It turns out that the files that I was using for testing were at 48000Hz, whereas the experimental stimuli were at only 16000Hz. In hindsight it then became obvious that the speed at which the audio was being read was correlated to the sampling rate of the file.
The simpler solution therefore was to tell the library what values to expect (in my case 16000Hz). Easy to address from a coder point of view, but I’m including an image here of how I integrated that into the code component of the experiment builder, in case somebody find themselves in a similar conundrum in the future.
