[FFmpeg-user] libavcodec adts/aac decoder

[Nisar Ahmed]

Ok, I changed samples parameter in SDL_AudioSpec to 1024, and multiplied
length of buffer returned in the callback by 2. It took me a while to
understand SDL_AudioSpec properly

Thank you for you help
Nisar


On Thu, Aug 1, 2013 at 9:07 PM, nisar <nisar.med at gmail.com> wrote:

> 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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