As another comment mentioned, this is done by conferencing software to deal with packet loss. It sounds like they either loop the previous DCT frame and gradually fade out, or feed the time domain output into a reverb, then cross-fade from 100% dry to 100% wet if the buffer is about to run out (someone on HN mentioned a while back that this approach was patented).

You could maybe make an argument that it would be useful in live music settings to prevent a bad situation from sounding even worse, and maybe you'd put it on some audio software so you can sort of still enjoy playing music on a crappy system, but really, it's best to have hardware and software that can 100% guarantee keeping up with audio processing.