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 _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user
