Good advice, and I did just that.
First I created various 25s audio samples, then:

br=64000 # 64kbps
for track in *.*; do
    # trac.ffmpeg.org/wiki/Encode/AAC#fdk_aac
    ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac -profile:a aac_he_v2 -b:a $br -vbr 5 "${track}.HEv2_$br.aac"
    ffmpeg -i "$track" -vn -ac 2 -c:a libopus -b:a $br -vbr on "${track}.vbr$br.ogg"
done

The results are equal in size, but in quality Opus clearly wins in this scenario.

Big Question No. 2:
Is it possible to stream in VBR?


On Sun, 2023-04-16 at 01:02 +0000, Sam Kuper wrote:
On Sat, Apr 15, 2023 at 03:58:12PM -0700, Len Ovens wrote:
On Sat, 15 Apr 2023, D.T. wrote:
* personally, I always had the feeling that opus (used a lot by youtube)
isn't so good with noisy, grungy, fuzzy, guitarry music

Well, don't use something you don't like  :)

I would almost suggest to try them all. I would think changing the
codec is just a change the parameter, restart kind of thing. It
doesn't have to be on the target machine for testing, just use which
ever machine has the files on (or at least a good chunk of them).

To see what a given input file sounds like after being transcoded to
each of several different output formats, at a couple of different
bitrates in your range of interest, just run a for loop in Bash, no?
Something like (untested):

    for i in {m4a,ogg,opus}; do
        for j in {64k,96k}; do
            ffmpeg -i infile.flac -b:a "$j" outfile_"$j"."$i"
        done
    done

Tweak this for more detail if desired.

Then listen to the results, and go with whichever combination of bitrate
& codec you find most satisfying?

Sam
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