[swatcoder]

The technique introduces latency and distortion because it's subtracting an estimate of sound that's traveling/reflecting in the listening environment, which is imperfect and involves the speed of sound.

That latency is within the tolerance that users are comfortable with for voice chat, and much less than video processing/transfer is introducing for video calls anyway, so it's a very obvious win there. Especially since those users are most interested in just picking out clear words using whatever random mic/speaker configuration happens to be most convenient.

But musicians, for instance, are much more interested in minimizing the delay between their voice or instrument being captured and returned through a monitor, and they generally choose a hardware arrangement that avoids the problem in the first place. And that's not really a niche use case.