Have been trying to wrap my mind around this (compressors/sion) for some time, then today stumbled on the ADSR abstraction it (almost ) suddenly became a lottt clearer.

And from last night, when I was working with brinkman's st-limiter and finally (!-ugh!) understood it was NOT trying to limit the sounds that came in (I have been searching for the Best Possible noisegate I can find, since i use a mic and acoustic guitar) but instead is intended to prevent clipping (like a normalize~ function/patch i have previously written) and so bring down sounds that are too loud.

Sought out other limiters and found one I completely comprehended so added/modified it to take a floor value as well as ramp time as parameters, not just its limit value. And decided I might as well go "all the way" and just make my own compressor, since I understand things waaayyy better when I just make them myself.

Also modified a ADSR (attack/decay/sustain/release) abstraction to be triggered with a metro+metro duration, to run continuously. CPU overhead is not too bad so am thinking a compressor is within reach.

So my question is:

Why can't I, or perhaps even "How could I?", conjoin these two patches (below+images of their _help files) to create a compressor, which has a, d, s, r + floor/threshold + ceiling/limit.

Also, i have translated "threshold" to be my "floor" so: am I right that "ratio" is the ratio of what I am calling "floor" and "ceiling"?

Feels like there is sooo much I am NOT getting. But as my ever-loyal son put it "Yes. But you DO know so much."

Ever is the humble pure-d artist,
s

p.s. thanks, in advance for any and all help you might give. It is ALL much appreciated. And once, I get this put together I will share it, and properly credit those who made the original patches.

internals-limiter~_sv.pd
adsr_apb_sv.pd
adsr_apb_help_sv.pd

@svanya Hello Svanya.....
Yes, what you are calling the "floor" is (probably) the level at which the compression starts to operate, and is called the "threshold". Some compressors also have expanders, which apply a gain before the compression threshold and aid with the "levelling" of the signal..............

Expansion raises the gain before the threshold, but raises the noise floor of course at the same time, so often they have a gate as well to mute the input below a certain level.

The "ratio" is the ratio between the output level without compression, and the level with compression applied. What you are calling the "ceiling" is probably the level at which "hard" limiting is applied.............

A soft "knee" at the threshold level helps to reduce the "obviousness" of the change for the listener. Of course the hard limit is much better applied after the compressor to "cap" the final output and prevent distortion if the compression is insufficient.
Some studio compressors then automatically apply an output gain (compensation) to increase the overall level after compression, but for "live" use that is highly dangerous, as if you reduce the threshold or increase the ratio the "automatic gain" will induce acoustic feedback immediately.
David.

@whale-av
Thank you, so much, David, for the quick and informative response. Glad to hear I am on the right track. Just wrote the following update. So will go ahead and include it here.

::

Couldn't sleep: "dreaming in pure data" yet again!

But DID figure out how to make the noisegate I wanted (see below).

By adding a "basement" (old MS Excel programmer concepts :---) sorry.)) you can/I did:

drop to 0 everything below the basement;
leave alone everything above it and below the floor;
boost what it is above the floor and below the ceiling;
and limit everything above the ceiling.

A VERY cheap and easy way to create the gate.

Notes:

1 It loadbangs everything to 0 so you have to set the values to make it work;
2 Start with the ceiling and work down;
3 All the values are synchronized to the ceiling as percentages of its value, ex. 0.5 floor of a 0.5 ceiling equates to a 0.25 value (so if you change the ceiling all of the other values change accordingly);
4 The "meter" array on the right aligns with the "ramps" array on the left to calibrate the values;
5 The hradio shows in which "pocket" ex. "below basement", "below floor" etc., the current envelope is to better align the values.

TODO:

Look in to how to incorporate the ADSR patch to include those values/attributes which I will look into next. But am pretty darn sure it can just be added on after the fact so speak as another patch after this one to change the tonality of the sound.

Peace. ALL comments more than welcome.

Tally Ho! Happy pd-ing.
-svanya

internals-limiter~-help_sv.pd
internals-limiter~_sv.pd

@whale-av:
Thinking...thinking...:is the soft knee what is happening with the/what I am calling [v compramp] i.e. how fast the ramp occurs?

I do know this is not the traditional way to approach compression. And am hoping to learn more as I look into the ADSR patch to see what it is doing. I think alot of what has made this hard(er) to grasp is the words have various shades of meaning (on the from above side and the below side ex. threshold-"Above it?" or "Below it?") Thanks again. Peace. I really do appreciate all the effort you put into this comment (and the community in general).

@svanya As you say...... it's a lot easier to understand when you have made it yourself!
The term "below threshold" is when the incoming signal is not high enough to reach the threshold...... and "above" when it is higher...... it is sort of logical.....
Choosing the threshold level is crucial to the best use of the compressor.

Some compressors (AKG were the first I remember) introduce a tiny delay to the main signal so as to introduce the required compression before the signal arrives (catch all transients). That is called "feed forward compression" if I remember well. It is ok for the studio, but less of a good idea live work. Very good for a "limiter" though as it catches everything...

You will also see (on manufactured units) "attack" (time for compression to ramp up....... approx.? 0-10ms.... to allow short transients to pass more naturally... un-"chopped") and "release" (the time to slowly release the compression after the signal has dropped below the threshold again....... 50ms-2sec.... usually about 300ms is good for vocals)

Compression is good for 2 things and maybe more....... clamping the dynamic range, and so smoothing out the effects of bad microphone technique.......
...... and increasing the "power under the curve" and so increasing loudness without an increase in level....... which guitarists value highly, as the transformer/valve combination of a valve amplifier introduces distortion of the signal as the transformer reaches saturation, that we now call "compression".... (and other distortions, but keep that to yourself!)........ So compression is always used for mastering now, as there is no analogue tape (which compressed everything anyway in the "good ole days")......
Don't get me going!
All of the extra touches (knee, attack, release, etc.) have been added in an attempt to hide any unpleasant effects of the compression for the human listener......
David.

@whale-av
I have now added "lookahead" and "makeup gain": lookahead just to the delay line and the gain much that it scales the output from between the ceiling value and 1.

As to attack/decay: can I use the D02.adsr help patch to accomplish that end?

(Had thought to take all the ADSR stuff out of this loop as it seems better suited for midi work. But has me wondering.)

This version (as it currently stands) would not do distort affects as it caps everything at 1.

Note: for those following this: the patch above is broken. I have corrected it but will need to post it later. The middle moses is mis-wired: the correct version should take all values above the basement and below the floor and raise them to the floor.

@svanya Hello again Svanya...... sorry..... chronic toothache (and the associated painkillers) has fried my brain, and your patch looks like a spaghetti omelette to me at the moment! I will come back a few days after the extraction......
You should add this one to your arsenal though. http://www.katjaas.nl/compander/compander.html
The approach is very different and so I doubt that you will glean anything appropriate to your patch..... but it is a useful addition to your library..... extended or vanilla.....
David.

@whale-av
'Some compressors (AKG were the first I remember) introduce a tiny delay to the main signal so as to introduce the required compression before the signal arrives (catch all transients). That is called "feed forward compression"':

I incorporated this as a "lookahead" passing a variable to the initial delay line;

'You will also see (on manufactured units) "attack" (time for compression to ramp up....... approx.? 0-10ms.... to allow short transients to pass more naturally... un-"chopped") and "release" (the time to slowly release the compression after the signal has dropped below the threshold again....... 50ms-2sec.... usually about 300ms is good for vocals)';

I included these and the numbers you suggest as the ramp values for attack: from below threshold to the "ratio" value and decay for below-threshold to a percent between that value and 0 (think this is a good option, because it sounds really screwy when it drops to nothing) and for those values above-the-limit dropping to the limit.

'bad microphone technique.': For example?

'knee': how to add this? via ADSR envelope? not sure how I would go about make the curve (soft knee)...suggestions?

'your patch looks like a spaghetti omelette to me at the moment!': I was sorry to hear this for two reason: the pain you must have/are undergoing and that perhaps my patch was a "mess". (not time to be sensitive tho. so pushed on.) Hope all is doing better for you.

Regarding this update:

ALL:

the controls:
limit, ratio, threshold, and makeup gain are all calibrated on the fly using the updateControls subpatch which scales the "left" and "right" values of each control (using range messages) accordingly

all signals are grouped into 4 categories using the moses's: below-threshold, above-threshold-below-ratio, above-ratio-below-limit, and above-limit

limit:
"above limit" signals are brought down to this limit

ratio:
"above ratio-below limit" signals are left untouched

threshold:
"above threshold-below ratio" are brought up to the ratio value

above_env:
Is to be used as a noisegate meter if the value is a positive number then the threshold is above the env~, i.e. background noise

attack/decay and ?:
? because I am not certain, if in the context of compressors I have applied these concepts correctly:
attack: the length of time it takes to make the upward ramps, 0-10 ms
decay: the length of time it takes to make the downward ramps, 50-2000 ms

% drop:
what percent between 0 and the threshold the signal is to be decreased, in other words, filter the noise completely or just a little bit, 100% drops it to 0, 1% drops it 1% below the threshold (fyi: I found dropping it completey to 0 sounded "wrong")

makeup gain:
the amount to increase the final output, between the limit and 1

lookahead:
how much time the delay line should be between 0-100 ms (have NO idea what range this "should" be, so input is requested)

ramps:
graphical representation (from left to right) of the the threshold, ratio, and limit on a 0 to 1 scale

meterLimit:
the env~ (really just a vu) and a line representing where the threshold is on that same scale (0 to 1) in order to better judge where to set the values

which_range:
A meter showing from left-to-right where the current sound is (to be used as simple guide to setting the parameters): silence, above silence-below threshold, above threshold-below ratio, above ratio-below limit, and above limit

I DO hope this is useful to someone/anyone. And ,if nothing else, might act as a guide to others addressing this pretty-darn challenging concept.

I have included some features which I have not seen other compressors include, while there are others I might or probably will/should include later, ex. a soft knee on the curves.

All opinions/insight/daily feelings/usage of/ etc. etc. etc. are very welcome.

Peace and may your days, hearts, minds, and loved ones be over-flow-ering with Music.
-svanya

p.s. will probably move this lateron to the abstractions section, once I am certain everything is working as planned.

compressor_apb~-help_sv.pd
compressor_apb_sv.pd

Learned this last night (thru trial and error):

The threshold is not actually a noisegate: because all the noise is still there when the compressor kicks in.

i.e. sure: it works great at controlling the volume/gain window. But it drops the noise when there is no other sound. But just compresses it when the voice or guitar are playing.

Am in hot pursuit of a solution to this on another thread. So Do stay tuned.

This like many/all other pd problem: Shall be overcome.

Thanks for listening
-svanya

p.s. hope folks don't think this is a crosspost. It's just different threads on basically the same web-of-an-idea.