It's not being played through a desktop audio stack, they're operating on the waveform decoded from the audio file. There's no constraints relative to the "natural speed" of the source, just like for video transcoding, it's limited by compute.
My point was that the "natural speed" of video (i.e. the frame rate - clear slow motion requires more frames, while fast blurred motion will generally have fewer) does actually impact how much data must be processed. It's true of audio too (sample rate) and downsampling is a common optimization technique.