Method for enhancing microphone array voice based on combined inhibition

Abstract

The invention provides a method for enhancing microphone array voice based on combined inhibition. The method comprises the following steps of: structuring a microphone array for receiving external signals; analyzing the signals and obtaining time delays of different array signals relative to benchmark array signals in the microphone arrays opposite to a target voice source; respectively performing time delay compensation on digital signals corresponding to the two microphones, obtaining the compensated signals; respectively performing subband decomposition on the compensated array signals, and then forming fixed beams on each subband; meanwhile, respectively using blocking matrixes on each subband to obtain noise reference signals on each subband; and then respectively removing the noise irrelative to the target voice form the fixed beam forming device on corresponding subband through an adaptive filtering processing algorithm, and then merging the subbands, thereby forming an initial gain signal; and meanwhile, making use of the previously compensated any two array signals to obtain a filter for inhibiting the noise signal related to the target voice through a recursive mutual power spectral density, thereby obtaining the final target voice signal through combining the initial gain signal.

Description

technical field [0001] The present invention relates to the technical field of speech enhancement, in particular, the present invention relates to a target speech enhancement algorithm oriented from a binary microphone small-scale array to a multi-element microphone large-scale array. Background technique [0002] At present, applications from mobile communication, teleconferencing, notebook voice communication, car calling to TV interaction, etc. are faced with the interference of various types of noise such as reverberation, echo and other environmental noise without exception, which causes inconvenience to both communication parties, so The problem of noise suppression is naturally raised and widely studied. At present, noise suppression or speech enhancement technology mainly has two directions. One is single-channel speech enhancement technology for single microphone. Limited; another direction is the microphone array speech enhancement technology. Compared with the si...

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Problems solved by technology

[0004] However, for the existing microphone array noise suppression technology, there are common characteristics of high algorithm complexity, large amount of calculation, or only for speech enhancement applications in a certain array environment, resulting in insufficient universality.

For example, the patent CN101351058 published on January 21, 2009, is a circular microphone array for orientation, which is only effective for specific applications. In addition, it expects to calculate the incident direction of other arc visits by obtaining the incident direction of a certain arc range, because the incident direction It is difficult to estimate, and the error is inevitable, so the error in the angle of incidence in other radians may be larger, which will have an adverse effect on the final speech enhancement; and the patent CN101369427 published on February 18, 2009, using the GSC structure, adopts The power spectral density estimation replaces the time domain or frequency domain signal, and performs adaptive estimation to obtain the estimated noise power spectrum, and then obtains the power spectral density of the beamforming signal by subtraction, and then performs post-filtering. In the beamforming algorithm of this algorithm, due to the need to calculate Power spectral density, when the number of array elements is large, the amount of calculation is relatively large compared with other methods. In addition, the power spectral density of noise is subtracted in beamforming, which is the same as the disadvantage of single-channel spectral subtraction, and it is easy to cause noise.

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