[Libav-user] Hard time encoding AAC (works for MP3) - what settings are needed?

[Jesper Taxbøl]

Den ons. 26. aug. 2020 kl. 23.02 skrev Jesper Taxbøl <jesper at taxboel.dk>:

> Den ons. 26. aug. 2020 kl. 22.35 skrev Carl Eugen Hoyos <
> ceffmpeg at gmail.com>:
> >
> > Am Mi., 26. Aug. 2020 um 22:01 Uhr schrieb Jesper Taxbøl <
> jesper at taxboel.dk>:
> > >
> > > I am struggling to make AAC encoding work with libav.
> >
> > (Your email is practically unreadable.)
> >
> > Contrary to mp3, there is no "raw" aac, you have to choose a muxer
> > like adts or mov.
> > (This is not 100% accurate, but like most other software, FFmpeg
> > does not support reading raw aac, faad might work.)
> >
> > Carl Eugen
> > _______________________________________________
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>
>
> I am sorry about the confusion of my writing.
>
> It seems I have understood some packetization wrong. Perhaps you can help
> me understand.
>
> When I run
>
> ffmpeg -i someaudiofile.mp3 -t 3 out.aac
>
> I was under the impression that the output would be raw aac without using
> a muxer?
>
> How do I identify the muxer used in this case?
>
> I basically want to mimic that command from my C++ program.
>
>
> Kind regards
>
> Jesper
>


Through search I found a patch that I could draw some inspiration from.

https://patchwork.ffmpeg.org/project/ffmpeg/patch/20170329234301.6182-1-p4olo_prete@yahoo.it/

Now the repo is updated with an ADTS muxer and the output can be played
with ffplay.

https://github.com/taxfromdk/audio_encode_test/blob/master/encode_audio.cpp

Thank you for the hints.

Jesper
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