[Libav-user] # of audio samples, calculated vs. codec context

[Paul B Mahol]

On 5/21/13, Brad O'Hearne <brado at bighillsoftware.com> wrote:
> On May 21, 2013, at 11:10 AM, Pradeep Karosiya <praks411 at gmail.com> wrote:
>
>> I had similar issue sometime back. I matched the number of sample to that
>> of
>> codec context. I have used buffering scheme and which still working for
>> me,
>> so I didn't explore any other option.
>> I used intermediate buffer to keep the remaining samples and pass them to
>> encoder when number of sample equals that of codec context.
>
> Thank you for your help, Pradeep. You not only confirmed the issue I"m
> seeing, but made clear that you had to roll your own code (vs. use some
> FFmpeg facility) to get it to work. That's the kind of informative answer I
> was looking for. I'll do the same.
>
> This situation would be good to address in either decoding_enocding.c or
> resampling_audio.c, and documentation somewhere.

One can use libavcodec/audio_frame_queue.c

>
> Thanks,
>
> Brad
>
>
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