El 14/03/18 a las 09:03, Michael IV escribió:
Hi.I have the following case:
I am receiving audio stream which consist
of 2 channel float 32 (non planar) audio frames. Then I am trying to
convert those into
AV_SAMPLE_FMT_FLTP in order to encode with AAC codec. The problem is
that I receive that data as packets of size different from what my
AVFrame has. AVFrame for
this codec has 2 buffers,each 1024 samples,which is 4096 bytes per
channel (32bit sample size),right? So it looks like I have to fill the
frame with all 4096 bytes before pushing it into encoder?Is it
possible to submit 'custom' frames,with different amount of data from
what I am getting in codec context?
No. You need to buffer the data. Look at the av_audio_fifo* set of
functions for a simple way of doing it.
--
Gonzalo Garramuño
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