10ms framesize limit and noise in denoised audio stream about python-webrtc-audio-processing

[xiongyihui]

It's a requirement of webrtc audio processing.
Not sure about the second question. Does the audio become a discontinuous signal?

[demeng]

Not sure about the second question. Does the audio become a discontinuous signal?

Xiongyi, thanks for the response. Here are two examples of audios processed by using different buffer sizes (chopped the original audio into small buffers, processed each buffer in order, then merged them all together)

This is the audio using buffer of 50ms.

To compare, this is the audio using buffer of 2s.

The audio with 50ms become very discontinuous, and it seems that there are artifacts introduced between boundaries of buffers. In fact, in the audio of buffer 2s, a (mild) discontinuity can be heard every 2 second. But it is much more noticeable when buffer is much smaller as it is much more frequent.

Any ways to solve this issue so that the noise suppressed audios sound the same regardless of buffer size? Any insight is appreciated!

[xiongyihui]

The audio using 50ms buffer is short than the one with 2s buffer. It seems it added some zeros every 50ms.

You should divide a audio into 10ms each, and do NS