• Title/Summary/Keyword: non-stationary noise

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Enhancement of Convergence Speed of Adaptive Algorithm using Wavelet Packet Transform (웨이브렛 패킷 변환을 이용한 적응알고리듬의 수렴속도 향상)

  • 박서용;김대성
    • The Journal of Information Technology
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    • v.2 no.2
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    • pp.127-138
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    • 1999
  • The wavelet transform is widely used in signal processing application. In this paper, a wavelet domain adaptive algorithm(WPTNLMS) is derived and its performances are evaluated in non-stationary environment. Where the input signals are decomposed by the wavelet packet transform for the multi-resolution adaptive processing. And the NLMS is used as an adaptive algorithm in wavelet domain. The proposed technique is applied to noise cancellation of the Doppler signal which is added with white Gaussian noise.

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Analysis of the Vibration of High Speed Trains and the Irregularity of Railway Using a Wavelet-based Frequency Response Function (웨이브렛 기반 주파수 응답함수를 이용한 고속철도차량의 진동 및 궤도불규칙 특성 분석)

  • Lee, Jun-Seok;Choi, Sung-Hoon;Kim, Sang-Soo;Park, Choon-Soo
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2008.04a
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    • pp.766-771
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    • 2008
  • In this paper, the vibration of high speed trains and the irregularity of railway are examined using a wavelet-based frequency response function. To investigate their characteristics, non-stationary acceleration data are acquired and processed using the wavelet transform. Also, the railway irregularity is examined by acquiring the data from the on-board laser-based measurement system. The correlation between the train vibration and the railway irregularity has been investigated. From the analysis, the wavelet-based frequency response function is a promised method for the dynamic characteristics of high speed trains.

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Classification of Underwater Transient Signals Using Gaussian Mixture Model (정규혼합모델을 이용한 수중 천이신호 식별)

  • Oh, Sang-Hwan;Bae, Keun-Sung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.16 no.9
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    • pp.1870-1877
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    • 2012
  • Transient signals generally have short duration and variable length with time-varying and non-stationary characteristics. Thus frame-based pattern matching method is useful for classification of transient signals. In this paper, we propose a new method for classification of underwater transient signals using a Gaussian mixture model(GMM). We carried out classification experiments for various underwater transient signals depending upon the types of noise, signal-to-noise ratio, and number of mixtures in the GMM. Experimental results have verified that the proposed method works quite well for classification of underwater transient signals.

An Analysis of the Wave Propagation of a Structure Based on STFT, Higher Order Time-frequency Analysis and Wavelet Transform (STFT, 고차위그너분포 및 웨이브렛 변환 기술을 이용한 탄성파 추적)

  • 이상권
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2003.05a
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    • pp.827-832
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    • 2003
  • There has been a number methods for the presentation of time-frequency analysis of non-stationary signal. In this paper, STFT(short time Fourier transform), wavelet transform, Wigner distribution, and higher order Wigner distribution are discussed in details with simulation signals. They are also applied to the analysis of the wave propagation of a semi finite beam. Wigner distribution and higher order Wigner distribution have good time-frewuency resolutions. Wavelet transform is required for impact analysis but should be applied carefully. STFT suffers from time-frequency resolutions. Each method is has its advantage and disadvantage depending on each application signals.

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Noise Reduction Using MMSE Estimator-based Adaptive Comb Filtering (MMSE Estimator 기반의 적응 콤 필터링을 이용한 잡음 제거)

  • Park, Jeong-Sik;Oh, Yung-Hwan
    • MALSORI
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    • no.60
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    • pp.181-190
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    • 2006
  • This paper describes a speech enhancement scheme that leads to significant improvements in recognition performance when used in the ASR front-end. The proposed approach is based on adaptive comb filtering and an MMSE-related parameter estimator. While adaptive comb filtering reduces noise components remarkably, it is rarely effective in reducing non-stationary noises. Furthermore, due to the uniformly distributed frequency response of the comb-filter, it can cause serious distortion to clean speech signals. This paper proposes an improved comb-filter that adjusts its spectral magnitude to the original speech, based on the speech absence probability and the gain modification function. In addition, we introduce the modified comb filtering-based speech enhancement scheme for ASR in mobile environments. Evaluation experiments carried out using the Aurora 2 database demonstrate that the proposed method outperforms conventional adaptive comb filtering techniques in both clean and noisy environments.

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Crank Angle Analysis

  • Gade, Svend;Hald, Jorgen
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2001.11b
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    • pp.1040-1043
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    • 2001
  • This paper describes the principle behind Crank Angle Analysis, as implemented by Bruel & Kjaer in the Non-Stationary Spatial Transformation of Sound Fields (NS-STSF) system. The NS-STSF system combines a Time Domain Holography measurement on for example an engine with two simultaneously recorded Tacho signals. The Tacho signals provide the crankshaft angle and the RPM at the instant of each instantaneous output (snap-shot) from Time Domain Holography. As a result, the system allows precise analysis of the temporal and spatial relation between the acoustical emission (or the vibration pattern) and the mechanical events during an engine cycle. Some results from a measurement on a DaimlerChrysler engine are presented.

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A Novel Speech Enhancement Based on Speech/Noise-dominant Decision in Time-frequency Domain (시간-주파수 영역에서 음성/잡음 우세 결정에 의한 새로운 잡음처리)

  • 윤석현;유창동
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.3
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    • pp.48-55
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    • 2001
  • A novel method to reduce additive non-stationary noise is proposed. The method requires neither the information about noise nor the estimate of the noise statistics from any pause regions. The enhancement is performed on a band-by-band basis for each time frame. Based on both the decision on whether a particular band in a frame is speech or noise dominant and the masking property of the human auditory system, an appropriate amount of noise is reduced using spectral subtraction. The proposed method was tested on various noisy conditions (car noise, Fl6 noise, white Gaussian noise, pink noise, tank noise and babble noise) and on the basis of comparing segmental SNR with spectral subtraction method and visually inspecting the enhanced spectrograms and listening to the enhanced speech, the method was able to effectively reduce various noise while minimizing distortion to speech.

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End Edge Cogging Force Minimization according to the Distance between Armatures of Stationary Discontinuous Armature PMLSM with Concentrated Winding (전기자 분산배치 집중권 PMLSM의 전기자 간격에 따른 단부 코깅력 최소화)

  • Kim, Yong-Jae
    • The Journal of the Korea institute of electronic communication sciences
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    • v.8 no.8
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    • pp.1241-1246
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    • 2013
  • Currently, The general transportation system arranges the armature on the full length of transportation lines. However, when this method is applied to the long distance transportation system, it causes an increase of material cost and manufacturing time. Thus, in order to resolve this problem, discontinuous arrangement method of the armature has been proposed. However, in the method of using stationary discontinuous armatures, mover can stop in the freewheeling section which is non-installations section when disturbance is generated and the mover can not be moved because armature control is impossible. Thus, the distance determination of armature is very important. Also, when the armature is arranged discontinuously the edge always exists due to the structure. Due to this edge, the cogging force is greatly generated during the entry and ejection of the mover to the armature. This cogging force causes thrust force ripple generating noise, vibration and decline of performance, it must be reduced. Therefore, in this paper, we examined the end edge cogging force generated by the stationary discontinuous armatures through 2-D numerical analysis using finite element method (FEM) and we figured out distance of armature for end edge cogging force minimization.

Design of the Noise Suppressor Using Wavelet Transform (웨이블릿 변환을 이용한 잡음제거기 설계)

  • 원호진;김종학;이인성
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.7
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    • pp.37-46
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    • 2001
  • This paper proposes a new noise suppression method using the Wavelet transform analysis. The noise suppressor using the Wavelet transform shows the more effective advantages in a babble noise than one using the short-time Fourier transform. We designed a new channel structure based on spectral subtraction of Wavelet transform coefficients and used the Wavelet mask pattern with more higher time resolution in high frequency. It showed a good adaptation capability for babble noise with a non-stationary property. To evaluate the performance of proposed noise canceller, the informal subjective listening tests (Mos tests) were performed in background noise environments (car noise, street noise, babble noise) of mobile communication. The proposed noise suppression algorithm showed about MOS 0.2 performance improvements than the suppression algorithm of EVRC in informal listening tests. The noise reduction by the proposed method was shown in spectrogram of speech signal.

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Speech Enhancement System Using a Model of Auditory Mechanism (청각기강의 모델을 이용한 음성강조 시스템)

  • 최재승
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.41 no.6
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    • pp.295-302
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    • 2004
  • On the field of speech processing the treatment of noise is still important problems for speech research. Especially, it has been noticed that the background noise causes remarkable reduction of speech recognition ratio. As the examples of the background noise, there are such various non-stationary noises existing in the real environment as driving noise of automobiles on the road or typing noise of printer. The treatment for these kinds of noises is not so simple as could be eliminated by the former Wiener filter, but needs more skillful techniques. In this paper as one of these trials, we show an algorithm which is a speech enhancement method using a model of mutual inhibition for noise reduction in speech which is contaminated by white noise or background noise mentioned above. It is confirmed that the proposed algorithm is effective for the speech degraded not only by white noise but also by colored noise, judging from the spectral distortion measurement.