Hi guys just wondering if anyone can point me in the direction fo some useful resources for this kind of project.

I am basically looking to model my FATMAN FAT1 compressor in PD as part of a university assignment and also for my own use as the thing is rather old and crackly.

Any help would be appreciated, im not looking for someone to do the work for me just some information to get me started from.

Any info would be great, thanks

I don't know all the details of what makes an analog compressor sound analog, but here is the chapter on amplitude corrections (limiters and compressors) from Johannes Kreidler's Pd book. I find that the limiter patch is a better place to start if you want to make the regular kind of compressor.

And for analog saturation sounds, waveshaping works quite nicely -- take a look at hardoff's volume-leveling patch (I like using tabread4~ instead of tabread~ for less artifacts). You can also try cubic soft clipping and other types of nonlinear distortion.

I just made a simple compressor, and I did exactly what ichabod said above, start from the limiter patch, and all will be good.

It will probably be a waste of time trying to faithfully model the compressor in Pd, it's easy to create a simple compressor, but what you need to do is physically model the circuitry. This will mean all the non-linearities present in the circuit will be recreated, and the compressor will sound like a real analogue compressor. I wanted to model analogue filters in Pd, but it can't be done, what people have done in the past is written them in C and compiled them into externals.

That's if you wanted to do a realistic imitation of it, which for a uni project I assume you would want to do. Analogue modeling is a difficult one, I just completed a uni project on physical modeling of a guitar using Pd and I wanted to model a Moog filter to pass the guitar signal through, but I can't write in C.

Actually, I remember reading a thesis where someone modeled the tone controls of a guitar in C, and to get a difference equation for a filter they built the circuit in SPICE, which is freeware for PC, but there is a Mac version. Once you build the circuit it plots graphs of the output, and you can swap around components. I don't know if that's any help, but it could be a start seeing as I assume you built it yourself so will have the circuit diagrams for it.

I don't think modeling an analog-style compressor in Pd would be as difficult as building something like a Moog filter. The problem with making the filter in Pd is that it would need one-sample feedback loops, which can be done with [block~ 1] but is also cpu intensive. I don't know exactly what it is about analog compressors that give them their distinctive sound, but I do know that compressors don't have a feedback portion to them. All compressors, analog or digital, are non-linear, so non-linearity in and of itself isn't an issue.

I was thinking more of the actual components of the compressor circuit, and how they won't behave as expected, certainly as they get hot they begin behaving differently.
If you are modeling a compressor with valves would be different to solid state, and I don't know how you'd go about modeling valves.
I suppose what I'm trying to say is your bog standard digital 1 pole filter response is a straight line, whereas the analogue equivalent would not be a perfect straight line, as the resistors and capacitors themselves don't behave linearly, and don't have a <perfect>linear relationship. And also, [tanh~] would create a saturation effect, but not one that sounds truly analogue, as the response of a real overdrive pedal would not be a perfect [tanh~] line.
I've never found any decent papers on Analogue Modeling, which is annoying, especially as it is such an important part of contemporary DSP.
I'm pretty interested in this topic myself now, it would be useful to be able to model electronic circuits in Pd

I don't think the non-ideal qualities of resistors and capacitors need to come into play here unless you're going for a very, very specific and accurate model (i.e. you're working for Universal Audio). And even then there should be some well considered justification before going down that rabbit hole. Just because you can imagine some potential effect doesn't mean it's significant or worth painstakingly recreating.

Also tanh~ is useful because that's the large signal transfer characteristic of a transistor differential pair. Of course it's not band limited, but you can always oversample. More sophisticated distortion algorithms include some state information and aren't just memory-free waveshapers.