparable to NLMS algorithm.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.04a
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• pp.681-688
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• 2009

This paper deals with a bearing faults localization technique based on holographic approach by visualizing sound radiated from the faults. The main idea stems from the phenomenon that bearing faults in a moving vehicle generate impulsive sound. To visualize fault signal from the moving vehicle, we can use the moving frame acoustic holography [H.-S. Kwon and Y.-H. Kim, "Moving frame technique for planar acoustic holography," J. Acoust. Soc. Am. 103(4), 1734-1741, 1998]. However, it is not easy to localize faults only by applying the method. This is because the microphone array measures noise (for example, noise from other parts of the vehicle and the wind noise) as well as the fault signal while the vehicle passes by the array. To reduce the effect of noise, we propose two ideas which utilize the characteristics of fault signal. The first one is to average holograms for several frequencies to reduce the random noise. The second one is to apply the partial field decomposition algorithm [K.-U. Nam, Y.-H. Kim, "A partial field decomposition algorithm and its examples for near-field acoustic holography," J. of Acoust. Soc. Am. 116(1), 172-185, 2004] to the moving source, which can separate the fault signal and noise. Basic theory of those methods is introduced and how they can be applied to localize bearing faults is demonstrated. Experimental results via a miniature vehicle showed how well the proposed method finds out the location of source in practice.

• Journal of IKEEE
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• v.14 no.1
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• pp.47-54
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• 2010

In this study, we analyzed the reduction of noise in a noise environment using 2, 3, 4 or 5 microphones in digital hearing aids. In order to be able to use this in actual digital hearing aids, we made the experiment microphone set similar to the behind-the-ear type (BTE) and then recorded the signal accordingly, with each situation. With the recorded signals, we reduced the noise in each signal by a noise reduction algorithm using multi-microphones. As a result, in the case of By comparing the SNR (Signal to Noise Ratio) and PESQ (Perceptual Evaluation of Speech) measurements, before and after the noise reduction, the results showed that the improvement in performance was highest when three or four microphones were used. Generally, when two or more microphones were used, we found that as the number of microphones increased there was an increase in performance.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.17 no.5
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• pp.134-140
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• 2003

In this paper, a hardware implementation of gas flow meter for accuracy improvement and saving repair costs at a field is investigated. An adaptive filter using LMS algorithms for estimating cavity vibration frequencies in noisy environments is also studied. The proposed cavity gas flow meter measures cavity sound signals in gas flow tube using microphone and signal processing systems estimate the cavity vibration frequency from the measured signal. The flow velocity and flow quantity can be calculated using the estimated cavity vibration frequency. Since cavity vibration frequency is corrupted by the environmental noise, an adaptive filter using NLMS algorithms is used for cancelling the environmental noise. Experiments using 1MS32OC32 digital signal processor are performed to show the effectiveness of the proposed system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.2
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• pp.294-300
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• 2017

This paper proposes a method for 3-D sound source localization (SSL) using region selection and TDOA. 3-D SSL involves the estimation of an azimuth angle and an elevation angle. With the aim of reducing the computation time, we compare signal energies to select one out of three regions. In the selected region, we compute only one TDOA value for the azimuth angle estimation. Also, to estimate the vertical angle, we choose the higher energy signal from the selected region and pair it up with the elevated microphone's signal for TDOA computation and elevation angle estimation. Our experimental results show that the proposed method achieves average error values of $0.778^{\circ}$ in azimuth and $1.296^{\circ}$ in elevation, which is similar to other methods. The method uses one energy comparison and two TDOA computations therefore, the total processing time is reduced.

• The Journal of the Acoustical Society of Korea
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• v.28 no.5
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• pp.461-470
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• 2009

We investigate and analyze the problems encountered in frame-based estimation of TDOA (Time Difference of Arrival) using CPSP function. Spectral leakage occurring in framing of a speech signal by a rectangular window could make estimation of CPSP spectrum inaccurate. Framing with other windows to reduce the spectral leakage distorts the signal due to the asynchronous weighting around the frame specifically both ends of the frame. These problems degrade the performance of the CPSP-based TDOA estimation. In this paper, we propose a method to alleviate those problems by pre-emphasis of the speech signal. It reduces the influence of the spectral leakage by reducing dynamic range of the spectrum of a speech signal with pre-emphasis. To validate the proposed method of pre-emphasis, we carry out TDOA estimation experiments in various noise and reverberation conditions, Experimental results have shown that the framing of pre-emphasized microphone output by a rectangular window achieves higher success rate of TDOA estimation than any other framing methods.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.10a
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• pp.187-192
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• 2010

In this paper, intrusion detection technique based on the sound field variation of audio frequency in the security space is proposed. The sound field formed by sound source can be detected with the microphone when the obstacle or intruder is positioned. The sound field variation due to the intruder is based on the interference of audio wave. With the help of numerical simulation of sound field formations, the increase or decrease of sound pressure level is analyzed not only the obstacle, but also the intruder. Even the microphone is positioned behind the source, sound pressure level can be increase or decrease due to the interference. Frequency response test is performed with Gaussian white noise signal to get the whole frequency response from 0 to half of sampling frequency. There are three security cases. Case 1 is the situation of empty space with and without intruder, case 2 is the situation of blocking obstacle with and without intruder, and case 3 is the situation of side blocking obstacle with and without intruder. At each case, the frequency response is obtained first at the security space without intruder, and second with intruder. From the experiment, intruder size of $50cm{\times}50cm$ can be successfully detected with the proposed technique. Moreover, the case 2 or case 3 bring about bigger sound field variation. It means that the proposed technique have the potential of more credible security sensing in real situation.

• The Journal of the Acoustical Society of Korea
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• v.36 no.5
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• pp.345-352
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• 2017

We propose a methods which is reducing direction estimation error of sound source in the reverberant and noisy environments. The proposed algorithm divides speech signal into voice and unvoice using VAD. We estimate the direction of source when current frame is voiced. TDOA (Time-Difference of Arrival) between microphone array using the GCC-PHAT (Generalized Cross Correlation with Phase Transform) method will be estimated in that frame. Then, we compare the peak value of cross-correlation of two signals applied to estimated time-delay with other time-delay in time-table in order to improve the accuracy of source location. If the angle of current frame is far different from before and after frame in successive voiced frame, the angle of current frame is replaced with mean value of the estimated angle in before and after frames.