• Title/Summary/Keyword: Noise Estimation

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Prediction of Total Acoustic Radiation Power of the Submerged Circular Cylindrical Structures (수중 원통형 구조물의 총 음향방사파워 예측)

  • Han, Seungjin;Lee, Jongju;Kang, Myunghwan
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.24 no.11
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    • pp.876-882
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    • 2014
  • This study investigates an efficient method to estimate the total acoustic radiation power of submerged circular cylindrical structures. Since the acoustic radiation power of submerged vehicles can be changed during the operation, the estimation for its monitoring onboard is required to accomplish the missions. The total acoustic radiation power is estimated using the measured velocity and the calculated radiation efficiency of the surface which consists of submerged rectangular plate elements. Experiments are carried out to validate the estimation approach. Comparisons of the estimation results with the measurements show that they are in a good agreement for the mid-high frequency range and match well for the cases of different excitation locations which correspond to the different operation modes of underwater vehicles as well. Therefore, this estimation method can be applied effectively to the development of the radiated noise monitoring-system.

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.

Selective Extended Kalman Filter based Attitude Estimation (선택적 확장 칼만 필터 방식의 자세 추정)

  • Yun, In-Yong;Shim, Jae-Ryong;Kim, Joong-Kyu
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2016.10a
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    • pp.973-975
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    • 2016
  • In this paper, we propose a selective extended Kalman filter based accurate pose estimation of the rigid body using a sensor fusion method. The pose of a rigid body can be estimated roughly by the Gauss-Newton method using the acceleration data and geomagnetic data, which can be refined with vision information and the gyro sensor information. However strong external interference noise makes the rough pose estimation difficult. In this paper, according to the measurement level of the external interference noise, the extended Kalman filter selectively uses mostly vision and gyro sensor information to increase the estimation credibility under strong interference noise environment.

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Optimum Selection of Equalizer Taps Losing Noise Power Estimation (잡음 전력 추정을 이용한 등화기 탭의 최적 선택 방법)

  • 성원진;신동준
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.26 no.12A
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    • pp.1971-1977
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    • 2001
  • Multipath Rayleigh fading channels for mobile radio transmission can be represented by the linear filter model, and depending on the delay path characteristics, only a selected number of taps may have significance in the receiver structure design. By using tap-selective equalization, reduction in both processing complexity and power consumption can be obtained. In this paper, we present an optimal tap selection method for a given channel model, and demonstrate the performance improvement over an existing method. We show the method performs the CFAR (Constant False Alarm Rate) detection when the noise power information is available, and derive exact expressions of the error probability for the case of noise power estimation. Using the derived formulas and simulation results, it is demonstrated that the error probability quickly approaches to the optimal performance as the number samples used for the noise power estimation increases.

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Filter Size Determination Algorithms for Decision-Directed Channel Estimators in Wideband CDMA Mobile Communication Systems (광대역 CDMA이동통신 시스템의 결정지향 채널추정기를 위한 필터크기 결정 방법)

  • Rim, Min-Joong;Ryu, Chul;Ahn, Jae-Min
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.40 no.5
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    • pp.171-180
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    • 2003
  • CDMA(Code Division Multiple Access) mobile communication systems require accurate channel estimation in the receiver to compensate the fading distortions. Instantaneous channel estimates are obtained by dividing the received symbol by the transmitted symbol and then refined by filtering to reduce the estimation variance. In the channel estimation filter, the determination of the filter size is a very important task which greatly affects the estimation quality. While conventional methods usually use only velocity estimators to determine the channel estimation filter size, this paper proposes a filter size determination method for decision-directed channel estimators considering the symbol error rate and the signal-to-noise ratio in addition to the velocity of the mobile station. This paper shows that the symbol error rate and the signal-to-noise ratio are important factors for the determination of the channel estimation filter size.

Extraction of Unvoiced Consonant Regions from Fluent Korean Speech in Noisy Environments (잡음환경에서 우리말 연속음성의 무성자음 구간 추출 방법)

  • 박정임;하동경;신옥근
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.4
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    • pp.286-292
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    • 2003
  • Voice activity detection (VAD) is a process that separates the noise region from silence or noise region of input speech signal. Since unvoiced consonant signals have very similar characteristics to those of noise signals, it may result in serious distortion of unvoiced consonants, or in erroneous noise estimation to can out VAD without paying special attention on unvoiced consonants. In this paper, we propose a method to extract in an explicit way the boundaries between unvoiced consonant and noise in fluent speech so that more exact VAD could be performed. The proposed method is based on histogram in frequency domain which was successfully used by Hirsch for noise estimation, and a1so on similarity measure of frequency components between adjacent frames, To evaluate the performance of the proposed method, experiments on unvoiced consonant boundary extraction was performed on seven kinds of noisy speech signals of 10 ㏈ and 15 ㏈ SNR respectively.

An Improved Speech Absence Probability Estimation based on Environmental Noise Classification (환경잡음분류 기반의 향상된 음성부재확률 추정)

  • Son, Young-Ho;Park, Yun-Sik;An, Hong-Sub;Lee, Sang-Min
    • The Journal of the Acoustical Society of Korea
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    • v.30 no.7
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    • pp.383-389
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    • 2011
  • In this paper, we propose a improved speech absence probability estimation algorithm by applying environmental noise classification for speech enhancement. The previous speech absence probability required to seek a priori probability of speech absence was derived by applying microphone input signal and the noise signal based on the estimated value of a posteriori SNR threshold. In this paper, the proposed algorithm estimates the speech absence probability using noise classification algorithm which is based on Gaussian mixture model in order to apply the optimal parameter each noise types, unlike the conventional fixed threshold and smoothing parameter. Performance of the proposed enhancement algorithm is evaluated by ITU-T P.862 PESQ (perceptual evaluation of speech quality) and composite measure under various noise environments. It is verified that the proposed algorithm yields better results compared to the conventional speech absence probability estimation algorithm.

A Study on the SNR Estimation Performance of Hierarchical 16QAM (계층 16QAM의 SNR 추정 성능에 대한 연구)

  • Kwak, Jae-Min
    • Journal of Advanced Navigation Technology
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    • v.16 no.6
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    • pp.975-981
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    • 2012
  • The SNR estimation performance of hierarchical 16QAM system, which is adopted to simultaneous transmission or efficient image transmission system, is analyzed. Hierarchical 16QAM is modulation system which has different constellation shape from conventional QAM and can provide users with high quality and low quality of data services simultaneously by controlling hierarchical modulation parameter. Assuming AWGN channel, SNR estimation performance characteristics are investigated considering hierarchical modulation parameter and type of constellation points. From simulation results, it is found that constellation point showing superior SNR estimation performance relative to other points is exist. Also, it is known that according to hierarchical modulation parameter, SNR estimation range with more accurate estimation performance is divided.

EVM Based SNR Estimation Performance in Cross QAM Using Selected Constellation Points (Cross QAM의 선택적 성좌점을 사용하는 EVM 기반 SNR 추정 성능)

  • Kwak, Jae-Min
    • Journal of Advanced Navigation Technology
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    • v.16 no.3
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    • pp.426-432
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    • 2012
  • In this paper, we investigate the signal to noise ratio (SNR) estimation performance of Cross quadrature amplitude modulation (QAM), which is being used for asymmetric digital subscriber line (ADSL), very high bit rate digital subscriber line (VDSL), and digital video broadcasting - cable (DVB-C), and has been found to be useful in adaptive modulation and blind equalization. At first, the symbol error rate (SER) performance of Cross QAM is analyzed in Rayleigh fading channel. Then we suggest error vector magnitude (EVM) based SNR estimation utilizing the selected constellation points having different types of decision region from one another, and verify that SNR estimation performance of each points have different performance pattern through simulation. From the simulation results, it has been found that when suggested selected constellation points are used for SNR estimation in Cross QAM, estimation performance is enhanced in additive white Gaussian noise (AWGN) channel or Ricean fading channel.

Active noise control algorithm based on noise frequency estimation (소음 주파수 추정 기법을 이용한 능동소음제어 알고리즘)

  • 김선민;박영진
    • 제어로봇시스템학회:학술대회논문집
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    • 1997.10a
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    • pp.321-324
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    • 1997
  • In this paper, Active Noise Control(ANC) algorithm is proposed based on the estimated frequency estimator of the reference signal. The conventional feedforward ANC algorithms should measure the reference and use it to calculate the gradient of the squared error and filter coefficients. For ANC systems applied to aircrafts and passenger ships, engines from which reference signal is usually measured is so far from seats where main part of controller is placed that the scheme might be difficult to implement or very costly. Feedback ANC algorithm which doesn't need to measure the reference uses the error signal to update the filter and is sensitive to unexpected transient noise like a sneeze, clapping of hands and so on The proposed algorithm estimates frequencies of the desired signal in real time using adaptive notch filter. New frequency estimation algorithm is proposed with the improved convergence rate, threshold SNR and computational simplicity. Reference is not measured but created with the estimated frequencies. It has strong similarity to the conventional feedback control because reference is made from error signal. Enhanced error signal is used to update the controller for better performance under the measurement noise and impact noise. The proposed ANC algorithm is compared with the conventional feedback control.

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