[FFmpeg-user] libavcodec adts/aac decoder

[nisar]

Thank you for you response.

The sound is better but the rate is slow and it seems that there is some of
the samples in the output are missing (silence) each frame. I think there
is some error in my calculations please help


On Thu, Aug 1, 2013 at 7:07 PM, Carl Eugen Hoyos [via FFmpeg-users] <
ml-node+s933282n4660391h40 at n4.nabble.com> wrote:

> 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
> FLTP to FLT?
>
> Your question may be better suited for the
> libav-user mailing list, Carl Eugen
>
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