does your prog has timespan output? caus compression is a function of time. if u think u can put one number in and get another number out and assume thats everythings going on your wrong. if your program print some sort of curvediagram with amplitude in the y-direction and time as x then maybe we could put a constant sinus at your input. but again rarely the inputs are constant so you can just throw that app in your bin lol.

read about it then try it. dont overdo it. try to mix your kick and your baseline into one channel and then add the compression and just turn the knobs and see whats gonna happen. As long as you have no idea what it does dont use it on masterchannel. Its easiest so understand/learn it over a baseline kick mix imo. Ive made some terrible tracks while learning this and i still do after 5 years lol

good luck

It is written, to essentially represent a moment in time for the sake of trying to understand some values.

The "Input" field represents the volume of the audio at this imaginary moment in time. (this would normally be the level going IN to the compressor from your DAW)

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that level in is also a function of time and not a constant value.

if ur lucky you can input a square puls and with the right compression settings you can get sinus out. lol. thats not really telling you anything right?

Yes, I realise that mate.

Just think of this program being a "freeze frame" of a moment in time, and you have to put the input level in manually, which would normally be done by the daw (over time)

Thanks for the advice though!

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It's the right idea, although you're ignoring common parameters like the attack, release, and knee, and as others have pointed out, a real compressor can't perform the "instant" compression you are assuming unless it has look-ahead (which most digital ones don't and no analog ones can).

Honestly, I think you're complicating a simple thing. A compressor reduces the gain above a certain level as opposed to the absolute volume. Maybe all these intermediate values will help you "understand", but they aren't going to help at all when it comes to applying it, which is almost all about having an intuitive sense of what parameters will cause what kind of change in the sound and dynamics.

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I have a habit of over-complicating things! lol

I know it doesnt include things like attack, release, knee, look ahead etc..., but I just was wanting to see if I had the theory right of what is actually happening at a given moment in time with the input signal and what effect the threshold / ratio has on the output volume.

I think doing this though has actually made me understand a little bit better what is actually going on, and now I may be able to apply this knowledge to some real world sounds!! haha

For example say I have a vocal that has an average db of -20db but has some peaks in it at -6db, then I could apply a compressor with a threshold of -20db and a ratio of say 10:1 to allow a more "even" dynamic range (the -6db peaks should now be at -18.6db ( -20 - (-20 - -6) / 10), I have yet to put this theory into practice though.

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Easiest way to understand it:

Go download s(m)exoscope vst plug-in, which is free. Adjust the time parameter and hit the freeze button to stop the time to see what compression is doing to the waveform. Or you could just render the output it and look at it through your DAW or audio editor.

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quote:

Originally posted by dj_alfi change your avatar for fucks sake.

why isnt it the easiest way to understand by trying and hear whats happening?

compression is like a senior citizen with a tv remote.

quote:

Originally posted by Jimb0b For example say I have a vocal that has an average db of -20db but has some peaks in it at -6db, then I could apply a compressor with a threshold of -20db and a ratio of say 10:1 to allow a more "even" dynamic range (the -6db peaks should now be at -18.6db ( -20 - (-20 - -6) / 10), I have yet to put this theory into practice though.

No, no, no, NO. Jesus, compressing a vocal with a 10:1 ratio? This is exactly why I said that throwing around some trivial math isn't going to help you apply it.

You shouldn't even be considering trying to squash a dynamic range of 20 dB down to 2 dB. You might use a ratio that high with a very low threshold as a kind of leveler, to eliminate sudden and unintended peaks in the source. If it's really truly necessary to increase the gain by 18 dB then you might use multiple stages of compression. But much more likely you should be adjusting your actual levels, EQ, and other parts of the mix if it has those kinds of problems.

My advice is to stop worrying about the nitty gritty of how compression works and start learning about how it's actually used, in practice, otherwise you're going to end up with some awful mixes.

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Agreed with others, ditch the math for the time being. You can use what ever method to help you understand, but for now use the compressor tweak settings, raise the volume adjust settings on compressor hear whats happening.