[FFmpeg-user] aconvert filter problems

[Roger Pack]

> Thanks, that work around works, and is obviously easier.  I assumed I needed
> to use a filter to make the output codec happy, but apparently it does some
> sample-type conversion automatically.  Perhaps what is happening is the
> output codec for wav is defaulting to pcm_s16le and it is automatically
> converting it from float back to s16.

Yes it would be nice if ffmpeg printed out in more detail the various
filters going on, when it started...