Thanks for the tip.Will look into it.
On Wed, Mar 14, 2018 at 6:48 PM Gonzalo Garramuño <[email protected]> wrote: > > > 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 > > _______________________________________________ > Libav-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/libav-user >
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