[schroffl]

This produces an error in version 7.1 for me:

[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

[atoav]

The theory behind it is simple: Subtract each audio sample in B from each audiosample in A.

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

[TheRealDunkirk]

> Then you can look how much of the signal has truly been affected by your 1000 bucks silver speaker cable.

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.

[Redoubts]

Have you ever checked out the https://hydrogenaud.io/index.php?action=forum forums?

[atoav]

Oh that sounds incredibly sad.

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.

[reliablereason]

use this instead: 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

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.

[gwbas1c]

It's because mp3 dramatically changes phase. As a result, merely mixing the inverted original won't leave you with what's filtered out.

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.)