Can this project not use re-sampling to support the 44100 sample rate?

作者您好，请问您有遇到过交叉编译到arm设备上后降噪速度非常慢的问题吗，这边试了一下rk1126的板子，3秒钟左右的音频需要10几秒的时间

你好 在使用的时候，我传入一个16000的音频，音频在经过处理后声音发生了改变拉长了。
降噪前后音频.zip

Hello cpuimage,
I try to run your project on the WSL system in windows using VScode.
However, I have no idea how to compile and run it.
Could you please give me some hints

Thank you!

Does webrtc ns support streaming processing, such as processing 640 audio samples at a time?

In fact this project's implement only support 8khz, 16khz sample

In the official implement, in order to support 32khz and 48khz, the input signal should be processed by a splitting filter and generate two bands or three bands signal, then applying noise suppression and synthesis back to one bank signal. I have review the codes and cannot find the related functions.

It‘’s a bug that this project don't contain these processing. In #1 , modfiy WebRtcNs_InitCore function is useless .

您好，请问这个项目可以使用在linux系统上吗？
期待您的回复，谢谢。