[FFmpeg-user] libavcodec adts/aac decoder

Carl Eugen Hoyos cehoyos at ag.or.at
Thu Aug 1 16:04:52 CEST 2013

nisar <nisar.med <at> gmail.com> writes:

> sample output format is AV_SAMPLE_FMT_FLTP

>     desiredSpec.format = AUDIO_F32SYS;

Did you run data through libswresample or the 
aresample filter in-between to convert from 

Your question may be better suited for the 
libav-user mailing list, Carl Eugen

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