On 5/21/13, Brad O'Hearne <[email protected]> wrote: > On May 21, 2013, at 11:10 AM, Pradeep Karosiya <[email protected]> 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 > > > _______________________________________________ > Libav-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/libav-user > _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
