Let me explain…
I believe the array that is returned is an array of float values representing the amplitude of the audio for each sample from the given start time for the supplied time length. 44100 samples per second is a commonly used sample rate in audio. You are just grabbing the very first of these samples for the time period that you have obtained. This may not be representative of the amplitude at all (audio has the potential to massively fluctuate between sample points).
I would personally grab a few of these values (5 - 10 perhaps) and calculate the median value, then use ‘FInterpTo’ to smoothly transition between this and the next median value.
