I need a framework of a sequenced sample player. This is becouse I want to start my own project, and don't want to invent everything from scratch. Ofcourse there is lot of this kind opensource applications, but I need the simplest. - It should use only /dev/dsp - the sequence should be in a form of an char array, where the bits decide what voice to play. For egzample: seq[]={1,0,0,0,2,0,1,3,0,3,1,0,2,0,1,0} seq[0]==1 means that in first step should be played only the first sample (bassdrum),seq[1]==0 means pause (no sample), seq[4]==2 second voice (snare), and seq[7]==3 means snare +bass drumm. - I need to understand how the sequence procesing, and sound mixing is done in real time, and how the gaps betwen sounds should be iplemented. - the samples should be held in raw arrays (16 of 8 bit), eventualy downloadable from hdd. - a simple dsp like lp-filter would be nice to see, becouse i am curios how the time needed to proces the signal should be counted into the latency (and buffer size). - there is also a need for bpm clock. - a simple interpolation would be nice. Ok. I think that this is all! :P Or wait!!!!! I have a better idea! I there a way to make those things using for example libmikmod? I would like to start a music program for live mixing of mod, and xm modules (plaing, and muting chanels, realtime patterns changing, looping sequences). If someoune is interested in helping, I am open for sugestions.