[FFmpeg-user] aconvert filter problems

[Davy Durham]

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.

Thanks

On 10/15/2012 02:03 AM, Roger Pack wrote:
>>     ffmpeg -i input_file.ext -map 0:0 -af "aconvert=flt:auto" -y foo.wav
>>     but it's always converting to s16
> I'm not sure why aconvert doesn't seem to be...converting it, as it were.
> You may be able to work around like
>
>   -acodec pcm_f32le
> ref: http://ffmpeg.org/trac/ffmpeg/wiki/audio%20types
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