• Title/Summary/Keyword: GCC (Generalized Cross Correlation)

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Real-Time Sound Localization System For Reverberant And Noisy Environment (반향음과 잡음 환경을 고려한 실시간 소리 추적 시스템)

  • Kee, Chang-Don;Kim, Ghang-Ho;Lee, Taik-Jin
    • Journal of the Korean Society for Aeronautical & Space Sciences
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    • v.38 no.3
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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.

An efficient space dividing method for the two-dimensional sound source localization (2차원 상의 음원위치 추정을 위한 효율적인 영역분할방법)

  • Kim, Hwan-Yong;Choi, Hong-Sub
    • The Journal of the Acoustical Society of Korea
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    • v.35 no.5
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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 %.

Spatially Mapped GCC Function Analysis for Multiple Source and Source Localization Method (공간좌표로 사상된 GCC 함수의 다 음원에 대한 해석과 음원 위치 추정 방법)

  • Kwon, Byoung-Ho;Park, Young-Jin;Park, Youn-Sik
    • Journal of Institute of Control, Robotics and Systems
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    • v.16 no.5
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    • pp.415-419
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    • 2010
  • A variety of methods for sound source localization have been developed and applied to several applications such as noise detection system, surveillance system, teleconference system, robot auditory system and so on. In the previous work, we proposed the sound source localization using the spatially mapped GCC functions based on TDOA for robot auditory system. Performance of the proposed one for the noise effect and estimation resolution was verified with the real environmental experiment under the single source assumption. However, since multi-talker case is general in human-robot interaction, multiple source localization approaches are necessary. In this paper, the proposed localization method under the single source assumption is modified to be suitable for multiple source localization. When there are two sources which are correlated, the spatially mapped GCC function for localization has three peaks at the real source locations and imaginary source location. However if two sources are uncorrelated, that has only two peaks at the real source positions. Using these characteristics, we modify the proposed localization method for the multiple source cases. Experiments with human speeches in the real environment are carried out to evaluate the performance of the proposed method for multiple source localization. In the experiments, mean value of estimation error is about $1.4^{\circ}$ and percentage of multiple source localization is about 62% on average.

Time delay estimation algorithm using Elastic Net (Elastic Net를 이용한 시간 지연 추정 알고리즘)

  • Jun-Seok Lim;Keunwa Lee
    • The Journal of the Acoustical Society of Korea
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    • v.42 no.4
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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.

Time Delay Estimation Using Automatic Tracking Window (자동추적윈도우를 이용한 시간지연 추정)

  • 윤병우;신윤기;박의열
    • Journal of the Korean Institute of Telematics and Electronics A
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    • v.28A no.5
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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.

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Time delay estimation between two receivers using weighted dictionary method for active sonar (능동소나를 위한 가중 딕션너리를 사용한 두 수신기 간 신호 지연 추정 방법)

  • Lim, Jun-Seok;Kim, Seongil
    • The Journal of the Acoustical Society of Korea
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    • v.40 no.5
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    • pp.460-465
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    • 2021
  • In active sonar, time delay estimation is used to find the distance between the target and the sonar. Among the time delay estimation methods for active sonar, estimation in the frequency domain is widely used. When estimating in the frequency domain, the time delay can be thought of as a frequency estimator, so it can be used relatively easily. However, this method is prone to rapid increase in error due to noise. In this paper, we propose a new method which applies weighted dictionary and sparsity in order to reduce this error increase and we extend it to two receivers to propose an algorithm for estimating the time delay between two receivers. And the case of applying the proposed method and the case of not applying the proposed method including the conventional frequency domain algorithm and Generalized Cross Correlation-Phase transform (GCC-PHAT) in a white noise environment were compared with one another. And we show that the newly proposed method has a performance gain of about 15 dB to about 60 dB compared to other algorithms.

Comparison of the sound source localization methods appropriate for a compact microphone array (소형 마이크로폰 배열에 적용 가능한 음원 위치 추정법 비교)

  • Jung, In-Jee;Ih, Jeong-Guon
    • The Journal of the Acoustical Society of Korea
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    • v.39 no.1
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    • pp.47-56
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    • 2020
  • The sound source localization technique has various application fields in the era of internet-of-things, for which the probe size becomes critical. The localization methods using the acoustic intensity vector has an advantage of downsizing the layout of the array owing to a small finite-difference error for the short distance between adjacent microphones. In this paper, the acoustic intensity vector and the Time Difference of Arrival (TDoA) method are compared in the viewpoint of the localization error in the far-field. The comparison is made according to the change of spacing between adjacent microphones of the three-dimensional microphone array arranged in a tetrahedral shape. An additional test is conducted in the reverberant field by varying the reverberation time to verify the effectiveness of the methods applied to the actual environments. For estimating the TDoA, the Generalized Cross Correlation-Phase transform (GCC-PHAT) algorithm is adopted in the computation. It is found that the mean localization error of the acoustic intensimetry is 2.9° and that of the GCC-PHAT is 7.3° for T60 = 0.4 s, while the error increases as 9.9°, 13.0° for T60 = 1.0 s, respectively. The data supports that a compact array employing the acoustic intensimetry can localize of the sound source in the actual environment with the moderate reflection conditions.

Time delay estimation by iterative Wiener filter based recursive total least squares algorithm (반복형 위너 필터 방법에 기반한 재귀적 완전 최소 제곱 방법을 사용한 시간 지연 추정 알고리즘)

  • Lim, Jun-Seok
    • The Journal of the Acoustical Society of Korea
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    • v.40 no.5
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    • pp.452-459
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    • 2021
  • Estimating the mutual time delay between two acoustic sensors is used in various fields such as tracking and estimating the location of a target in room acoustics and sonar. In the time delay estimation methods, there are a non-parametric method, such as Generalized Cross Correlation (GCC), and a parametric method based on system identification. In this paper, we propose a time delay estimation method based on the parametric method. In particular, we propose a method that considers the noise in each receiving acoustic sensor. Simulation confirms that the proposed algorithm is superior to the existing generalized cross-correlation and adaptive eigenvalue analysis methods in white noise and reverberation environments.

Direction-of-Arrival Estimation of Speech Signals Based on MUSIC and Reverberation Component Reduction (MUSIC 및 반향 성분 제거 기법을 이용한 음성신호의 입사각 추정)

  • Chang, Hyungwook;Jeong, Sangbae;Kim, Youngil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.18 no.6
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    • pp.1302-1309
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    • 2014
  • In this paper, we propose a method to improve the performance of the direction-of-arrival (DOA) estimation of a speech source using a multiple signal classification (MUSIC)-based algorithm. Basically, the proposed algorithm utilizes a complex coefficient band pass filter to generate the narrow band signals for signal analysis. Also, reverberation component reduction and quadratic function-based response approximation in MUSIC spatial spectrum are utilized to improve the accuracy of DOA estimation. Experimental results show that the proposed method outperforms the well-known generalized cross-correlation (GCC)-based DOA estimation algorithm in the aspect of the estimation error and success rate, respectively.Abstract should be placed here. These instructions give you guidelines for preparing papers for JICCE.

Improved time delay estimation by adaptive eigenvector decomposition for two noisy acoustic sensors (잡음이 있는 두 음향 센서를 이용한 시간 지연 추정을 위한 향상된 적응 고유벡터 추정 기반 알고리즘)

  • Lim, Jun-Seok
    • The Journal of the Acoustical Society of Korea
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    • v.37 no.6
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    • pp.499-505
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    • 2018
  • Time delay estimation between two acoustic sensors is widely used in room acoustics and sonar for target position estimation, tracking and synchronization. A cross-correlation based method is representative for the time delay estimation. However, this method does not have enough consideration for the noise added to the receiving acoustic sensors. This paper proposes a new time delay estimation method considering the added noise on the receiver acoustic sensors. From comparing with the existing GCC (Generalized Cross Correlation) method, and adaptive eigen decomposition method, we show that the proposed method outperforms other methods for a colored signal source in the white Gaussian noise condition.