Den ons. 26. aug. 2020 kl. 23.02 skrev Jesper Taxbøl <[email protected]>:
> Den ons. 26. aug. 2020 kl. 22.35 skrev Carl Eugen Hoyos < > [email protected]>: > > > > Am Mi., 26. Aug. 2020 um 22:01 Uhr schrieb Jesper Taxbøl < > [email protected]>: > > > > > > 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 > > _______________________________________________ > > Libav-user mailing list > > [email protected] > > https://ffmpeg.org/mailman/listinfo/libav-user > > > > To unsubscribe, visit link above, or email > > [email protected] with subject "unsubscribe". > > > 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/[email protected]/ 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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