the source has Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
and the output has Audio: aac (LC), 44100 Hz, stereo, fltp, 64 kb/s I thought the whole point of using filters os to not have to resampling and fifo manually... What should I do now? Thanks, Marcelo 2017-02-16 1:33 GMT+01:00 Ratin Rahman <rat...@gmail.com>: > I have seen this kind of issues while rendering audio to ALSA but just > re-sampled myself, not using filters. I think what you need to keep an eye > on is how many bytes of raw data you are getting out of the decoder, and > does that makes sense for the number of channels that you specified. If the > filter is expecting n number of samples as inputs and the decoded buffer > size is less, then obviously you will see this error. Just print all the > values and see if they match. > > On Wed, Feb 15, 2017 at 2:24 AM, Marcelo Emmerich < > marcelo.emmer...@gmail.com> wrote: > >> Hi All, >> >> I am refactoring a streaming application from manually resampling and >> FIFO buffering audio to using filters. For testing I run the " >> filtering_audio.c >> <https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/filtering_audio.c>" >> example, however it does not work, I always get the error >> >> more samples than frame size (avcodec_encode_audio2) >> >> In my previous implementation I had this working by manually filling a >> FIFO and handling the resampling myself, however I need to switch to using >> filters. The current filtergraph in filtering_audio.c actually has an >> auto-inserted fifo filter, but I still get the error. >> >> What am I doing wrong? >> >> Thanks, >> Marcelo >> >> >> >> _______________________________________________ >> Libav-user mailing list >> Libav-user@ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/libav-user >> >> > > _______________________________________________ > Libav-user mailing list > Libav-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/libav-user > >
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