System and method for improving audio speech quality

Abstract

The invention discloses a system and a method for improving audio speech quality comprises the following steps that: firstly, preemphasis and Fourier transform treatment of the current frame of time domain first audio signal output by an audio collecting device and the current frame of the time domain second audio signal output by an audio playing device are completed; secondly, according to the frequency domain error signal parameter and frequency domain learning rate parameter obtained by the previous frame of the current frame, a frequency domain second audio signal is corrected; thirdly, de-emphasis treatment and subtraction of the first audio signal and the second audio signal are completed, thereby eliminating the echo in the first audio signal; meanwhile, according to the corrected time domain second audio signal and frequency domain second audio signal and preemphasized time domain first audio signal, the frequency domain error signal parameter and frequency domain learning rate parameter of the current frame are calculated and used for the treatment of the next frame, thereby effectively eliminating speech echo; in addition, through estimating the residual echo and noise of the audio signal with the echo eliminated, the invention effectively eliminates the noise in the audio signal and greatly improves the speech quality.

Description

technical field [0001] The invention relates to a system and method for improving audio communication quality. Background technique [0002] Voip has been widely used in the Internet at present, and its voice communication quality has become a key factor, and the echo noise elimination technology has become the focus of attention. [0003] Usually, the echo is caused by an acoustic loop between the speaker and the microphone, that is, after the signal from one end of the call is played by the speaker at the other end of the call, it will be picked up by the microphone at the end of the call and sent back to the end of the call. of talkers are able to hear their own echo, severely affecting call quality. In addition, the steady state or non-stationary noise signal generated by the surrounding environment of the microphone also has a great impact on the call quality. [0004] At present, the most effective way to solve the echo is to use an adaptive filter to construct an ac...

AI Technical Summary

Problems solved by technology

[0006] Although the existing combination of adaptive filter, DTD, and non-linear processing module (Non-Linear Processor, NLP) can achieve a better echo cancellation function, but the existing duplex detection algorithms will have misjudgments to varying degrees. problems, resulting in distortion of the local signal, which reduces the quality of the voice

In the existing commonly used echo cancellation schemes, some special situations cannot be handled, including: two parties talking at the same time, one party in a call in an environment with strong background noise, non-linear signal distortion caused by microphones or loudspeakers and other factors, etc.

Under these circumstances, it is difficult for an adaptive filter to work normally, and it is prone to filter divergence, coefficient update errors, etc., which eventually leads to echoes that cannot be eliminated; moreover, the audio signal collected by the microphone will also bring noise and abnormality. Signals that are linearly distorted, which can seriously affect voice quality

Images