• Transactions of the Korean Society for Noise and Vibration Engineering
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• 제20권6호
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• pp.568-576
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• 2010

It is widely known that the leak locating of underground plastic pipelines is much more difficult than that of cast iron pipelines. The precision of the leak locating depends upon the speed of leak signal and the time delay estimation between the two sensors on the pipeline. In this paper, six different windowing filters are considered to improve the time delay estimation especially for the plastic pipelines. The time delay is usually estimated from the peak time of cross-correlation functions. The filtering windows including rectangle, Roth, Wiener, SCOT, PHAT and maximum likelihood are applied to derive the generalized cross-correlation function and compared each other. Experimental results for the actual plastic underground water supply pipeline show that the introduction of the filtering windows improved the precision of time delay estimation. Some window functions provide excellent leak locating capability for the plastic pipe of 98 m long, which is less than 1 % of the pipe lengths. Also a new probabilistic approach that the combinations of all results from each filtering window is suggested for the better leak locating.

• Phonetics and Speech Sciences
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• 제1권3호
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• pp.95-102
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• 2009

In this paper, an efficient target speech detection algorithm is proposed for the performance improvement of multi-input speech enhancement. Using the normalized cross correlation value between two selected channels, the proposed algorithm estimates the probabilistic distribution function of the value from the pure noise interval. Then, log-likelihoods are calculated with the function and the normalized cross correlation value to detect the target speech interval precisely. The detection results are applied to the generalized sidelobe canceller-based algorithm. Experimental results show that the proposed algorithm significantly improves the speech recognition performance and the signal-to-noise ratios.

• The Journal of the Acoustical Society of Korea
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• 제30권3호
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• pp.115-122
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• 2011

In this paper, we present a method to estimate an accurate real-time sound source direction based on time delay of arrival by using generalized cross correlation with four cross-type microphones. In general, existing systems have two disadvantages such as system embedding limitation due to the necessity of data acquisition for signal processing from microphone input, and real-time processing difficulty because of the increased number of channels for sound direction estimation using DSP processors. To cope with these disadvantages, the system considered in this paper proposes hardware design for enhanced real-time processing using microphone array signal processing. An accurate direction estimation and its design time reduction is achieved by means of an efficient hardware design using spatial segmentation methods and verification techniques. Finally we develop a system which can be used for embedded systems using a sound codec and an FPGA chip. According to experimental results, the system gives much faster real-time processing time compared with either PC-based systems or the case with DSP processors.

• IEMEK Journal of Embedded Systems and Applications
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• 제12권5호
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• pp.267-275
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• 2017

Time of difference of arrival (TDOA) method using passive sonar sensor array has normally been used to estimate the location of a concealed moving target in underwater environment. Particle filter has been introduced for effective target estimation for non-Gaussian and nonlinear systems. In this paper, we propose a GPU-based acceleration of target position estimation using particle filter and propose efficient embedded system and software architecture. For the TDOA measurement from the passive sonar sensor, we use the generalized cross correlation phase transform (GCC-PHAT) method to obtain the correlation coefficient of the signal using FFT and we try to accelerate the calculation of GCC-PHAT based TDOA measurements using FFT with GPU CUDA. We also propose parallelization method of the target position estimation algorithm using the GPU CUDA to update the state of each particle for the target position estimation using the measured values. The target estimation algorithm was verified using Matlab and implemented using GPU CUDA. Then, we realized the proposed signal processing acceleration system using NVIDIA Jetson TX1 as the target board to analyze in terms of the execution time. The execution time of the algorithm is reduced by 55% to the CPU standalone-operation on the target board. Experiment results show that the proposed architecture is a feasible solution in terms of high-performance and area-efficient architecture.

• The Journal of the Acoustical Society of Korea
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• 제42권4호
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• pp.364-369
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• 2023

Time-delay estimation between two receivers is a technique that has been applied in a variety of fields, from underwater acoustics to room acoustics and robotics. There are two types of time delay estimation techniques: one that estimates the amount of time delay from the correlation between receivers, and the other that parametrically models the time delay between receivers and estimates the parameters by system recognition. The latter has the characteristic that only a small fraction of the system's parameters are directly related to the delay. This characteristic can be exploited to improve the accuracy of the estimation by methods such as Lasso regularization. However, in the case of Lasso regularization, the necessary information is lost. In this paper, we propose a method using Elastic Net that adds Ridge regularization to Lasso regularization to compensate for this. Comparing the proposed method with the conventional Generalized Cross Correlation (GCC) method and the method using Lasso regularization, we show that the estimation variance is very small even for white Gaussian signal sources and colored signal sources.

• The Journal of the Acoustical Society of Korea
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• 제32권6호
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• pp.548-555
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• 2013

The localization of sources has a numerous number of applications. To estimate the position of sources, the relative delay between two or more received signals for the direct signal must be determined. Although the generalized cross-correlation method is the most popular technique, an approach based on eigenvalue decomposition (EVD) is also popular one, which utilizes an eigenvector of the minimum eigenvalue. The performance of the eigenvalue decomposition (EVD) based method degrades in the low SNR and the correlated environments, because it is difficult to select a single eigenvector for the minimum eigenvalue. In this paper, we propose a new adaptive algorithm based on Canonical Correlation Analysis (CCA) in order to extend the operation range to the lower SNR and the correlation environments. The proposed algorithm uses the eigenvector corresponding to the maximum eigenvalue in the generalized eigenvalue decomposition (GEVD). The estimated eigenvector contains all the information that we need for time delay estimation. We have performed simulations with uncorrelated and correlated noise for several SNRs, showing that the CCA based algorithm can estimate the time delays more accurately than the adaptive EVD algorithm.

• Journal of the Korean Institute of Telematics and Electronics A
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• 제28A권5호
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• pp.347-354
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• 1991

In this paper, the Automatic Tracking Window(ATW) algorithm is applied to the Generalized Cross-Correlation(GCC) time delay estimation algorithm as a preprocessing. The Linear Prediction(LP) algorithm, which is a pararmetric spectral estimation algorithm, is applied to the time delay estimation. And the ATW, a preprocessing algorithm is applied to this algorithm too. This paper shows that the ATW algorithm attenuates the sidelobes very much and improves the resolution of the timedelay estimation.

• The Journal of the Korea institute of electronic communication sciences
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• 제7권2호
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• pp.263-269
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• 2012

Code-Division Multiple-Access(CDMA) allows several users simultaneous access to a common channel by assigning a distinct pseudonoise sequence called spectrum code to each user. Each user in a CDMA system uses a assigned spectrum code to modulate their signal. Choosing the codes used to modulate the signal is very important in the performance of CDMA systems. The best performance will occur when there is good separation between the signal of a desired user and the signals of other users. The receiver synchronizes the code to recover the data. The use of an independent code allows multiple users to access the same frequency band at the same time. In this paper we propose a generalized model of non-linear binary sequence using trace function and analyze cross-correlation of these sequences. These sequences with low correlation, large linear span and large family size, in a direct-sequence spread spectrum communication system, help to minimize multiple access interference, increase security degree of system and enlarge user number.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• 제38권3호
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• pp.258-263
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• 2010

Sound localization algorithm usually adapts three step process: sampling sound signals, estimating time difference of arrival between microphones, estimate location of sound source. To apply this process in indoor environment, sound localization algorithm must be strong enough against reverberant and noisy condition. Additionally, calculation efficiency must be considered in implementing real-time sound localization system. To implement real-time robust sound localization system we adapt four low cost condenser microphones which reduce the cost and total calculation load. And to get TDOA(Time Differences of Arrival) of microphones we adapt GCC-PHAT(Generalized Cross Correlation-Phase Transform) which is robust algorithm to the reverberant and noise environment. The position of sound source was calculated by using iterative least square algorithm which produce highly accurate position data.

• The Journal of the Acoustical Society of Korea
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• 제35권5호
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• pp.358-367
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• 2016

SSL (Sound Source Localization) has been applied to several applications such as man-machine interface, video conference system, smart car and so on. But in the process of sound source localization, angle estimation error is occurred mainly due to the non-linear characteristics of the sine inverse function. So an approach was proposed to decrease the effect of this non-linear characteristics, which divides the microphone's covering space into narrow regions. In this paper, we proposed an optimal space dividing way according to the pattern of microphone array. In addition, sound source's 2-dimensional position is estimated in order to evaluate the performance of this dividing method. In the experiment, GCC-PHAT (Generalized Cross Correlation PHAse Transform) method that is known to be robust with noisy environments is adopted and triangular pattern of 3 microphones and rectangular pattern of 4 microphones are tested with 100 speech data respectively. The experimental results show that triangular pattern can't estimate the correct position due to the lower space area resolution, but performance of rectangular pattern is dramatically improved with correct estimation rate of 67 %.