Some muxers (eg, qtmux) only support raw AAC, so this allows linking
an encoder that outputs ADTS only to those muxers.
The conversion is simple (omit the first 7 or 9 bytes of the frame),
but has to be done in pre_push instead of handle_frame as 1.0 does
not seem to allow skipping bytes there as 0.10 used to.
Other conversions are not supported (yet).
The LOAS variant seems to have three different subvariants itself,
only one of them is implemented as my two samples happen to be
using that one.
The sample rate is not always reported correctly, as the "main"
sample rate is apparently sometimes half what it should be (both
of my samples report 24000 Hz there), and there are two other
parts of the subvariant with different sampling rates. One of them
is parsed, but not the other, as it's located after some other
large amount of variable data that needs parsing first, and there
seems to be a LOT of it, which is useless for our needs here.
This ends up being rather inconsequential, as ffdec_aac_latm,
which is the only decoder that can decode such streams, does not
need the sample rate on the caps anyway.
https://bugzilla.gnome.org/show_bug.cgi?id=665394
