Wavelet transform and variable-step least mean square algorithm-based voice denoising method

A wavelet transform and variable step size technology, applied in speech analysis, instruments, etc., can solve the problems of slow convergence speed, low calculation efficiency, fast convergence speed, etc., to reduce the degree of dispersion, improve calculation efficiency, and fast convergence speed. Effect

Active Publication Date: 2015-04-08
NORTHWESTERN POLYTECHNICAL UNIV
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Problems solved by technology

Although researchers at home and abroad have done a lot of work on adaptive filtering algorithms and put forward some improved algorithms, there are still many problems: (1) The contradiction between convergence speed and steady-state error cannot be fundamentally overcome: step If the length factor is large, the convergence speed will be fast, but the misalignment will be large; if the step size factor is small, the misalignment will be small, but the convergence speed will be slow; (2) The algorithm is sensitive to noise, which is more obvious only in environments with high signal-to-noise ratios. (3) The convergence speed is sensitive to the distribution of the eigenvalues ​​of the autocorrelation function matrix of the input signal: if the distribution is too scattered, the maximum If the difference from the minimum value is too large, the convergence speed will be very slow
Therefore, directly applying the existing adaptive filtering algorithm to the noise reduction system will cause problems such as slow convergence speed, large steady-state error, and low computational efficiency.

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  • Wavelet transform and variable-step least mean square algorithm-based voice denoising method
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  • Wavelet transform and variable-step least mean square algorithm-based voice denoising method

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Embodiment Construction

[0029] Now in conjunction with embodiment, accompanying drawing, the present invention will be further described:

[0030] The hardware environment used for implementation is: AMD Athlon(tm) 2.60G computer, 2.00GB internal memory, 128M graphics card, and the running software environment is: Matlab7.0 and Windows XP. We have realized the method that the present invention proposes with Matlab software. The pure speech is selected from the 863 Chinese speech recognition corpus, and the noise is obtained from the non-stationary noise signal of the jet aircraft cockpit in the NOISEX-92 database. The pure speech and the noise are linearly added in proportion to generate a noisy speech with a signal-to-noise ratio of -5dB Signal.

[0031] The present invention is specifically implemented as follows:

[0032] 1. Preprocessing: 8kHz sampling (40,000 sampling points in total) and 16-bit linear quantization are performed on the noisy speech signal and the reference noise signal with a ...

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Abstract

The invention relates to a wavelet transform and variable-step least mean square algorithm-based voice denoising method. The method is technically characterized by comprising the following steps of: reducing the dispersion degree of an input vector self-correlation matrix characteristic value of an adaptive filter by using a time-frequency local characteristic of a wavelet, and increasing step factors of the algorithm; and simultaneously, establishing a non-linear function relationship between the step factors and an error signal to ensure that the step factors are adaptively increased at the initial stage and the time change stage and are adaptively decreased at the steady state stage. The method not only can ensure a relatively high convergence speed and little maladjustment but also has certain robustness and denoising performance, so a better denoising effect can be obtained by combining the wavelet transform and the variable-step least mean square algorithm.

Description

technical field [0001] The invention relates to a speech noise reduction method based on wavelet transform and variable step size least mean square algorithm, which can be applied to various speech signal noise reduction processing systems. Background technique [0002] In the process of voice communication, it is inevitable to be interfered by the noise introduced from the surrounding environment and transmission media, the electrical noise inside the communication equipment, and even other speakers. These interferences eventually make the speech received by the listener no longer the original pure speech signal, but a noisy speech signal polluted by noise. As an effective means to control low-frequency noise, adaptive speech active noise reduction technology has been greatly developed in recent years. [0003] Adaptive noise cancellation system is a typical application of adaptive speech active noise reduction technology, it is an adaptive filter with two inputs: the adap...

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Application Information

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Patent Type & Authority Patents(China)
IPC IPC(8): G10L21/0208G10L21/0216
Inventor 郭雷程塨赵天云
Owner NORTHWESTERN POLYTECHNICAL UNIV
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