I'd love for this to make a difference for simple video calls with relatives -- in my imagination, I set it up, and we find out we've been in the audio equivalent of the uncanny valley this whole time. But I'm afraid in reality, the default audio codecs used in WebRTC are probably good enough to make poor microphones and pure connection latency the bigger issue. Am I wrong?<p>Obviously, even if I'm right, this has a target audience -- musicians, interviewers and other people who talk for a living -- for which it might be fantastic.