Implementation of an stream HQ audio in a WebRTC video call

As the title suggests, I am looking for a way to implement a WebRTC videocall application while ensuring maximum audio quality by routing virtual audio. For example, when you want to directly share audio from a program without incurring screen sharing.

I wonder if there is an open-source or paid technology that would allow this mechanism to be integrated into one of the many open-source WebRTC technologies (such as LiveKit, OpenVidu, etc.). Or is there a way to implement it by leveraging WebRTC?