No. The best you can do with this AI architecture would be something like NVIDIA's NeMo, meaning you group the input audio stream into 2s blocks with 1s overlap and then run the speech recognition on that.

If you want real-time speech recognition with less than 0.5s of delay between speaking the word and it being fully recognized, then one needs to implement a different architecture. And that one is much more difficult and expensive to train that this one (which was already expensive).

That said, I want a fully offline and privacy-respecting voice assistant myself.

So attempting to build the AI for real-time streamed live English speech recognition will be my next project. I plan to ship it as an OpenGL-accelerated binary with WebRTC server so that others can easily combine my recognition with their logic. But it probably won't be free since I'm looking at >$100k in compute costs to build it. In any case, here's the waiting list: https://madmimi.com/signups/f0da3b13840d40ce9e061cafea6280d5...