I need some help with compression techniques (I think)

meter~ and levelmeter~ give visual feedback of audio levels. I use meter~.

My strategy to avoid clipping is setting the gain levels of my individual sources so that my mix in the loudest part of the piece comes close to but stays a couple of dB's under 0dB. To catch rogue peaks I put a limiter on the outputs. A brickwall limiter at -0.5dB with 0 gain will simply cut off at that amplitude and leave the rest unchanged. You could opt for a compressor plus limiter but my projects don't call for that.

if you want to be on the safe side you need to lower the volume about 1/n for n source signals between 0. and 1.

so for example when adding 8 vst instruments, you need to do [*~ 0.125] to their sum to make sure you dont exceed 1. at least in theory! there are some instruments, which are louder than 1.0 anway.

so i think the best solution is to add a little mixer in your patch, one which also uses meters to help you see whats going on.

I know that --- I already have a mixer in my system but at a higher level. In fact it was from adjusting volumes through the mixer that I realized the problem was more likely a clipping issue than a CPU issue.

But my design (and desired implementation) is such that I need to automate this stuff. The way it works is that I have virtual" ports to which any VST can send audio. Depending on other factors, those virtual ports may get mapped to separate physical outputs (when I'm running in 8-channel mode) or they may all go to one output. In fact there's often both because there might be a stereo monitoring pair independent of the 8 separate outputs.

The problem with the solution of multiplying by 0.125 when there are 8 VSTs is that those 8 VSTs will most likely not be running simultaneously. So the actual number of non-zero feeds to an output can change on the fly and it will also depend on exactly how loud is each VST at a particular point in time. So what I want is something that will to reduce the overall volume just enough when the sum of instantaneously live outputs will clip. I think a block limiter is what I need but I don't totally understand the best way to implement that directly in MaxMSP and I think I have to do it that way rather than use yet another plugin.

Hi
Peter McCulloch recently posted a nice little squasher:

Which may be of some use, if strapped across your main output.

Brendan

hi, hope that works for you(gen compressor with lookahead). Try to stay away from the hard-lim though, it just implements a tanh~ function. (A look up is the safest way to keep levels down but introduces distortion as you know.) Emphasis is also something custom i wouldn't use if you want a smooth output, it's just something like and don't know how to call it.
cheers

please report if you find something not working etc

poly guts (save as "newcomp1_1.p.maxpat'):

@Brendan Thank you, I'll try that out --- not sure how well it will work as it seems more like normalization than just reducing stuff that's going to clip

@Woyteg I'm still using Max 5 for live performance so can't leverage anything with gen in it. Nevertheless, I appreciate the opportunity to see these alternative solutions.

really? gen~ is great for compressors, that's a pitty.

why don't you use the good ol' omx, btw?

a gain controller for every physical output is a must have in any audio application, no matter how creative it looks otherwise. :)

@Roman Uhmmm, yeah...that's why I posted the question --- but I need something that's automatic and can handle an arbitrary number of signals going to the output. As I said in my first post, I'm not that fluent with the signal processing side of things.

@Woyteg The omx was one of the options I mentioned in my original post but if that's the right object, then I don't know how to best configure it --- I'm experimenting with all these things but figured I'd ask for some guidance from people who know it better than me.