Every vendor publishes a latency figure and none of them are measuring the same thing from the same place. Rather than repeat numbers that cannot be reproduced, this page explains what sets time-to-first-audio architecturally and gives you a script that gets the number for your region, your text and your network.
Time to first audio is the only latency number
Total synthesis time barely matters if you are streaming, because playback starts as soon as the first chunk arrives and the rest is generated while the listener is still hearing the beginning. What matters is the delay before sound starts, and the guarantee that generation stays ahead of playback.
The second condition has a name: the real-time factor, generation time divided by audio duration. An RTF of 0.3 means one second of speech takes 300 ms to synthesise, so the buffer fills three times faster than it drains and you will never stutter. An RTF above 1.0 means audio arrives slower than it plays and the listener hears gaps, no matter how quick the first chunk was.
What sets it
