[AVFilterGraph @ 0x6000008b59d0] More input link labels specified for filter 'aeval' than it has inputs: 2 > 1
[AVFilterGraph @ 0x6000008b59d0] Error linking filters
Failed to set value '[0:a][1:a]aeval=val(0)-val(1):c=same' for option 'filter_complex': Invalid argument
Error parsing global options: Invalid argument
But honestly the only thing you get is something that subjectively sounds exactly the same, but lower volume. Probably due to the fact that subjective sound experience is more related to the fourier transform of the waves than it is to the waves themselves.
That technique will work with simpler compression techniques, like companding. (Companding is basically doing the digital equivalent of the old Dolby NR button from the cassette days.)
You can do the same thing in your DAW¹ by putting A (e.g. the original) onto one channel and B (the processed sound) onto another. Then you invert the phase of B and listen to/export the sum.
This trick works also for audio gear that claims it does amazing things to your sound (here you just need to make sure to match the levels if they have been changed). Then you can look how much of the signal has truly been affected by your 1000 bucks silver speaker cable.
¹ Digital Audio Workstation, something as simple as Audacity should do the trick
I have a friend who has spent ridiculous sums of money on audio gear. Like, he's in his 50's, and still lives with his parents (in part) because of it. Over the years, I've learned I will never convince him that he's being fleeced, but I've wanted to make a site to host such A/B comparisons for a very long time, to perhaps get through to others what a waste most of the "audiophile" gear is.
On your A/B comparison website: I think it is important to make a "blind" test default. So they can listen to e.g. ten repetations and vote for for one each time and in the end they get a score which one they liked better. and by how much.
Because of course they want to hear the difference if there is an expensive price tag.
"Using the python library headspace, and a reverb model of a small diner, I began to construct a virtual 3-d space. Beginning by fragmenting and scrambling the more transient material, I applied head related transfer functions to simulate the background conversation one might hear in a diner. Tracking the amplitude of the original melody in the verse, I applied a loose amplitude envelope to these signals. Thus, a remnant of the original vocal line comes through in its amplitude contour."
ffmpeg -i original.wav -codec:a libmp3lame -b:a 192k output.mp3 && \
ffmpeg -i output.mp3 decoded.wav && \
ffmpeg -i original.wav -i decoded.wav -filter_complex "[1:a]aresample=async=1,volume=-1.0[inverted];[0:a][inverted]amix=inputs=2:weights=1 1" difference.wav