• Title/Summary/Keyword: 신호잡음

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Subband Based Spectrum Subtraction Algorithm (서브밴드에 기반한 스펙트럼 차감 알고리즘)

  • Choi, Jae-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.8 no.4
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    • pp.555-560
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    • 2013
  • This paper first proposes a classification algorithm which detects a voiced, unvoiced, and silence signal using distance measure, logarithm power and root mean square methods at each frame, then a spectrum subtraction algorithm based on a subband filter. The proposed algorithm subtracts spectrums of white noise and street noise from noisy signal based on the subband filter at each frame. In this experiment, experimental results of the proposed spectrum subtraction algorithm demonstrate using the speech and noise data of Aurora-2 database. Based on measuring the speech-to-noise ratio (SNR), experiments confirm that the proposed algorithm is effective for the speech by contaminated the noise. From the experiments, the improvement in the output SNR values was approximately 2.1 dB and 1.91 dB better for white noise and street noise, respectively.

Denoising Algorithm using Wavelet and Element Deviation-based Median Filter (웨이브렛과 원소 편차 기반의 중간값 필터를 이용한 잡음제거 알고리즘)

  • Bae, Sang-Bum;Kim, Nam-Ho
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.14 no.12
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    • pp.2798-2804
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    • 2010
  • The audio and image signal are corrupted by various noises in signal processing, many studies are being accomplished to restore those signals. In this paper, the algorithm is proposed to remove additive Gaussian noise and impulse noise at one dimension signal like an speech signal. The algorithm is composed to remove Gaussian noise after removing impulse noise. And the method using wavelet coefficient accumulation is used to remove the Gaussian noise, and the median filter based on element deviation is applied to remove the impulse noise. Also we compare existing methods using SNR(signal-to-noise ratio) as the standard of judgement of improvemental effect.

A Study on Translation-Invariant Wavelet De-Noising with Multi-Thresholding Function (다중 임계치 함수의 TI 웨이브렛 잡음제거 기법)

  • Choi, Jae-Yong
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.7
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    • pp.333-338
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    • 2006
  • This paper proposes an improved do-noising method using multi-thresholding function based on translation-invariant (W) wavelet proposed by Donoho et al. for underwater radiated noise measurement. The traditional wavelet thresholding de-noising method causes Pseudo-Gibbs phenomena near singularities due to discrete wavelet transform. In order to suppress Pseudo-Gibbs Phenomena, a do-noising method combining multi-thresholding function with the translation-invariant wavelet transform is proposed in this paper. The multi-thresholding function is a modified soft-thresholding to each node according to the discriminated threshold so as to reject かon external noise and white gaussian noise. It is verified by numerical simulation. And the experimental results are confirmed through sea-trial using multi-single sensors.

Tonal Signal Detection for Acoustic Targets using ASM Neural Network (ASM 신경망을 이용한 음향 표적의 토날 신호 탐지)

  • 이성은
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1996.06a
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    • pp.22-28
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    • 1996
  • 수동 소나 시스템에서 표적을 탐지, 식별하는데 가장 중요한 인자는 표적에서 발생되는 토날 신호 성분이다. 수중의 주변잡음과 표적소음이 복합된 환경하에서 표적의 토날 신호성분을 정확히 추출하는데는 신호 탐지 준위 설정이나 주변 잡음의 변화에 의해 어려움이 있다. 본 논문에서는 ASM 신경망을 이용하여 신호 탐지 준위 설정이나 주변잡음의 변화에 강인한 음향 표적의 토날 신호 탐지 방식을 제안한다. 모의 시뮬레이션 및 실제 표적 신호에 적용하여 우수한 토날 신호 탐지 성능을 보인다.

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A Comparison of Representative Beat Extraction Algorithms in ECG (심전도 신호에서의 대표 비트 설정에 관한 알고리즘 비교)

  • 김동석;전대근;윤형로
    • Journal of Biomedical Engineering Research
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    • v.20 no.3
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    • pp.299-305
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    • 1999
  • In thls paper, the representative beal textraction algorIthms for the diagnostic parameter extraction in noisy signal were compared. We used the avernge, median, mode, and trmmed mean to calculale the central tendency. In our experimenl, we have restricted to four kinds of noises -EMG noise, 60Hz powerline inlerference, ahrupl baseline shift, and baselme drift due to respimtion-which were commonly occurred in ECG mgnal, then we have calculated signal-to-noise ratios(SNRs) for the ECG corrupted with each noise and all noises together. As the result of this paper, we have proved that the average method has super lor performance than the others in the ECG corrupted wilh EMG noise. When the signal mcludes extreme value such as abrupt baseline shIft, the median, mode, trimmed mean methods have supenor performance in the SNR ratios. Especially when the ECG corrupted with baseline drift due to respirallon, the trimmed mean method was most efficient because ST level change was 0 V.

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Noise Statistics Estimation Using Target-to-Noise Contribution Ratio for Parameterized Multichannel Wiener Filter (변수내장형 다채널 위너필터를 위한 목적신호대잡음 기여비를 이용한 잡음추정기법)

  • Hong, Jungpyo
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.26 no.12
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    • pp.1926-1933
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    • 2022
  • Parameterized multichannel Wiener filter (PMWF) is a linear filter that can control the trade-off between residual noise and signal distortion using the embedded parameter. To apply the PMWF to noisy inputs, accurate noise estimation is important and multichannel minima-controlled recursive averaging (MMCRA) is widely used. However, in the case of the MMCRA, the accuracy of noise estimation decreases when a directional interference is involved into the array inputs. Consequently, the performance of the PMWF is degraded. Therefore, we propose a noise power spectral density (PSD) estimation method for the PMWF in this paper. The proposed method is based on a consecutive process of eigenvalue decomposition on noisy input PSD, estimation of the target component contribution using directional information, and exponential weighting for improved estimation of the target contribution. For evaluation, four objective measures were compared with the MMCRA and we verify that the PMWF with the proposed noise estimation method can improve performance in environments where directional interfereces exist.

균일한 크기의 스탭을 갖는 양자기에서 발생하는 음성신호의 잡음해석

  • Park, Ui-Yeol;Park, Jong-Yeon
    • Proceedings of the KIEE Conference
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    • 1979.08a
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    • pp.8-11
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    • 1979
  • 음성신호의 진폭이 갖는 학율밀도함수(PDF)를 이용하여 음성신호가 균일한 스탭을 갖는 양자기 입력에 가해질 때 발생하는 잡음을 양자화 잡음 및 포화성잡음으로 분리하여 잡음에 관한 표현식을 유도하였다. 이러한 잡음 표현식에서 빗트수 및 포화점을 결정하는 상수에 관해서 수치해석함으로서 각 빗트마다 SNR을 최대로하기 위한 조건을 제시 하였다.

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A Simple Noise Reduction Method using SVD(Singular Value Decomposition) (SVD(Singular Value Decomposition)을 이용한 간편한 잡음 제거법)

  • Shin, Ki-Hong
    • Journal of the Korean Society for Precision Engineering
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    • v.16 no.2 s.95
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    • pp.116-122
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    • 1999
  • 저차 동적계(특히 카오스계)에서 측정한 시계열의 잡음을 제거하기 위해서 SVD(Singular Value Decomposotion)을 이용한 새로운 간편하고 매우 효과적인 새로운 잡음 제거법이 소개되었다. 이 방법은 위상궤적(phase portraint)을 재구성하는데 중점을 두었으며, 궤적행렬(trajectory matrix)을 구성하는데 그 기본을 두었다. 이 궤적행렬에 SVD를 반복적으로 사용하여 신호와 잡음을 분리하였다. 이 방법은 Duffing계에서 측정한 잡음이 섞인 카오스 신호에 적용되었으며, 또한 실험에 의한 진폭변조된 신호에도 적용되었다.

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Design of Audio Watermarks by Noise Shaping (잡음 형상화에 의한 오디오 워터마크 설계)

  • Lee, Jin-Geol
    • Journal of Korea Multimedia Society
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    • v.8 no.11
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    • pp.1432-1438
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    • 2005
  • A psychoacoustic model based noise shaping method is proposed. The method shapes the noise in the frequency domain such that its presence with a host signal will not be perceptually noticeable. The derivation of imperceptible noise levels from the masking thresholds of the signal involves deconvolution associated with the spreading function in the psychoacoustic model. It has been known as an ill-conditioned Problem. In this paper, a constrained optimization is applied such that the noise excitation level conforms to the masking thresholds of the signal. Thus, the noises embedded in the signal will not be perceived by human ear, and its performance is demonstrated experimentally.

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Distance Measures Based Upon Adaptive Filtering For Robust Speech Recognition In Noise (잡음 환경하에서 음성 인식을 위한 적응필터링 거리 척도에 관한 연구)

  • 정원국;은종관
    • The Journal of the Acoustical Society of Korea
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    • v.11 no.1E
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    • pp.15-22
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    • 1992
  • 잡음이 있는 환경하에서는 음성 인식의 성능이 현저하게 떨어지게 된다. 본 논문에서는 이렇나 잡음의 영향에 강한 거리척도를 제안하고자 한다. 우리는 잡음이 더해진 음성신호의 특징벡터를 깨끗한 음성신호의 특징벡터가 FIR 시스템을 거쳐 변형된 것이라고 가정한다. 여기서 FIR 시스템은 잡음의 영 향을 모델링한 것이라고 할 수 있다. 미지의 FIR 시스템 계수잡음의 영향을 모델링한 것이라고 할 수 있다. 미지의 FIR 시스템계수들은 RLS 적응 알고리즘을 이용하여 구한다. 제안된 거리척도는 적응 여파 기의 예측 오차에 관한 식으로 표시되어진다. 여러 가지 적응 여파기의 구조중 단일 채널 일차 FIR 구 조가 가장 좋은 음성 인식 성능을 보이며, 이 경우 효과적인 거리척도 알고리즘을 구할 수 있다. 여러 가지 신호대 잡음비에 관하여 화자독립 격리단어 인식 실험을 DTW 알고리즘을 이용하여 수행하여 본 결과 제안된 거리척도가 거의 모든 신호대 잡음비에 대하여 우수한 성능을 보였다.

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