An Unsupervised Speech Enhancement Method Based on Robust Nonnegative Matrix Factorization and Data Fusion

A non-negative matrix decomposition and data fusion technology, which is applied in speech analysis, instruments, etc., can solve the problems of limited and limited speaker changes

Inactive Publication Date: 2017-12-12
PLA UNIV OF SCI & TECH
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Problems solved by technology

[0005] The purpose of the embodiments of the present invention is to provide an unsupervised speech enhancement method based on robust non-negative matrix decomposition and data fusion, which aims to solve the problem that existing supervised speech enhancement algorithms are limited to the language to which the speech content belongs, limited by The problem that speaker variation is limited by the type of noise

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  • An Unsupervised Speech Enhancement Method Based on Robust Nonnegative Matrix Factorization and Data Fusion
  • An Unsupervised Speech Enhancement Method Based on Robust Nonnegative Matrix Factorization and Data Fusion
  • An Unsupervised Speech Enhancement Method Based on Robust Nonnegative Matrix Factorization and Data Fusion

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[0039] In order to make the object, technical solution and advantages of the present invention more clear, the present invention will be further described in detail below in conjunction with the examples. It should be understood that the specific embodiments described here are only used to explain the present invention, not to limit the present invention.

[0040] The application principle of the present invention will be further described below in conjunction with the accompanying drawings and specific embodiments.

[0041] figure 1 It shows the flow of the unsupervised speech enhancement method based on robust non-negative matrix decomposition and data fusion of the present invention. As shown in the figure, the present invention is implemented in this way, an unsupervised speech enhancement method based on robust non-negative matrix decomposition and data The supervised speech enhancement method is implemented as follows:

[0042] S101. For the input time-domain signal y(...

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Abstract

The invention discloses a non-supervision speech enhancement method based robust non-negative matrix decomposition and data fusion. The method comprises windowing and framing input time-domain signals, and then performing Fourier transform and delivery to obtain the amplitude value of a sentence; performing estimation to obtain sparse phonetic components S and a noise basis matrix W (n); estimating speech components and noise components to obtain estimates of enhanced speech; decomposing the obtained estimates through a robust non-negative matrix, and fusing the decomposed estimates with estimates obtained through spectrum substraction (SS) and minimum mean square error (MMSE) through a geometric mean value filtering module to obtain final amplitude spectrum estimates; reconstituting the time-domain signals during speech enhancement through the amplitude spectrum estimates and the phases of noisy speeches. The non-supervision speech enhancement method based robust non-negative matrix decomposition and data fusion is not limited to the language that speech content belongs to, the change of speakers nor the variety of noise, and compared with traditional spectrum estimation algorithms of SS and MMSE which are based on stationarity assumption, can be free from dependence on the stationarity assumption and accurately estimate the frequency spectrum of stationary or mutational noise.

Description

technical field [0001] The invention belongs to the field of speech signal processing, in particular to an unsupervised speech enhancement method based on robust non-negative matrix decomposition and data fusion. Background technique [0002] Speech enhancement is of great significance both for improving the auditory effect of the speech signal and as a front-end processing to improve the performance of the speech recognizer. A key issue in the implementation of speech enhancement is noise estimation. In order to estimate the noise spectrum, some classical algorithms have been proposed, such as Spectrum Subtraction (SS), Minimum Mean Square Error (MMSE), etc., and have been widely used in voice communication. However, these methods are generally based on the assumption of stationarity of the noise, which is poor for spectral estimation of non-stationary abrupt noise. [0003] In order to estimate the spectrum of abrupt noise, noise estimation models based on dictionary lea...

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Application Information

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Patent Type & Authority Patents(China)
IPC IPC(8): G10L21/0216G10L21/0224
Inventor 孙蒙张雄伟李轶南
Owner PLA UNIV OF SCI & TECH
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