Hello, newbie here.
This is what i'm trying to do: create a patch that will loop a period (currently i'm working with 8000ms) of live input audio continuously, and allow you to add new audio into that loop.
essentially, it's a loop station, like the Boss stompbox unit.
Eventually i'd like to extend it so that one can have several independent audio loops.
has anyone else worked on a similar project?
I'm using record~ and buffer~ to create and hold the 8000ms of audio. I then send index~ to dac~ (so you can hear the sample) and also send it to the original record~/buffer~ objects.
Here are the questions:
Are tapin~ and tapout~ better options than record~/buffer~/index~?
With looping back, I need to avoid a constant incrementation (feedback) and adding new audio which will quickly push the signal over the dac~ limit, so I've use normalize~ objects. Is there a better way?
Even with the normalize function, I am experiencing some clipping, especially on lower end frequencies. I'm not 100% clear on attentuation methods, and sometimes confuse myself =) is there some comprehensive online resource that'd help me understand how to keep all my sounds within the 1 -> -1 limit without destroying the dynamics in the audio? ie. the louds are still loud, softs are still soft...i suppose it would be a compressor of sorts? I've never encountered such clip sensitivity in any other audio software; it frightens me how quickly i'm hitting the limit of the dac~!!
i posted another query a few days ago re. my soundcard...i'm experiencing a getclocksource error on the ASIO 2.0 driver... how will this affect me? am i running without a clock now??
lots of stuff here, i suppose. many thanks for any assistance!