Hello.
Part of my group project involves a music playing system. What we have
is a bunch of stages connected together with buffers and pipes. The
first stage decodes an mp3 file with mplayer and sends the raw audio
down the pipe in 44.1KHz 16bit. No headers are ever sent (as far as I
know). What I'm trying to do is add an EQ stage. The problem is that the
samples which come in seem to be equalised to the maximum level possible
with 16bit. Therefore if I increase the volume of any band it will cause
clipping. The only solution I can think of at the moment is to just EQ
down, but that sucks a bit.
Can anyone shed some light on what it is I'm doing wrong?
Thanks.
Simon
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Received on Sun Mar 16 20:15:03 2008
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