[Libav-user] filtering_audio.c problems
ratin3 at gmail.com
Thu Feb 16 02:33:45 EET 2017
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.emmerich at 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
> 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?
> Libav-user mailing list
> Libav-user at ffmpeg.org
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