Allegedly, on or about 4 February 2018, Wolfgang Pfeiffer sent:
> it's definitely true that with fine tools you can encode to mp3's
> with a quality so high that for  me at least it's difficult to find a
> difference to the wav's they were encoded from, even with decent
> stereo equipment. Also true, I'm old, so I might have ruined ears
> enough to be unable to hear differences where they actually are. 
> 
> Easy test: try this in a dir with  wav's, and the command below will 
> (should) code them to mp3's. With the resulting mp3's I'd bet 
> anyone will have difficulties to find a remarkable difference 
> between the wav's and the mp3's ...
> 
> for f in *.wav; do ffmpeg -i "$f" -codec:a libmp3lame -qscale:a 0
> "${f/%wav/mp3}"; done

Well, if you're going to encode to an unusually high bit rate (that
example did it at 320kB/s), I'm going to agree with you (that most
people won't pick the difference).

However, I find most people encode MP3s to a much lower bitrate, where
I can hear burbles, squeaks and squealies, and the quieter nuances of
some music disappears completely.  There's also a number of old, and
not very good, codecs around, to which I notice that treble seems to be
lacking.  But it's the added noises that I particularly notice and
dislike.

One thing I notice with MP3 encoding that I can give it a wave with
specific lead-in and lead-out time, and the encoded file is missing
that (screwing up audio comprised of multiple files).  Sometimes to the
point where it's actually slightly cutting off the start of the audio. 
Whatever Audacity was doing behind the scenes tended to do that a lot.

-- 
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Linux 4.14.14-200.fc26.x86_64 #1 SMP Fri Jan 19 13:27:06 UTC 2018 x86_64

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