[FFmpeg-user] aconvert filter problems

[Davy Durham]

Hi, I'm trying to use ffmpeg to convert the audiostream from any given 
file into .wav with a floating point sample format.

My command line is:

    ffmpeg -i input_file.ext -map 0:0 -af "aconvert=flt:auto" -y foo.wav
    but it's always converting to s16

Even trying u8 for the format doesn't affect it. It's almost as if 
ffmpeg believes .wav only supports s16

any ideas?


output here: http://pastebin.com/fQyqYgXH


Thanks