• Journal of Korea Multimedia Society
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• v.21 no.8
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• pp.925-933
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• 2018

In this paper, we propose a new smart ordering application system for granting the customer numbers. The proposed system assigns sequence number of customers who visit at offline store as using speaker and microphone of smart device. This system has more advantage than the existing ordering system. Because the proposed system does not need any customer information like as phone number or SNS ID, it can protect customer privacy information. In this system, we use high frequency which is inaudible to the human ear as communication signal between speaker and microphone. To confirm performance evaluation, we perform a test with ten smart devices like as iPhone 6, 7, 8, Galaxy s8 and the result shown a success rate of 98.7 percent. Therefore, the proposed system can be useful service technology at the various offline store which need to assign a sequence number for customers, because many customers visit at the store.

• Proceedings of the KIEE Conference
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• 1991.11a
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• pp.378-381
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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. The signal processor DSP56001 is used to implement the real-time controller and the experimental results shows that our system can reduce the noise level in duct to $20{\sim}40$ [db].

• Journal of Information Technology Applications and Management
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• v.24 no.1
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• pp.157-167
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• 2017

This paper proposes a simple sound source localization (SSL) method based on signal energies comparison and partial cross correlation for TDOA computation. Many sound source localization methods include multiple TDOA computations in order to eliminate front-back confusion. Multiple TDOA computations however increase the methods' computation times which need to be as minimal as possible for real-time applications. Our aim in this paper is to achieve the same results of localization using fewer computations. Using three microphones, we first compare signal energies to predict which quadrant the sound source is in, and then we use partial cross correlation to estimate the TDOA value before computing the azimuth value. Also, we apply a threshold value to reinforce our prediction method. Our experimental results show that the proposed method has less computation time; spending approximately 30% less time than previous three microphone methods.

• Journal of the Korean Society for Precision Engineering
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• v.21 no.5
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• pp.91-98
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• 2004

Human uses level difference and time difference to get space information. Therefore this paper shows that method to presume direction of sound source by time difference and to mark presumed position. The position means direction from geometrical center of sensors to the sound source. To get the time difference of microphones input level, we will be explained about arrangement of microphones which used for the sensor to take the sound signal. It is included distance among the 3 microphones and distance between microphones and sound source. Secondly, input signals are transmitted to CPU througth digital process. CPU is used to DSP(Digital Signal Processor) for manage the signal by real time. Finally, the position of sound source is perceived by an explained algorithm in this paper.

• Geophysics and Geophysical Exploration
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• v.2 no.4
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• pp.180-183
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• 1999

A proto-type seismic measurement system with a reference was developed to improve S/N (signal-to-noise ratio) of seismic data, especially in noisy urban areas. Two pairs of correlation measurements (the one for microphone and geophone, and another for electromagnetic (EM) loop and geophone) were carried out near Kimpo Airport and at Kimje. The spectrum analyses were also performed to investigate the correlation of two pairs of time series; one for microphone and geophone, and another for EM loop and geophone. The sound waves measured with the microphone and the geophone are highly correlated. However, differences in the reponses are readily identifiable across 200 Hz; in the vicinity of 100 Hz, the spectral energy for geophone is 20 dB higher than that for microphone, and at near 500 Hz, the spectral energy for microphone is 30 dB higher than that for geophone. Overall, the spectral energy appears concentrated on the frequency window below 600 Hz for geophone. It contrasts with the observation of dominant frequency at the range of above 200 Hz for microphone. The wave forms of EM noise (due to an ACDC inverter) measured with EM loop and geophone are consistently and highly correlated each other. The power spectrum of the EM noise for EM loop shows that the spectral energies at odd harmonic frequencies of 60 Hz are higher than those at even harmonic frequencies of 60 Hz. It is compared to the power spectrum for geophone; the spectral energies at odd harmonics are nearly same as those at even harmonic frequencies.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.4 s.304
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• pp.111-118
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• 2005

In this paper, we employ the modulation technique for improving the characteristics of beamformer in the low frequencies and thus improving the overall noise reduction performance. In the 1-dimensional uniform linear microphone arrays, we can suppress the narrowband noise component using the delay-and-sum beamforming. But, for the wideband noise signal, the delay-and-sum beamformer does not work well for the reduction of low frequency component because the inter-element spacing is usually set to avoid spatial aliasing at high frequencies. Hence, the beamwidth is not uniform with respect to each frequency and it is usually wider at the low frequencies. In order to obtain the beamwidth independent of frequencies, subarray systems[1][2][3][4] and multi-beamforming[5] have been proposed. However these algorithms need large space and more microphones since they are based on the theory that the size of the array is proportional to the wavelength of the input signal. In the proposed beamformer, we reduce the low frequency noise by using modulation technique that does not need additional sensors or non-uniform spacing. More Precisely, the array signals are split into subbands, and the low frequency components are shifted to high frequencies by modulation and reduced by the delay-and-sum beamforming techniques with small size microphone array. Experimental results show that the proposed technique Provides better performance than the conventional ones, especially in the low frequency band.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.35-38
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• 2004

In this paper, 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 makes estimation of CPSP spectrum inaccurate. Framing with a Hamming window to reduce the spectral leakage effect distorts the signal due to the different weighting at temporally same sample, which make the TDOA estimation using CPSP function inaccurate. In this paper, we solve this problem by reducing the dynamic range of the spectrum of a speech signal with preemphasis. Experimental results confirm that the framing of pre-emphasized microphone output with a rectangular window shows higher success ratio of TDOA estimation than any other framing methods.

• Journal of the korean Society of Automotive Engineers
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• v.14 no.5
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• pp.117-127
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• 1992

In this work, the possibility to diagnose valve lashes of an automotive diesel engine via cylinder head vibration/noise analysis is studied. First of all the measurement signals and conditions are selected after considering which signals and conditions are most suitable to diagonse valve lashes. Both accelerometer and microphone are used to measure cylinder head accelerations and acoustic pressure due to valve impact on cylinder head. The signals are measured in both cranking and engine firing conditions. Finally, it was found that acceleration signal obtained in engine operating condition is the most reliable signal to diagnose the valve lash condition. The valve closing angle and the peak acceleration due to valve close are chosen to analyze the valve lash condition. The measured cylinder head acceleration signals are statistically tested to derive information which are useful to judge the valve lash. In conclusion, it was found that the developed technique can be one of feasible methods to diagnose the valve conditions while the engine is in operation.

• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.416-418
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• 2006

In this paper, a new acoustic echo cancellation approach based on the DUET algorithm and scaling factor estimation is proposed to solve the scaling ambiguity in case of blind separation based acoustic echo cancellation in a noisy environment. In hands-free full-duplex communication system. acoustic noises picked up by the microphone are mixed with echo signal. For this reason, the echo cancellation system may provide poor performance. For that purpose, a degenerate unmixing estimation technique, adjusted in the time-frequency domain, is employed to separate undesired echo signals and noises. Also, since scaling and permutation ambiguities have not been solved in the blind source separation algorithm, kurtosis for the desired signal selection and a scaling factor estimation algorithm are utilized in this rarer for the separation of an echo signal. Simulation results demonstrate that the proposed approach yields better echo cancellation and noise reduction performances, compared with conventional methods.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.