Speech enhancement method of microphone array under non-stationary noise environment

Abstract

The invention relates to a speech enhancement method of a microphone array under a non-stationary noise environment. The probability of an effective signal is accurately judged by means of phase information between passageways in order to improve non-stationary noise environment interference resistant performance. The method which estimates the probability of the effective signal specifically comprises the following steps: (51) according to a required beam width, calculating a threshold T1=NL sin (theta)/Fs, wherein N represents frame length, L represents microphone unit distance, theta represents beam width, and Fs represents sample frequency, (52) calculating the valve of the sum of phase differences omega of input signals of all passageways on all frequency points pd (n, omega); and (53) when the pd (n, omega)>omega (M-1) T1, the effective signal probability exists on the frequency point p1 (n, omega)=1, otherwise, p1 (n, omega)=0, wherein M represents the number of microphones. Due to the fact that the phase information between the passageways is used, judgment to the probability of the effective signal is enabled to be more accurate, the signal to noise ratio of picked up voice is improved, whole performance is improved, and particularly non-stationary noise environment interference resistant performance is improved.

Description

technical field [0001] The invention relates to the field of microphone array voice enhancement, in particular to a microphone array voice enhancement method in a non-stationary noise environment. Background technique [0002] In applications such as video conferencing systems, vehicle hands-free systems, and speech recognition systems, it is usually necessary to pick up the target voice at a relatively long distance. However, due to the existence of background noise, interference noise and echo, the signal-to-noise ratio of the picked-up speech is usually low, and the speech enhancement technology based on the microphone array is one of the common means to improve the signal-to-noise ratio of the received speech. [0003] At present, there are two main types of microphone array speech enhancement technologies. One is based on conventional beamforming. The beam shape of this method is fixed and will not change with the change of the interfering sound signal. For example, th...

AI Technical Summary

Problems solved by technology

However, due to the existence of background noise, interference noise and echo, the signal-to-noise ratio of the picked-up speech is usually low, and the speech enhancement technology based on the microphone array is one of the common means to improve the signal-to-noise ratio of the received speech

[0003] At present, there are two main types of microphone array speech enhancement technologies. One is based on conventional beamforming. The beam shape of this method is fixed and will not change with the change of the interfering sound signal. For example, the reverberation sound has a better suppression effect, and the speech enhancement performance improves with the increase of the number of microphones, but the suppression effect on directional interference noise is limited; the other type is based on an adaptive beamforming algorithm, and its beam shape will vary with The adaptive beamforming algorithm can effectively suppress the directional interference noise, but in the weakly correlated noise field or in the scattered noise field, the performance is not as good as the traditional beamforming algorithm.

The above two algorithms usually require the interference noise to be a stationary process. However, in the actual use environment, the interference noise is usually sudden and non-stationary, such as speech interference. In this case, most of the current speech enhancement technologies have limited performance.

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