I suspect they mean pre- and post-ringing which would 'mask' the actual speed of the transients. Hearing is close to logarithmic so a signal has to become very small in signal level when it isn't perceived any more. All the nice squarewave and impulse plots are all linear. So even when a signal appears to not be there any more in such a plot it is still there but say -30dB.
The longer the pre-and post ringing the 'worse' the temporal resolution is said to be.
The ear canal rings at 3kHz and attenuates around 20kHz already. The 0.7ms is the time it takes for the ear canal to stop ringing at around 3kHz. Most likely a certain attenuation 'border' is reached after that time. No idea what amplitude it is at that point 10% ? (-10dB) or more... no idea.
The filter rings at around 20kHz (for 44kHz files) which is inaudible and only rings when the recording has those steep transients in it or close to it.
It doesn't start to magically 'ring' or pre-echo or whatever at 20kHz when lower frequencies are present.
It never is a 0dB 'spike' as per the test signal. The 20kHz level is low in amplitude compared to the mids anyway. The 0.7ms thus only applies to the 3kHz signal in the ear canal which is NOT 'activated' by a 20kHz frequency.
Another thing I encourage anyone to do. You know what..... I am not spilling the beans and want those that want to know let them search for it themselves.
Here is what you should do:
Search for a piece of music that has transients, the fast sharp ones....
Then open the file with something like audacity.
Zoom in on those transients till you arrive at sample frequency levels.
Closely inspect that transient and see if you can spot an instant 'rise' in amplitude in the file .... within 1 or 2 samples to see if they are there during a perceived transient.
Then report back if you find anything in the recording that is that fast related to that perceived transient.
Highly educational....
have fun.